Asterisk Information

Re: [asterisk-users] buffer-issue when piping live-streams into musiconhold

2008-01-17T14:32:00.000-08:00

Michael Kamleitner wrote:
> thx a lot russel...your hack actually works!! :)

Awesome. :)

> Meanwhile I've found something about the musiconhold-conf-option
> "cachertclasses", which might help in starting a separate instance for every
> caller. however, that didn't really work for me... probably this option only
> works for mode=files?!
>
> http://www.asterisk.org/doxygen/trunk/Config_moh.html
> http://lists.digium.com/pipermail/asterisk-commits/2007-November/017911.html

Well, that option only exists in Asterisk trunk, and is only relevant when using
realtime for music on hold. I assume you're probably using one of the released
versions of Asterisk, so this wouldn't be available.

> anyway, thx a lot for your suggestions :)

You're quite welcome. I'm glad I could help out.

--
Russell Bryant
Senior Software Engineer
Open Source Team Lead
Digium, Inc.

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Re: [asterisk-users] IAX Trunk between two Asterisks

2008-01-17T13:32:00.000-08:00

This is my configuration in the extensions.conf,
iax.conf at Site A and Site B, so anyone can help why
the call refused?

Site A:

[IPLink]
type=friend
context=IPLinkIncoming
host=192.168.2.3
usename=IPLink
secret=password
canreinvite=no
nat=no

[SiteBInternal]

exten => _2XX,1,Dial(IAX2/${EXTEN}@IPLink)
exten => _2XX,2,Playback(vm-nobodyavail)
exten => _2XX,3,Hangup()
exten => _2XX,102,Playback(tt-allbusy)
exten => _2XX,103,Hangup()

[IPLinkIncoming]

include => SiteBInternal
include => SiteBExternal

And at Site B:

[IPLink]
type=friend
context=IPLinkIncoming
host=192.168.2.2
usename=IPLink
secret=password
canreinvite=no
nat=no

[SiteAInternal]

exten => _2XX,1,Dial(IAX2/${EXTEN}@IPLink)
exten => _2XX,2,Playback(vm-nobodyavail)
exten => _2XX,3,Hangup()
exten => _2XX,102,Playback(tt-allbusy)
exten => _2XX,103,Hangup()

[IPLinkIncoming]

include => SiteAInternal
include => SiteAExternal

Regards
Bilal

------------------

> Hi All;
>
> I did an IP Trunk using IAX between two Asterisk
> boxes, now Asterisk A can send a call for B but B
> refuse it. The IAX type was configured to be
"friend"
> in the iax.con for Asterisk A and B, is there any
> thing else need to be done to let B accept the call
> from A?
>
> Also, I used an static IP address for the host when
I
> configured the iax client in the iax.conf file.
>
> Any help?
> Regards
> Bilal
>

I used to see this problem when I used to use IAX2.
Sometimes it would
just
go away. I seem to remember using insecure=very to
get it working but
I may
be wrong.

Anyways, post the relevant parts of your IAX2 confs
from both boxes and
someone might be able to spot something right off the
bat.

Thanks,
Steve Totaro

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Re: [asterisk-users] IMAP client in asterisk not trying to contact IMAP server

2008-01-17T13:19:00.000-08:00

On Jan 17, 2008 7:55 AM, KodaK <sakodak@gmail.com> wrote:
>
> Thanks, if that was in any of the docs I just completely glossed over
> it. I'll give it
> a shot.

Yes, I skipped over that in the docs. I'm good at that.

Thanks for the help.

I've also written up a quickie how-to on how to enable this on a
trixbox system. Don't know how helpful it is, but it's there.

http://www.trixbox.org/wiki/trixbox-imap

--J(K)

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Re: [asterisk-users] Asterisk desktop tools for OS X

2008-01-17T13:18:00.000-08:00

Thanks for your response guys. There are still some issues with the
code (Svn on SourceForge). I am working on getting these fixed up and
will post a message when its ready for download.

I will yell out if I need some Asterisk/Cocoa help. Thanks a lot.

On Jan 18, 2008 7:19 AM, Adrià Vidal <adriavidal@gmail.com> wrote:
> I'm interested too Devraj, please send a copy of if possible to try it.
> Thanks.
>
>
>
> On Jan 17, 2008 12:25 PM, Devraj Mukherjee <devraj@gmail.com> wrote:
> >
> >
> >
> > Hi everyone,
> >
> > I have been long working on a project ( http://asterisktools.org, to be
> > released under GPL) that aims to provide desktop tools for Macs. I am
> > finally getting to the release stages of this application and hope to
> > have an early BETA available next weekend.
> >
> > If there is anybody who is interested in this tool, please send me an
> > email as I am looking for people who can test the application for me
> > before we make a final release.
> >
> > The code is already available via SVN and there are some really cool
> > and thoughtful features.
> >
> > Thanks a lot.
> >
> > --
> > "I never look back darling, it distracts from the now", Edna Mode (The
> > Incredibles)
> >
> >
> > _______________________________________________
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >

http://lists.digium.com/mailman/listinfo/asterisk-users
> >
>
>
>
> --
> --
> Adrià Vidal
> adriavidal@gmail.com
> _______________________________________________
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>
> asterisk-users mailing list
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>

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>

--
"I never look back darling, it distracts from the now", Edna Mode (The
Incredibles)

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Re: [asterisk-users] buffer-issue when piping live-streams into musiconhold

2008-01-17T13:17:00.000-08:00

thx a lot russel...your hack actually works!! :)

Meanwhile I've found something about the musiconhold-conf-option "cachertclasses", which might help in starting a separate instance for every caller. however, that didn't really work for me... probably this option only works for mode=files?!

http://www.asterisk.org/doxygen/trunk/Config_moh.html
http://lists.digium.com/pipermail/asterisk-commits/2007-November/017911.html

anyway, thx a lot for your suggestions :)

regards,
michael

On Jan 17, 2008 9:52 PM, Russell Bryant < russell@digium.com> wrote:

Michael Kamleitner wrote:
> 10:00 I'm calling the pbx, musiconhold starts correctly to play the
> live-stream (almost live, with very small delay) - that's OK.
> 10:01 I hangup.
>
> -- than I pause for 20 min --
>
> 10:20 I'm calling a second time. However moh now doesn't stream live, but
> starts to continue playing the stream from 10:01. This goes on for about
> 30secs, then the replay stops for a second and continues at the correct
> position (once again, rather "live"). along I get this message at the
> console:

<snip>

> musiconhold.conf:
>
> [default]
> mode=custom
> application=/etc/asterisk/mohstream.sh
>
> mohstream.sh
>
> #!/bin/bash
> /usr/bin/wget -q -O - http://my.stream.com:8000 | /usr/bin/madplay -Q -z -o
> raw:- --mono -R 8000 -a -12 -

Most players don't work quite correctly with Asterisk MOH. For it to work the
way you expect, the player you are using must throw away the audio when Asterisk
isn't currently reading from the stream. There was a magic version of mpg123
(0.59r IIRC) that did that, and that is why it was the recommended version.

If you're reading from a raw TCP stream, then you can use the small streamplayer
utility included with Asterisk. Otherwise, I don't really have a good
suggestion for you right now. I suppose that you could use some sort of hack to
ensure that music on hold is always playing so that the stream is being serviced.

extensions.conf:

[moh_hack]

exten => hack,1,Answer
exten => hack,n,StartMusicOnHold(default)
exten => hack,n,While(1)
exten => hack,n,Wait(300)
exten => hack,n,EndWhile()

*CLI> originate Local/hack@moh_hack application Echo

--
Russell Bryant
Senior Software Engineer
Open Source Team Lead
Digium, Inc.

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--
Mag. Michael Kamleitner
- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
E-Mail: michael.kamleitner@gmail.com
Xing: https://www.xing.com/profile/Michael_Kamleitner
- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
Phone: +43 699 116 07 923
- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -
Web: http://www.kamleitner.com

Re: [asterisk-users] Asterisk desktop tools for OS X

2008-01-17T13:16:00.000-08:00

Hi Tzafrir,

Yes it does use the Manager Interface. It account does require "call"
level access. That may then result in "umlimited access" to Asterisk
(well to originate calls anyway). However I have made real conscious
efforts to filter messages that are being transmitted over the socket
so the application doesn't listen or talk on behalf of a single
extension.

If this is a concern, is every desktop application that integrates
using the Manager Interface a problem for Asterisk administrators?

Also, what is a way around it then? I see desktop tools for Asterisk
being one of the biggest advantages over traditional PBXes.

http://lists.digium.com/mailman/listinfo/asterisk-users
>

--
"I never look back darling, it distracts from the now", Edna Mode (The
Incredibles)

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Re: [asterisk-users] Polycom Remotely Cancel Call Forward

2008-01-17T13:04:00.000-08:00

Kevin Kiely wrote:
>
> I have a remote user on a Polycom IP Phone who has set call forwarding
> by accident and is away from the phone. Does anyone know of a way to
> remotely un-forward the phone? I tried to reboot the phone but that
> didn't work and removing the mac-phone.cfg caused problems
>
Remove the XML element tag from within mac-phone.cfg that it updated with the forwarding information and then reboot it again.

--
Bird's The Word Technologies, Inc.
http://www.btwtech.com/

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[asterisk-users] not understanding Cisco call manager connection for incoming calls

2008-01-17T12:55:00.000-08:00

I am connected to CCM and have a sip.conf entry like:

[CCMHEART]
type=friend
host=X.y.X.A
allow=ulaw
allow=alaw
allow=all
canreinvite=yes
qualify=yes
context=CCMHEART

In extensions.conf I have a context of:

[CCMHEART]
exten => s,1,Goto(default,s,1)

exten => 45801,1,Goto(default,s,1)
exten => 4545801,1,Goto(default,s,1)

I do have a default context.

However calling the above 4545801 number asterisk
does not answer as it says it cannot find the 45801
in the current context.

Once I put the 3 context lines above in the default context
asterisk answers just fine.

Why do I need to put the 3 lines in the default context?
The sip.conf entry has the context being "CCMHEART" shouldnt it look there?

Jerry

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Re: [asterisk-users] buffer-issue when piping live-streams into musiconhold

2008-01-17T12:52:00.000-08:00

Michael Kamleitner wrote:
> 10:00 I'm calling the pbx, musiconhold starts correctly to play the
> live-stream (almost live, with very small delay) - that's OK.
> 10:01 I hangup.
>
> -- than I pause for 20 min --
>
> 10:20 I'm calling a second time. However moh now doesn't stream live, but
> starts to continue playing the stream from 10:01. This goes on for about
> 30secs, then the replay stops for a second and continues at the correct
> position (once again, rather "live"). along I get this message at the
> console:

<snip>

> musiconhold.conf:
>
> [default]
> mode=custom
> application=/etc/asterisk/mohstream.sh
>
> mohstream.sh
>
> #!/bin/bash
> /usr/bin/wget -q -O - http://my.stream.com:8000 | /usr/bin/madplay -Q -z -o
> raw:- --mono -R 8000 -a -12 -

Most players don't work quite correctly with Asterisk MOH. For it to work the
way you expect, the player you are using must throw away the audio when Asterisk
isn't currently reading from the stream. There was a magic version of mpg123
(0.59r IIRC) that did that, and that is why it was the recommended version.

If you're reading from a raw TCP stream, then you can use the small streamplayer
utility included with Asterisk. Otherwise, I don't really have a good
suggestion for you right now. I suppose that you could use some sort of hack to
ensure that music on hold is always playing so that the stream is being serviced.

extensions.conf:

[moh_hack]

exten => hack,1,Answer
exten => hack,n,StartMusicOnHold(default)
exten => hack,n,While(1)
exten => hack,n,Wait(300)
exten => hack,n,EndWhile()

*CLI> originate Local/hack@moh_hack application Echo

--
Russell Bryant
Senior Software Engineer
Open Source Team Lead
Digium, Inc.

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[asterisk-users] Paging Recording File

2008-01-17T12:37:00.001-08:00

I am looking to see if anyone has seen this problem before. I am
setting the MEETME_RECORDINGFILE variable in a macro, then using the r
option with the Page application to record the page. But the page is
only recorded to the file specified in MEETME_RECORDINGFILE
sometimes... Sometimes it works and sometimes it doesn't. When it
doesn't work it places the recorded file in the sounds dir with a
meetme-conf-..... name. Here is my Macro.

Basically it is getting my phones that begin with a certain number
from the realtime database to create a variable with a value that ='s
SIP/6001&SIP/6002&SIP/6003.... this is passed to the macro as ARG1

I added a System command to log the variables to a text file so I know
when the page is made, the variables are correct.

[macro-pageall]
; Context for paging all devices.
; This will search the sip table in the realtime database
; for all phones that start with a number. That number is
; passed to this macro as ${ARG1}.
;
; ARG1 = The first digit of the phones to be paged
; ARG2 = Device for the PA system. If the user selected to
; page the PA system. That will be included.
;
exten => s,1,Set(MEETME_RECORDINGFORMAT=wav)
exten => s,2,Set(MEETME_RECORDINGFILE=custom/paging/${EPOCH})
exten => s,3,System(/bin/echo "${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)} $
{MEETME_RECORDINGFORMAT} ${MEETME_RECORDINGFILE}" >> /var/log/asterisk/
pagemacro_var.log)
exten => s,4,MYSQL(Connect connid ${realdb_host} ${realdb_user} $
{realdb_pass} ${realdb_db})
exten => s,5,MYSQL(Query resultid ${connid} SELECT\ name\ FROM\ sip\
WHERE\ name\ LIKE\ "'${ARG1}%'")
exten => s,6,MYSQL(Fetch fetchid ${resultid} number)
exten => s,7,GoToIf($["${fetchid}" = "1"]?8:10)
exten => s,8,Set(pagedevice=${pagedevice}&SIP/${number})
exten => s,9,GoToIf($["${fetchid}" = "1"]?6:10)
exten => s,10,Set(pagedevice=${pagedevice:1})
exten => s,11,MYSQL(Clear ${resultid})
exten => s,12,MYSQL(Disconnect ${connid})
exten => s,13,GoToIf($["${ARG2}" != ""]?14:15)
exten => s,14,Set(pagedevice=${pagedevice}&${ARG2})
exten => s,15,SIPAddHeader(Call-Info:answer-after=0)
exten => s,16,SIPAddHeader(Alert-Info: Ring Answer)
exten => s,17,NoOp(Page Recording ${MEETME_RECORDINGFILE})
exten => s,18,Set(CALLERID(all)=System Page <1010>)
exten => s,19,Page(${pagedevice},r)

;On hangup, run script that will email the recording to shared
conference.
exten => h,1,System(/var/lib/asterisk/scripts/mail_lastpage ${ARG1} $
{MEETME_RECORDINGFILE})
exten => h,2,Hangup()

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[asterisk-users] Polycom Remotely Cancel Call Forward

2008-01-17T12:37:00.000-08:00

I have a remote user on a Polycom IP Phone who has set call forwarding by accident and is away from the phone. Does anyone know of a way to remotely un-forward the phone? I tried to reboot the phone but that didn’t work and removing the mac-phone.cfg caused problems

Re: [asterisk-users] Asterisk desktop tools for OS X

2008-01-17T12:19:00.000-08:00

I'm interested too Devraj, please send a copy of if possible to try it. Thanks.

On Jan 17, 2008 12:25 PM, Devraj Mukherjee <devraj@gmail.com> wrote:

Hi everyone,

I have been long working on a project ( http://asterisktools.org, to be
released under GPL) that aims to provide desktop tools for Macs. I am
finally getting to the release stages of this application and hope to
have an early BETA available next weekend.

If there is anybody who is interested in this tool, please send me an
email as I am looking for people who can test the application for me
before we make a final release.

The code is already available via SVN and there are some really cool
and thoughtful features.

Thanks a lot.

--
"I never look back darling, it distracts from the now", Edna Mode (The
Incredibles)

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--
--
Adrià Vidal
adriavidal@gmail.com

[asterisk-users] buffer-issue when piping live-streams into musiconhold

2008-01-17T12:15:00.000-08:00

Hi Folks,

I'm currently trying to configure musiconhold (on a asterisk-1.4.17) for replaying a live mp3-stream (Icecast2). after reading the related material on voip-info and several other pages, I've successfully tried out mpg132, madplay and mplayer to pipe a stream into moh.

however, there is one major problem involving some kind of buffer-issue. let me try to explain this problem using a timeline:

10:00 I'm calling the pbx, musiconhold starts correctly to play the live-stream (almost live, with very small delay) - that's OK.
10:01 I hangup.

-- than I pause for 20 min --

10:20 I'm calling a second time. However moh now doesn't stream live, but starts to continue playing the stream from 10:01. This goes on for about 30secs, then the replay stops for a second and continues at the correct position (once again, rather "live"). along I get this message at the console:

[Jan 17 20:37:15] NOTICE[6220]: res_musiconhold.c:544 monmp3thread: Request to schedule in the past?!?!
[Jan 17 20:37:15] NOTICE[6220]: res_musiconhold.c:544 monmp3thread: Request to schedule in the past?!?!

I've installed the ztdummy-module as I've read that the message "Request to schedule in the past?!?!" might have something to do with that, however this didn't help.

It looks like there's some kind of buffering going on...

Thanks a lot for any suggestions, at this point I'm rather clueless ;)

regards,
michael

musiconhold.conf:

[default]
mode=custom
application=/etc/asterisk/mohstream.sh

mohstream.sh

#!/bin/bash
/usr/bin/wget -q -O - http://my.stream.com:8000 | /usr/bin/madplay -Q -z -o raw:- --mono -R 8000 -a -12 -

Re: [asterisk-users] Device state of SIP doesn't change

2008-01-17T12:01:00.001-08:00

Atis Lezdins wrote:
> Hi,
>
> I'm wondering - why SIP device state doesn't get updated to anything
> else, except Not In Use.
>
> For queue call (with Local channel) i get:
> app_queue.c: Device 'SIP/21168' changed to state '1' (Not in use)
> app_queue.c: Device 'SIP/21168' changed to state '1' (Not in use)
> app_queue.c: The device state of this queue member, Agent/21168, is
> still 'Not in Use' when it probably should not be! Please check
> UPGRADE.txt for correct configuration settings.
>
> Of course, i checked UPGRADE.txt, and lot of other resources, enabled
> few settings in sip.conf, but this still doesn't change.
>
> my sip.conf is:
> [general]
> port = 5060
> bindaddr = 0.0.0.0
> context = default-external
> tos_sip=0x18
> tos_audio=0x18
> callerid = Unknown
> dtmfmode=rfc2833
> ignoreregexpire=yes
>
> limitonpeer=yes
> notifyringing=yes
> notifyhold=yes
> allowsubscribe=yes
> call-limit=1
>
> and the corresponding realtime entry is:
> name: 21168
> accountcode: NULL
> amaflags: NULL
> callgroup: NULL
> callerid: device <21168>
> canreinvite: no
> context: default-sip
> defaultip: NULL
> dtmfmode: rfc2833
> fromuser: NULL
> fromdomain: NULL
> fullcontact: NULL
> host: dynamic
> insecure: NULL
> language: NULL
> mailbox: 21168@device
> md5secret: NULL
> nat: yes
> deny: NULL
> permit: NULL
> mask: NULL
> pickupgroup: NULL
> port: 5061
> qualify: no
> restrictcid: NULL
> rtptimeout: NULL
> rtpholdtimeout: NULL
> secret: xxx
> type: friend
> username: 21168
> disallow:
> allow: all
> musiconhold: NULL
> regseconds: 1200593168
> ipaddr: xxx.xxx.xxx.xxx
> regexten:
> cancallforward: yes
> setvar:
>
> Any help would be appreciated.
>
> Regards,
> Atis

The relevant portion of UPGRADE.txt mentions that a call-limit is necessary in
order for SIP devices to report proper device state. I see in your sip.conf file
that you have set call-limit in the general section. This setting, however, may
only be set per peer (or user). Unfortunately, there's no warning message output
if an unrecognized option is set in the general section.

Mark Michelson

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[asterisk-users] asterisk-1.2.26.tar.gz Thoughts?

2008-01-17T12:00:00.000-08:00

What are people's thoughts on asterisk 1.2.26? Any show stopping bugs?

Re: [asterisk-users] SIP Proxy Issues

2008-01-17T11:51:00.001-08:00

On Jan 17, 2008 2:28 PM, Nicholas Blasgen <nicholas@blasgen.com> wrote:

I've set up plenty of Asterisk boxes but never one that had to deal with a proxy server to be able to use a line. Using "X-Lite" I have no issue with settings as follows:

Display Name: Any Name

User name: 00575000010XXXX

Password: 00575000010XXXX

Authorization user name: <blank>

Domain: directnationalloan.com

Checked "Register with domain" and "Send outbound via: Proxy Address: las-obproxy.voipzone.us"

X-Lite has no issues with registration or placing calls.

Now the fun part, Asterisk I've been able to get to register.

register => 00575000010XXXX@directnationalloan.com: 00575000010XXXX:00575000010XXXX@las-obproxy.voipzone.us

It's the placing of calls that I'm getting an error. I've tried so many different configurations that it's somewhat pointless to show you my settings. The one I've been playing around with most recently is:

[voipexten]
auth=00575000010XXXX:00575000010XXXX@las-obproxy.voipzone.us
username=00575000010XXXX
secret=00575000010XXXX
fromdomain= directnationalloan.com
type=peer
qualify=yes
insecure=port,invite
outboundproxy=las-obproxy.voipzone.us

But of corse that doesn't work. Maybe someone here has an idea.

--
/Nick

Try dropping the auth line and changing the outboundproxy to host= ?

Thanks,
Steve Totaro

[asterisk-users] PostgreSQL query results truncated 255 characters

2008-01-17T11:51:00.000-08:00

I am querying an postgresql database from my 1.4.13 system and the results seem to be truncating each column at 255 characters. The columns are typed as character varying 1000.

Any suggestion on how to remove this limit?

TIA

Vic

[asterisk-users] SIP Proxy Issues

2008-01-17T11:28:00.000-08:00

I've set up plenty of Asterisk boxes but never one that had to deal with a proxy server to be able to use a line. Using "X-Lite" I have no issue with settings as follows:

Display Name: Any Name

User name: 00575000010XXXX

Password: 00575000010XXXX

Authorization user name: <blank>

Domain: directnationalloan.com

Checked "Register with domain" and "Send outbound via: Proxy Address: las-obproxy.voipzone.us"

X-Lite has no issues with registration or placing calls.

Now the fun part, Asterisk I've been able to get to register.

It's the placing of calls that I'm getting an error. I've tried so many different configurations that it's somewhat pointless to show you my settings. The one I've been playing around with most recently is:

[voipexten]
auth=00575000010XXXX:00575000010XXXX@las-obproxy.voipzone.us
username=00575000010XXXX
secret=00575000010XXXX
fromdomain= directnationalloan.com
type=peer
qualify=yes
insecure=port,invite
outboundproxy=las-obproxy.voipzone.us

But of corse that doesn't work. Maybe someone here has an idea.

--
/Nick

Re: [asterisk-users] More voicemail cards needed...

2008-01-17T11:04:00.000-08:00

TMOB

http://support.t-mobile.com/knowbase/root/public/tm22131.htm

Thanks,
Steve Totaro

On Jan 17, 2008 1:54 PM, Justin Newman < nt_jnewman@yahoo.com> wrote:

Thank you all for the voicemail cards you sent.

If you have the following in PDF or laying around (scan):

* AT&T/Cingular flow voicemail card
* Verizon flow voicemail card
* Sprint flow voicemail card
* TMobile flow voicemail card
* Alltel flow voicemail card
* Avaya Nortel Octel flow voicemail card
* Comedian Mail (Asterisk) -- I have the flow, need a card if someone has one

I will work on getting these integrated with EVM. Users will be able to select via user prefs and admin on a per user setting of their preferred VM flow.

Final prompts are coming this week; need the cards for any additions.

I have the following: Audix, 3COM, IC IMail, NEC EliteMail LX, Nortel Call Pilot, Olle's, and a customized Octel. Feel free to send others that may be of interest.

Send all cards to: nt_jnewman at yahoo.com.

Justin

____________________________________________________________________________________
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know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ

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[asterisk-users] More voicemail cards needed...

2008-01-17T10:54:00.000-08:00

Thank you all for the voicemail cards you sent.

If you have the following in PDF or laying around (scan):

* AT&T/Cingular flow voicemail card
* Verizon flow voicemail card
* Sprint flow voicemail card
* TMobile flow voicemail card
* Alltel flow voicemail card
* Avaya Nortel Octel flow voicemail card
* Comedian Mail (Asterisk) -- I have the flow, need a card if someone has one

I will work on getting these integrated with EVM. Users will be able to select via user prefs and admin on a per user setting of their preferred VM flow.

Final prompts are coming this week; need the cards for any additions.

I have the following: Audix, 3COM, IC IMail, NEC EliteMail LX, Nortel Call Pilot, Olle's, and a customized Octel. Feel free to send others that may be of interest.

Send all cards to: nt_jnewman at yahoo.com.

Justin

____________________________________________________________________________________
Be a better friend, newshound, and
know-it-all with Yahoo! Mobile. Try it now.

http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ

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Re: [asterisk-users] modem through Zaptel/Digium?

2008-01-17T10:39:00.000-08:00

On Jan 17, 2008 1:28 PM, Jeremy Mann <jmann@txhmg.com> wrote:

Is it bridging the Zap channels? We have asterisk doing FXO-FXS modem calls working fine, the key is making sure the channels are bridging and EC is NOT turning on. If you have anything preventing that the modem calls won't work.

-----Original Message-----
From: asterisk-users-bounces@lists.digium.com [mailto: asterisk-users-bounces@lists.digium.com] On Behalf Of Dave Fullerton
Sent: Thursday, January 17, 2008 12:05 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] modem through Zaptel/Digium?

Greg Woods wrote:
> This is just a low priority curiosity question because I have a usable
> workaround.
>
> I have Digium card that uses the Zaptel driver (can't get to my home
> machine right now to get the exact model, but it probably doesn't
> matter). It's a card with one POTS line and three extension hookups. I'm
> using Asterisk 1.4 and Zaptel 1.4.7 .
>
> One of the extension ports is connected to a modem on another computer.
> This is a FAX modem that works well; I have * programmed to detect
> incoming faxes and route them to this modem, and it works seamlessly. I
> can also send outbound faxes with no problem.
>
> The curiosity is that this modem does not work for dialup unless I
> bypass the * server and connect it directly to the wallplate, then it
> works fine. I don't see why it would be able to detect carrier and
> negotiate with a fax machine through * and Zaptel, but not with a dialup
> server.
>
> --Greg

I think asterisk has the ability to detect fax tones and disable echo
cancellation for those calls. I don't know if that is the case with a
regular modem call. I'd check to make sure that echo cancellation is
disabled on the extension the modem is plugged into. The only other idea
is to try connecting at a lower speed (I would think this would happen
automatically though).

-Dave

Try setting the modem to 9600 baud. It will probably work.

Thanks,
Steve Totaro

Re: [asterisk-users] modem through Zaptel/Digium?

2008-01-17T10:28:00.000-08:00

Is it bridging the Zap channels? We have asterisk doing FXO-FXS modem calls working fine, the key is making sure the channels are bridging and EC is NOT turning on. If you have anything preventing that the modem calls won't work.

-----Original Message-----
From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Dave Fullerton
Sent: Thursday, January 17, 2008 12:05 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] modem through Zaptel/Digium?

Greg Woods wrote:
> This is just a low priority curiosity question because I have a usable
> workaround.
>
> I have Digium card that uses the Zaptel driver (can't get to my home
> machine right now to get the exact model, but it probably doesn't
> matter). It's a card with one POTS line and three extension hookups. I'm
> using Asterisk 1.4 and Zaptel 1.4.7 .
>
> One of the extension ports is connected to a modem on another computer.
> This is a FAX modem that works well; I have * programmed to detect
> incoming faxes and route them to this modem, and it works seamlessly. I
> can also send outbound faxes with no problem.
>
> The curiosity is that this modem does not work for dialup unless I
> bypass the * server and connect it directly to the wallplate, then it
> works fine. I don't see why it would be able to detect carrier and
> negotiate with a fax machine through * and Zaptel, but not with a dialup
> server.
>
> --Greg

I think asterisk has the ability to detect fax tones and disable echo
cancellation for those calls. I don't know if that is the case with a
regular modem call. I'd check to make sure that echo cancellation is
disabled on the extension the modem is plugged into. The only other idea
is to try connecting at a lower speed (I would think this would happen
automatically though).

-Dave

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Re: [asterisk-users] [IAX] Up-to-date list of soft- and hardphones?

2008-01-17T10:09:00.000-08:00

Cavalera Claudio Luigi wrote:
>> -----Original Message-----
>> From: asterisk-users-bounces@lists.digium.com
>> [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of
>> Gordon Henderson
>>
>> However, you'll need to do similar things to your asterisk
>> box & router if
>> it's behind NAT for IAX as you do for SIP. (You will need a static IP
>> address on the NAT router and port-forward 4569 to the
>> asterisk box, just
>> as you'd port-forward 5060 and 10000-20000 for SIP)
>
> Please correct me if I'm wrong, for Iax clients you don't need to do
> static port-forwarding as they will create upon registration one entry
> in NAT table with UDP port for both signalling and media. On the other
> hand, sip clients (without Stun) are difficult to manage behind Nat
> because of RTP/RTCP ports.
> I don't want to start a flame Iax vs Sip, just to clarify respective
> advantages.
>
> Best Regards,
> Claudio

I believe you are correct, as long as the client sends *something* to
the server at frequent enough intervals that the router keeps the
connection in it's active list.

-Dave

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Re: [asterisk-users] modem through Zaptel/Digium?

2008-01-17T10:05:00.000-08:00

-Dave

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[asterisk-users] Voicemail Callback

2008-01-17T09:47:00.000-08:00

Hi all

Someone has make a voicemail callback on * ?
Thanks

--
Gilberto Nunes

Itajaí - SC

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Re: [asterisk-users] Asterisk desktop tools for OS X

2008-01-17T09:39:00.000-08:00

Yaah!!! Mac! I am a big user of OS X. Can't help it. Macs eye candy draws me in like my wofe. :) And.. I've never had a single issue with it. I also host virtual Ubuntu, Red Hat and XP :( on the same box using VMware.

Sorry about the Mac rant. Just glad to see some Mac / Asterisk attention...

I have multiple Asterisk servers in place and would REALLY be interested in your tool set. I can test it on Leopard or Tiger as I have both in available.

Thanks,
Jim

----- "Devraj Mukherjee" <devraj@gmail.com> wrote:
> Hi everyone,
>
> I have been long working on a project (http://asterisktools.org, to
> be
> released under GPL) that aims to provide desktop tools for Macs. I
> am
> finally getting to the release stages of this application and hope to
> have an early BETA available next weekend.
>
> If there is anybody who is interested in this tool, please send me an
> email as I am looking for people who can test the application for me
> before we make a final release.
>
> The code is already available via SVN and there are some really cool
> and thoughtful features.
>
> Thanks a lot.
>
> --
> "I never look back darling, it distracts from the now", Edna Mode
> (The
> Incredibles)
>
> _______________________________________________
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>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>

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[asterisk-users] Device state of SIP doesn't change

2008-01-17T09:36:00.000-08:00

Hi,

I'm wondering - why SIP device state doesn't get updated to anything
else, except Not In Use.

For queue call (with Local channel) i get:
app_queue.c: Device 'SIP/21168' changed to state '1' (Not in use)
app_queue.c: Device 'SIP/21168' changed to state '1' (Not in use)
app_queue.c: The device state of this queue member, Agent/21168, is
still 'Not in Use' when it probably should not be! Please check
UPGRADE.txt for correct configuration settings.

Of course, i checked UPGRADE.txt, and lot of other resources, enabled
few settings in sip.conf, but this still doesn't change.

my sip.conf is:
[general]
port = 5060
bindaddr = 0.0.0.0
context = default-external
tos_sip=0x18
tos_audio=0x18
callerid = Unknown
dtmfmode=rfc2833
ignoreregexpire=yes

limitonpeer=yes
notifyringing=yes
notifyhold=yes
allowsubscribe=yes
call-limit=1

and the corresponding realtime entry is:
name: 21168
accountcode: NULL
amaflags: NULL
callgroup: NULL
callerid: device <21168>
canreinvite: no
context: default-sip
defaultip: NULL
dtmfmode: rfc2833
fromuser: NULL
fromdomain: NULL
fullcontact: NULL
host: dynamic
insecure: NULL
language: NULL
mailbox: 21168@device
md5secret: NULL
nat: yes
deny: NULL
permit: NULL
mask: NULL
pickupgroup: NULL
port: 5061
qualify: no
restrictcid: NULL
rtptimeout: NULL
rtpholdtimeout: NULL
secret: xxx
type: friend
username: 21168
disallow:
allow: all
musiconhold: NULL
regseconds: 1200593168
ipaddr: xxx.xxx.xxx.xxx
regexten:
cancallforward: yes
setvar:

Any help would be appreciated.

Regards,
Atis

--
Atis Lezdins
VoIP Developer,
IQ Labs Inc.
atis@iq-labs.net
Skype: atis.lezdins
Cell Phone: +371 28806004
Work phone: +1 800 7502835

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Re: [asterisk-users] Iax Encryption

2008-01-17T09:28:00.000-08:00

> -----Original Message-----
> From: asterisk-users-bounces@lists.digium.com
> [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of
> Russell Bryant

> > I would like to understand if someone is using this in production.
>
> I have no idea if anyone is using it. It's easy to use, so I
> assume that some
> people are ...
>

I guess what you are meaning here is it's easy to configure on asterisk
side.
So this encryption is now considered robust enough to be used in
production?
I'm asking this because of comments I've found here:
http://www.voip-info.org/wiki/index.php?page=IAX%20encryption
about beta stage encryption.

Thanks,
Claudio

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Re: [asterisk-users] Iax Encryption

2008-01-17T08:54:00.000-08:00

Cavalera Claudio Luigi wrote:
> Is this the libiax used currently on asterisk
> http://ftp.digium.com/pub/libiax/ ?

No. Asterisk has its own IAX2 implementation.

> I would like to understand if someone is using this in production.

I have no idea if anyone is using it. It's easy to use, so I assume that some
people are ...

> Moreover which Iax client do you use to test this?

I'm actually not aware of any IAX clients that have implemented encryption.

--
Russell Bryant
Senior Software Engineer
Open Source Team Lead
Digium, Inc.

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[asterisk-users] Asterisk SVN mirror back up to date

2008-01-17T08:46:00.000-08:00

The public Asterisk SVN mirror is back up to date. I apologize for the
inconvenient downtime. Re-syncing with a repository that has almost 100,000
revisions took a while. :)

--
Russell Bryant
Senior Software Engineer
Open Source Team Lead
Digium, Inc.

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[asterisk-users] Iax Encryption

2008-01-17T08:41:00.000-08:00

Hello,
from what I've understood Iax2 should support aes128 encryption.
I've found this old info:
http://www.voip-info.org/wiki/view/IAX+encryption
and this (unanswered?) post
http://lists.digium.com/pipermail/asterisk-security/2005-August/000060.h
tml
Is this the libiax used currently on asterisk
http://ftp.digium.com/pub/libiax/ ?
I would like to understand if someone is using this in production.
Moreover which Iax client do you use to test this?

Best Regards,
Claudio

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[asterisk-users] modem through Zaptel/Digium?

2008-01-17T08:38:00.000-08:00

This is just a low priority curiosity question because I have a usable
workaround.

I have Digium card that uses the Zaptel driver (can't get to my home
machine right now to get the exact model, but it probably doesn't
matter). It's a card with one POTS line and three extension hookups. I'm
using Asterisk 1.4 and Zaptel 1.4.7 .

One of the extension ports is connected to a modem on another computer.
This is a FAX modem that works well; I have * programmed to detect
incoming faxes and route them to this modem, and it works seamlessly. I
can also send outbound faxes with no problem.

The curiosity is that this modem does not work for dialup unless I
bypass the * server and connect it directly to the wallplate, then it
works fine. I don't see why it would be able to detect carrier and
negotiate with a fax machine through * and Zaptel, but not with a dialup
server.

--Greg

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[asterisk-users] sip channel - redirection - which context is used

2008-01-17T08:25:00.000-08:00

Hi,

When asterisk receives 302 Moved Temporary sip response what is the logic for selecting the domain and context to use?

Thanks for any help
Tomasz

Re: [asterisk-users] AEL includes?

2008-01-17T08:21:00.000-08:00

AEL was an experimental feature in Asterisk 1.2.x and you may not implement all funcionts.

Jay Moore wrote:
> voip*CLI> ael reload
> Jan 17 08:53:30 NOTICE[20600]: pbx_ael.c:1146 handle_root_token: Unknown
> root token '#include'
>
> Asterisk 1.2.14. Old, I know but my boss won't spring for a spare box,
> and I don't want to upgrade our only production computer.
>
> Jay
>
> Rodrigo R Passos wrote:
>
>> Jay,
>>
>> What error?
>>
>>
>> Jay Moore wrote:
>>
>>> How do I include a file (not a context) in AEL? #include "filename"
>>> returns an error.
>>>
>>> Thanks,
>>> Jay
>>>
>>> _______________________________________________
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>>>
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>>> To UNSUBSCRIBE or update options visit:
>>>

http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>>
>>>
>> _______________________________________________
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>>

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>>
>>
>
>
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>

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Re: [asterisk-users] AEL includes?

2008-01-17T07:16:00.000-08:00

On 1/17/08, Jay Moore <jaymoore@accu-com.com> wrote:
> voip*CLI> ael reload
> Jan 17 08:53:30 NOTICE[20600]: pbx_ael.c:1146 handle_root_token: Unknown
> root token '#include'
>
> Asterisk 1.2.14. Old, I know but my boss won't spring for a spare box,
> and I don't want to upgrade our only production computer.

I suppose, that it doesn't support AEL2. You can dump ael to conf file
with command i posted before. Oh, and you will need to grab 1.4, and
compile aelparse from it.

Regards,
Atis

>
> Jay
>
> Rodrigo R Passos wrote:
> > Jay,
> >
> > What error?
> >
> >
> > Jay Moore wrote:
> >> How do I include a file (not a context) in AEL? #include "filename"
> >> returns an error.
> >>
> >> Thanks,
> >> Jay
> >>
> >> _______________________________________________
> >> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> >>
> >> asterisk-users mailing list
> >> To UNSUBSCRIBE or update options visit:
> >>

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> >>
> >>
> >
> >
> > _______________________________________________
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> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >

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> >
>
>
> _______________________________________________
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>

--
Atis Lezdins
VoIP Developer,
IQ Labs Inc.
atis@iq-labs.net
Skype: atis.lezdins
Cell Phone: +371 28806004
Work phone: +1 800 7502835

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Re: [asterisk-users] Single T1 with DIDs

2008-01-17T07:12:00.000-08:00

Steve,

That is very helpful, How much are we talking about in terms of the loop and minute charges. If you want it offline I can send you a private my with my phone number.

On 1/17/08, Steve Totaro <stotaro@totarotechnologies.com> wrote:

On Jan 17, 2008 5:23 AM, broadband Voice <broadbandvoice@gmail.com > wrote:

Can anyone share their experience with me? I am looking for a provider that delivers Dialtone over T1 to terminate to my asterisk box and also provide DIDs. Does the DIDs come with the T1 services or those are purchased/charged seperately. Any help greatly appreciated. My target markets are Philadelphia and Washington DC Metro areas.

I would be glad to help you out with this as I have T1s in both PA and MD and have been through all the paces with all of the big players in the area from T1s to T3s.

I pay $.65 per DID per month on top of the loop and minute charges.

Thanks,
Steve Totaro

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Re: [asterisk-users] AEL includes?

2008-01-17T06:57:00.000-08:00

voip*CLI> ael reload
Jan 17 08:53:30 NOTICE[20600]: pbx_ael.c:1146 handle_root_token: Unknown
root token '#include'

Asterisk 1.2.14. Old, I know but my boss won't spring for a spare box,
and I don't want to upgrade our only production computer.

Jay

Rodrigo R Passos wrote:
> Jay,
>
> What error?
>
>
> Jay Moore wrote:
>> How do I include a file (not a context) in AEL? #include "filename"
>> returns an error.
>>
>> Thanks,
>> Jay
>>
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>>

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>

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Re: [asterisk-users] Voicemail systems- flow charts, digit/key cards, etc

2008-01-16T21:01:00.001-08:00

I too would like this, Please feel free to post a link on the list :)

Regards
Kevin

Justin Newman wrote:
> Does anyone have flow charts or digit/key cards for some of the more popular voicemail systems out there?
> (shows which digits/keys to press, where it takes you, etc.)
>
> I need to create some of the new voicemail system.
>
> Send 'em my way if you have them.
>
> nt_jnewman at yahoo.com
>
> Justin
>
>
> ____________________________________________________________________________________
> Looking for last minute shopping deals?
> Find them fast with Yahoo! Search.

http://tools.search.yahoo.com/newsearch/category.php?category=shopping
>
> _______________________________________________
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>

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>

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Re: [asterisk-users] Voicemail systems- flow charts, digit/key cards, etc

2008-01-16T21:00:00.000-08:00

I have the ones from T-Mobile & Sprint PCS and probably the New AT&T
Wireless... email me if you are interested.

On Jan 16, 2008 11:27 PM, Justin Newman <nt_jnewman@yahoo.com> wrote:
> Does anyone have flow charts or digit/key cards for some of the more popular voicemail systems out there?
> (shows which digits/keys to press, where it takes you, etc.)
>
> I need to create some of the new voicemail system.
>
> Send 'em my way if you have them.
>
> nt_jnewman at yahoo.com
>
> Justin
>
>
> ____________________________________________________________________________________
> Looking for last minute shopping deals?
> Find them fast with Yahoo! Search.

http://lists.digium.com/mailman/listinfo/asterisk-users
>

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Re: [asterisk-users] Voicemail systems- flow charts, digit/key cards, etc

2008-01-16T20:43:00.000-08:00

3Com http://www.sjc.cc.nm.us/documents/ots/docs/VoiceMailGuide.pdf
NEC Elitemail http://gigshowcase.com/EndUserFiles/2912.pdf

A system similar to Elitemail would rock!

Thanks,
Steve Totaro

On Jan 16, 2008 11:27 PM, Justin Newman <nt_jnewman@yahoo.com > wrote:

Does anyone have flow charts or digit/key cards for some of the more popular voicemail systems out there?
(shows which digits/keys to press, where it takes you, etc.)

I need to create some of the new voicemail system.

Send 'em my way if you have them.

nt_jnewman at yahoo.com

Justin

____________________________________________________________________________________
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Re: [asterisk-users] IMAP client in asterisk not trying to contact IMAP server

2008-01-16T20:34:00.000-08:00

> I'm trying to test IMAP in 1.4.17 and it appears to be not working.
>
> I've compiled imap-2007 with the following on a CentOS 5 box:
>
> make slx EXTRACFLAGS="-I/usr/include/openssl -fPIC"
>
> and I've configured and compiled asterisk with the following:
>
> ./configure --with-imap=/usr/local/src/imap-2007

And now in "make menuselect" you have to go to voicemail options and set IMAP
support to on.

> Here's my voicemail.conf:
>
> [general]
> imapserver=localhost
> imapfolder=Inbox
> ;pollmailboxes=yes
> ;pollfreq=30
> imapflags=notls
> authuser=asttest
> expungeonhangup=yes
> authpassword=whatever

I had to enable pollmailboxes in order to update MWI.

__Yehavi:

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[asterisk-users] Voicemail systems- flow charts, digit/key cards, etc

2008-01-16T20:27:00.000-08:00

Does anyone have flow charts or digit/key cards for some of the more popular voicemail systems out there?
(shows which digits/keys to press, where it takes you, etc.)

I need to create some of the new voicemail system.

Send 'em my way if you have them.

nt_jnewman at yahoo.com

Justin

____________________________________________________________________________________
Looking for last minute shopping deals?
Find them fast with Yahoo! Search.

http://tools.search.yahoo.com/newsearch/category.php?category=shopping

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Re: [asterisk-users] Problem with a channel

2008-01-16T19:50:00.000-08:00

The problem is that i have random hangup in calls in the PSTN.

After that I check in asterisk -rvvvvvv
Sip show channels

And I see the extension....

The only way that I can place another call in the extension was to restart
the Asterisk.

-----Mensaje original-----
De: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] En nombre de Moises Silva
Enviado el: Miércoles, 16 de Enero de 2008 09:31 p.m.
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: Re: [asterisk-users] Problem with a channel

And the problem is? ...

I think you should read this: http://catb.org/~esr/faqs/smart-questions.html

Regards,

Moisés Silva

On Jan 16, 2008 6:42 PM, Ruben Zamora <ruben.zamora@zys.com.mx> wrote:
>
>
>
>
> I have install a Server with Centos 1 TDM400: Asterisk 1.4.9, Zaptel
1.4.5
>
>
>
> I having these problem :
>
>
>
> Zap/2-1 is busy
>
> Hangup ZAP/2-1
>
> Everyone is busy/congested at this time (1:1/010)
>
> Autofallthrough channel "SIP/202-b7b08ab0" Status is busy.
>
>
>
> And then HANGUP.
>
>
>
>
> _______________________________________________
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>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>

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>

--
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to get out."

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Re: [asterisk-users] Problem with a channel

2008-01-16T19:31:00.000-08:00

And the problem is? ...

I think you should read this: http://catb.org/~esr/faqs/smart-questions.html

Regards,

Moisés Silva

On Jan 16, 2008 6:42 PM, Ruben Zamora <ruben.zamora@zys.com.mx> wrote:
>
>
>
>
> I have install a Server with Centos 1 TDM400: Asterisk 1.4.9, Zaptel 1.4.5
>
>
>
> I having these problem :
>
>
>
> Zap/2-1 is busy
>
> Hangup ZAP/2-1
>
> Everyone is busy/congested at this time (1:1/010)
>
> Autofallthrough channel "SIP/202-b7b08ab0" Status is busy.
>
>
>
> And then HANGUP.
>
>
>
>
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>

http://lists.digium.com/mailman/listinfo/asterisk-users
>

--
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to get out."

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Re: [asterisk-users] Unable to open master device '/dev/zap/ctl'

2008-01-16T19:09:00.000-08:00

any version of asterisk not create nodes into /proc/zap
create to command, view into make file how to create nodes

On Jan 16, 2008 8:48 PM, Walter Willis < walterwn@gmail.com> wrote:

create nodes and links /proc/zap

On Jan 16, 2008 3:39 PM, Chris Bagnall <lists@minotaur.cc> wrote:
Make sure asterisk is in the "dialout" group in /etc/passwd

The default gentoo ebuild of zaptel creates /dev/zap/* with group dialout, and if you're using the gentoo ebuild of asterisk, it'll run as asterisk:asterisk, so you need to make sure asterisk is a member of the dialout goup otherwise it'll never be able to access /dev/zap/*

FWIW, as a fellow Gentoo user, 2006.1 is a bit dated, and you'd be well worth updating to 2007.0 if you can spare the time - it'll save you a lot of messing around with gcc versions etc. later down the line.

Regards,

Chris
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Re: [asterisk-users] Can DB() use SQLite instead of BerkeleyDB?

2008-01-16T18:23:00.000-08:00

On Wednesday 16 January 2008 16:22:10 Vincent wrote:
> On Wed, 16 Jan 2008 12:10:35 -0600, Tilghman Lesher
>
> <tilghman@mail.jeffandtilghman.com> wrote:
> >No, it cannot. You could use func_odbc to formulate your own queries,
> >though.
>
> Thanks. I don't like ODBC, but if it's stable and not a pain to
> install/use, that could be the solution.

It's not a pain, other than the multiple configuration files. In fact,
it's really quite versatile, especially given that ODBC drivers exist for
virtually every database out there.

> Otherwise, there's a new solution to use MySQL:
>
> http://www.voip-info.org/wiki/view/Asterisk+cmd+MYSQL

That's nothing new. It's been there since pre-1.0.

--
Tilghman

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Re: [asterisk-users] Anyone Using a Dell PowerEdge T105 in Production

2008-01-16T17:54:00.000-08:00

On Jan 16, 2008 7:28 PM, Steve Totaro <stotaro@totarotechnologies.com> wrote:
>
> You can add the raid option for $199. I think I might pickup about ten of
> them at this price. I can always resell them as general purpose servers or
> even workstations if Asterisk/Zaptel/Linux does not like the boxen.

Ahh - nice. That wasn't an option when I ordered the SC440.

-erik

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Re: [asterisk-users] Unable to open master device '/dev/zap/ctl'

2008-01-16T17:48:00.000-08:00

create nodes and links /proc/zap

On Jan 16, 2008 3:39 PM, Chris Bagnall <lists@minotaur.cc> wrote:

Make sure asterisk is in the "dialout" group in /etc/passwd

The default gentoo ebuild of zaptel creates /dev/zap/* with group dialout, and if you're using the gentoo ebuild of asterisk, it'll run as asterisk:asterisk, so you need to make sure asterisk is a member of the dialout goup otherwise it'll never be able to access /dev/zap/*

FWIW, as a fellow Gentoo user, 2006.1 is a bit dated, and you'd be well worth updating to 2007.0 if you can spare the time - it'll save you a lot of messing around with gcc versions etc. later down the line.

Regards,

Chris
--
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For full contact details visit http://www.minotaur.it
This email is made from 100% recycled electrons

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[asterisk-users] Asterisk on ClarkConnect

2008-01-16T17:34:00.000-08:00

Has anyone tried installing Asterisk on ClarkConnect? It looks like
ClarkConnect runs on RHEL so it should work if they haven't modified it too
much.

It appears that ClarkConnect is working on adding Asterisk and integrating
it into their GUI but until then I'd also be interested in trying to use
FreePBX.

Anyone?

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Re: [asterisk-users] Anyone Using a Dell PowerEdge T105 in Production

2008-01-16T17:28:00.000-08:00

On Jan 16, 2008 8:11 PM, Erik Anderson <erikerik@gmail.com> wrote:

On Jan 16, 2008 6:39 PM, Steve Totaro <stotaro@totarotechnologies.com> wrote:
> Unbeatable price for a low end Asterisk server (or any server for that
> matter)
>
> http://configure.us.dell.com/dellstore/config.aspx?c=us&cs=04&kc=6W300&l=en&oc=bednv4k&s=bsd
>
> I wonder if anyone has any experience with this box and Digium or Sangoma
> hardware? Any compatibility issues? If not, I might stock up on them.

Wow - that *is* a great price. I don't have any of this particular
box in production, but I do have 2 PowerEdge SC440s (one step up from
the T105) running asterisk along with Sangoma PRI cards. They're
working great. I really only have two issues with these low-end
servers:

1. You can't order 'em with RAID support. I'm getting around this by
using software RAID1 in linux, but I'd much prefer having a hardware
RAID controller.
2. The Dell DRAC remote management cards aren't compatible with these
low-end server motherboards. I've become *completely* addicted to the
DRAC cards on the high-end PowerEdges, to the point that I now refuse
to order a server without a DRAC card.

That said, I'm sure this server would run a small/medium asterisk
install just fine.

-Erik

You can add the raid option for $199. I think I might pickup about ten of them at this price. I can always resell them as general purpose servers or even workstations if Asterisk/Zaptel/Linux does not like the boxen.

Thanks,
Steve Totaro

Re: [asterisk-users] Anyone Using a Dell PowerEdge T105 in Production

2008-01-16T17:11:00.000-08:00

On Jan 16, 2008 6:39 PM, Steve Totaro <stotaro@totarotechnologies.com> wrote:
> Unbeatable price for a low end Asterisk server (or any server for that
> matter)
>
> http://configure.us.dell.com/dellstore/config.aspx?c=us&cs=04&kc=6W300&l=en&oc=bednv4k&s=bsd
>
> I wonder if anyone has any experience with this box and Digium or Sangoma
> hardware? Any compatibility issues? If not, I might stock up on them.

Wow - that *is* a great price. I don't have any of this particular
box in production, but I do have 2 PowerEdge SC440s (one step up from
the T105) running asterisk along with Sangoma PRI cards. They're
working great. I really only have two issues with these low-end
servers:

1. You can't order 'em with RAID support. I'm getting around this by
using software RAID1 in linux, but I'd much prefer having a hardware
RAID controller.
2. The Dell DRAC remote management cards aren't compatible with these
low-end server motherboards. I've become *completely* addicted to the
DRAC cards on the high-end PowerEdges, to the point that I now refuse
to order a server without a DRAC card.

That said, I'm sure this server would run a small/medium asterisk
install just fine.

-Erik

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[asterisk-users] Asterisk Now Beta 6 and CISCO IP 7910

2008-01-16T16:55:00.000-08:00

The phones are configured in the "Users" section of AsteriskGUI.

The bigger problem you'll have is that you probably also need to
replace/update the firmware on the 7910; by default they're configured to
work with Cisco's CallManager software. Start with this link:

http://www.voip-info.org/wiki/index.php?page=Asterisk+phone+cisco+79xx

Hope that helps. Good luck!

Jason Burbage

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[asterisk-users] IMAP client in asterisk not trying to contact IMAP server

2008-01-16T16:54:00.000-08:00

I'm trying to test IMAP in 1.4.17 and it appears to be not working.

I've compiled imap-2007 with the following on a CentOS 5 box:

make slx EXTRACFLAGS="-I/usr/include/openssl -fPIC"

and I've configured and compiled asterisk with the following:

./configure --with-imap=/usr/local/src/imap-2007

The compile and install went just fine, no warnings and no errors that I saw.

However, when actually trying to use it, it doesn't appear that asterisk is even
trying to use the local IMAP server.

The local IMAP server is dovecot, with a master password configured. I've
tried plain and SHA auth, but from the logs I don't even see the asterisk
master user trying to connect.

Here's my voicemail.conf:

[general]
imapserver=localhost
imapfolder=Inbox
;pollmailboxes=yes
;pollfreq=30
imapflags=notls
authuser=asttest
expungeonhangup=yes
authpassword=whatever
[default]

5252 => 5252,Test,5252@localhost,,imapuser=5252

(I have also tried this line as:
5252 => 5252,Test,,,imapuser=5252
5252 => 5252,Test,5252@localhost,,imapuser=5252|imappass=pass
5252 => 5252,Test,,,imapuser=5252|imappass=pass

all with and without the authuser and authpassword in the general section.)

I can authenticate against the * server using 5252*asttest as the username and
"whatever" as the password, which I'm lead to believe is how * will
try to connect.
(Also, the imap user 5252 exists and can receive mail.)

Is there something else I'm missing? Is there some other place in the
dial plan that
I have to say "use IMAP"? Is there some way to confirm that the imap client
has been compiled in? Some hidden CLI command to debug it?

doing "grep -i imap /var/log/asterisk/*" gives absolutely no results.

I'm almost convinced that I've got something wrong in the configuration because
I tried the latest SVN and I didn't see it hit the IMAP server, but it
also segfaulted
so who knows.

Any ideas at all? Am I missing something obvious that I'll find as
soon as I press
"send" and wish I hadn't sent the message?

Thanks,

--J(K)

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Re: [asterisk-users] HDLC errors

2008-01-16T16:44:00.000-08:00

Trixbox 2.2... I assume you are using the latest version. Normally I
will ignore messages from trixbox users because they ask kindergarten
stuff... but you seem to be knowledgeable and I'll assume you chose
trixbox to make your life easier when it comes to dealing with others
regarding the PBX.

I also assume the PRI is delivered via some sort of HDSL terminated at
an NIU ("SmartJack") Which is a box that will usually have 2 or 4
positions for line cards and 2 or 4 jacks marked "CPE1" etc....
usually at the bottom. Usually also you can look through the window at
the top and see various lights.

What is between the smartjack and your T1 card? What sort and length
of cable? Any splices? Punchdown or patch panels?

Also I'm not sure if Trixbox has this but ssh in and see if there is
an application called zttool. What are the statistics it is providing?

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[asterisk-users] Problem with a channel

2008-01-16T16:42:00.000-08:00

I have install a Server with Centos 1 TDM400: Asterisk 1.4.9, Zaptel 1.4.5

I having these problem :

Zap/2-1 is busy

Hangup ZAP/2-1

Everyone is busy/congested at this time (1:1/010)

Autofallthrough channel “SIP/202-b7b08ab0” Status is busy.

And then HANGUP.

[asterisk-users] Anyone Using a Dell PowerEdge T105 in Production

2008-01-16T16:39:00.000-08:00

Unbeatable price for a low end Asterisk server (or any server for that matter)

http://configure.us.dell.com/dellstore/config.aspx?c=us&cs=04&kc=6W300&l=en&oc=bednv4k&s=bsd

I wonder if anyone has any experience with this box and Digium or Sangoma hardware? Any compatibility issues? If not, I might stock up on them.

Thanks,
Steve Totaro

Re: [asterisk-users] HDLC errors

2008-01-16T16:32:00.000-08:00

On Jan 16, 2008 7:07 PM, Russell Bryant <russell@digium.com> wrote:

Steven wrote:
> I'm not sure what to try next, other than calling the telco and asking
> them to check their equipment. Does any one have a suggestion before I
> do that?

I have a suggestion. Have you contacted Digium technical support for assistance
with resolving this issue?

--
Russell Bryant
Senior Software Engineer
Open Source Team Lead
Digium, Inc.

Excellent suggestion. Make sure you can give them SSH access and screen so you can see what they are doing. Before that, check (remake) your T1 cables and if it is punched down on a block, re-punch it.

Work with your telco as well. I call that burning the candle from both ends. You said there were no errors looping port one to port two and generating calls with call files. That may indicate a telco issue. I usually open a ticket with the telco right away just in case so it can be escalated quicker if in fact it is the telco. Sometimes you have to be a jerk to these guys to get someone with half a brain to look into your problem rather than blaming CPE (the easiest way to close their ticket and get you off the phone).

If Digium says it is the telco and the telco says it is your CPE (Asterisk/Digium/Server/CPE wiring) then put them together on a conference call!

I am sure it won't come to that if it is truly a Digium/Asterisk issue. They will take care of it.

Thanks,
Steve Totaro

[asterisk-users] AddQueueMember and Flash Operator Panel

2008-01-16T16:11:00.000-08:00

Hello users!

Recently I read that AgentCallbackLogin is going to be deprecated soon.
Wanting to set up a few callback type queues, I set them up as suggested
in queues-with-callback-members.txt.

I was able to set the queues up completely this way, however, I'm trying
to use Flash Operator Panel (aka AsterNIC) to monitor the agents' login
status. FOP monitors their status if I call AddQueueMember with their
actual interface (which, by the way, makes more sense to me than logging
them in via chan_local), and it even seems to work with
Local/${AGENT_EXTEN}@default. But if I use any context other than
"default" here, FOP doesn't recognize that the agent is logged in.

(The users' default context isn't even set to default, and it behaves this
way even if the users' voicemail context is something else, so I am
guessing that is hard-coded in FOP somewhere.)

If I log them in from Local/${AGENT_EXTEN}@default, FOP works and the
agents get the calls, but then it's just dialing them directly - there is
no way to increment OUTBOUND_GROUP or check the value of GROUP_COUNT. As
a result, calls are routinely sent to agents who are already on the phone,
which I don't want.

Obviously, the next reasonable solution would be to use some other context
for the default context, and use [default] instead of [agents] for
incrementing OUTBOUND_GROUP and checking GROUP_COUNT, but I'm pretty sure
this would break the functionality of AsteriskGUI almost completely, and
I'm trying to preserve as much of that as possible.

Am I missing something? Is there a way to make all of this work together
without modifying some source code?

Thanks in advance!

Jason Burbage
jason@mhonetworks.com

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Re: [asterisk-users] HDLC errors

2008-01-16T16:07:00.000-08:00

Steven wrote:
> I'm not sure what to try next, other than calling the telco and asking
> them to check their equipment. Does any one have a suggestion before I
> do that?

I have a suggestion. Have you contacted Digium technical support for assistance
with resolving this issue?

--
Russell Bryant
Senior Software Engineer
Open Source Team Lead
Digium, Inc.

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[asterisk-users] HDLC errors

2008-01-16T15:52:00.000-08:00

I'm running Asterisk 1.2.26.1 svn rev 79171 on Trixbox 2.2. libpri
1.2.7 and zaptel 1.2.22.1. The hardware is a HP dl360 single cpu with a
TE220B. The system load is below 0.10.

I moved the server into production, with one PRI, on Friday. On that
day we handled a couple thousand calls and I only saw one HDLC abort
message. On Saturday half the calls and two abort messages an hour
apart. On Sunday, after 1500 when there was only a couple calls, the
HDLC messages went crazy.

We're getting non-stop Abort messages, with Bad FCS thrown in about
every tenth message. They come in bunches, with short 10-30 second
breaks. Then every once and awhile there is an 30 minute break,
sometimes a 3 hour break. The messages seems completely separate from
system load. The system will be idle and get the messages and have no
messages when I load up dozens of calls on it (using call files to
complete calls)

After reading the mailing list and various websites (asteriskguru.com
has a couple articles), the first thing I did was look for IRQ
conflicts. The module for the usb bus (no usb devices attached) was on
the same IRQ. Disabling USB had no effect. zttool shows no IRQ misses.

The second PRI was installed on Monday, that day with only two calls,
the message came 11 times. Three times on Tuesday with no calls, then
late at night I loaded it up with calls for testing (having call files
call out on the second PRI to the first PRI) and no messages were
generated. Again today its had a few messages with only a couple calls.

I'm not sure what to try next, other than calling the telco and asking
them to check their equipment. Does any one have a suggestion before I
do that?

Thanks.

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Re: [asterisk-users] [IAX] Up-to-date list of soft- and hardphones?

2008-01-16T14:45:00.000-08:00

----- Original Message -----
From: "Vincent" <vincent.delporte@bigfoot.com>
To: asterisk-users@lists.digium.com
Sent: 16 January 2008 22:01:55 o'clock (GMT) Europe/London
Subject: Re: [asterisk-users] [IAX] Up-to-date list of soft- and hardphones?

On Wed, 16 Jan 2008 18:08:23 +0000 (GMT), Gordon Henderson
<gordon+asterisk@drogon.net> wrote:
>However, you'll need to do similar things to your asterisk box & router if
>it's behind NAT for IAX as you do for SIP. (You will need a static IP
>address on the NAT router and port-forward 4569 to the asterisk box, just
>as you'd port-forward 5060 and 10000-20000 for SIP)

Am I wrong to understand that IAX only needs one port, TCP4569 by
default? So I only need one port for each phone, while SIP requires at
least 3 (SIP, and one RTP each way)?

--------

That's UDP 4569.
Also, depending on your configuration, you may not need to do any port forwarding
at the 'client' end. Just have all your phones send registrations frequently
and your natting router will do the rest.

Tim.

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Re: [asterisk-users] asterisk to mysql database!

2008-01-16T14:26:00.000-08:00

On Wednesday 16 January 2008 13:29:20 Simon Elliston Ball wrote:
> Simon Elliston Ball
> simon@simonellistonball.com
>
> On 16 Jan 2008, at 19:11, Naveen Palani wrote:
> > Hello,
> >
> > Is there a possibility to connect from asterisk to mysql database
> > without the interface application like Ruby or PHP.
> >
> > If i can connect to mysql database from asterisk, i can update the
> > database for manipulations.
>
> Try:
> http://www.voip-info.org/wiki/view/Mysql
>
> and the links thereon.

Or read configs/func_odbc.conf.sample.

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Re: [asterisk-users] Can DB() use SQLite instead of BerkeleyDB?

2008-01-16T14:22:00.000-08:00

On Wed, 16 Jan 2008 12:10:35 -0600, Tilghman Lesher
<tilghman@mail.jeffandtilghman.com> wrote:
>No, it cannot. You could use func_odbc to formulate your own queries,
>though.

Thanks. I don't like ODBC, but if it's stable and not a pain to
install/use, that could be the solution.

Otherwise, there's a new solution to use MySQL:

http://www.voip-info.org/wiki/view/Asterisk+cmd+MYSQL

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Re: [asterisk-users] [IAX] Up-to-date list of soft- and hardphones?

2008-01-16T14:01:00.000-08:00

Am I wrong to understand that IAX only needs one port, TCP4569 by
default? So I only need one port for each phone, while SIP requires at
least 3 (SIP, and one RTP each way)?

>And a SIP phone behind a NAT router is also solvable if it supports STUN.

But not all NAT routers support STUN, ie. keeping UDP ports open so
that incoming packets can make it.

>I know that SIP behind NAT isn't perfect, but with care, it's very usable
>and workable

But unless I'm mistaken, when NAT is involved, canreinvite must be set
to no, ie. all RTP packets must go through Asterisk instead of flowing
from one phone to the other?

Thanks guys.

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Re: [asterisk-users] Unable to open master device '/dev/zap/ctl'

2008-01-16T12:39:00.000-08:00

Make sure asterisk is in the "dialout" group in /etc/passwd

The default gentoo ebuild of zaptel creates /dev/zap/* with group dialout, and if you're using the gentoo ebuild of asterisk, it'll run as asterisk:asterisk, so you need to make sure asterisk is a member of the dialout goup otherwise it'll never be able to access /dev/zap/*

FWIW, as a fellow Gentoo user, 2006.1 is a bit dated, and you'd be well worth updating to 2007.0 if you can spare the time - it'll save you a lot of messing around with gcc versions etc. later down the line.

Regards,

Chris
--
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For full contact details visit http://www.minotaur.it
This email is made from 100% recycled electrons

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[asterisk-users] Asterisk 1.4.17 and RXFAX via T38

2008-01-16T12:24:00.000-08:00

I was pointed to the following:

http://asteriskforum.ru/viewtopic.php?t=1761

It is in Russian, which I don't speak, but it references an Asterisk patch.

Is this patch in 1.4.17?
Is it scheduled to be in 1.4.18 (or whatever ships after 1.4.17?)

Anyone work with this?

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Re: [asterisk-users] asterisk to mysql database!

2008-01-16T11:29:00.000-08:00

Try:
http://www.voip-info.org/wiki/view/Mysql

and the links thereon.

simon

Simon Elliston Ball
simon@simonellistonball.com

On 16 Jan 2008, at 19:11, Naveen Palani wrote:

> Hello,
>
> Is there a possibility to connect from asterisk to mysql database
> without the interface application like Ruby or PHP.
>
> If i can connect to mysql database from asterisk, i can update the
> database for manipulations.
>
> Appreciate your response.
> Regards,
> Naveen.Palani
>
>
> "Quinnox, a global IT services company prides itself on its SEI-CMM
> Level 5, ISO‑9001:2000 assessed delivery processes and provides
> solutions in areas of E-Business, ERP, Application Management
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> Quinnox Consultancy Services and/or any of its sister companies owns
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[asterisk-users] asterisk to mysql database!

2008-01-16T11:11:00.000-08:00

Hello,

Is there a possibility to connect from asterisk to mysql database without the interface application like Ruby or PHP.

If i can connect to mysql database from asterisk, i can update the database for manipulations.

Appreciate your response.

Regards,
Naveen.Palani

"Quinnox, a global IT services company prides itself on its SEI-CMM Level 5, ISO‑9001:2000 assessed delivery processes and provides solutions in areas of E-Business, ERP, Application Management Services, and EAI to customers in BFSI, Manufacturing, Retail, Telecom and Healthcare sector, powered by our Global Delivery Model."

This e-mail and any attached files are confidential, proprietary, and may also be legally privileged information, and are intended solely for the use of the individual or entity to whom they are addressed. If you are not the intended recipient of this e-mail, please send it back to the person who sent it to you and delete the e-mail and any attached files and destroy any copies of it; you may call us immediately at + 91 22 2829 0100 or email us at systems@quinnox.com

Quinnox Consultancy Services and/or any of its sister companies owns no responsibility for the views presented in the e-mail and any attached files unless the sender mentions so, with due authority of Quinnox Consultancy Services.

Unauthorized reading, reproduction, publication, use, dissemination, forwarding, printing or copying of this e-mail and its attachments is prohibited.
We have checked this message for any known viruses; however we decline any liability, in case of any damage caused by a non-detected virus.

For more details about our company, visit http://www.Quinnox.com

Re: [asterisk-users] Digium Part#'s (Was: Difference between TE121 and TE122)

2008-01-16T11:04:00.000-08:00

Dave Fullerton wrote:

> If you want to know what a card's capabilities are you're better off
> just memorizing each part number. Maybe there's a scheme I'm just not
> capable of understanding here.

We gave up (intentionally) on trying to have model numbers that
reflected all the capabilities of each card, because they would turn
into unintelligible (and unmemorizable) part numbers. We now have 'part
numbers' that represent a given card with the options it was ordered
with (analog module(s), echo canceler, etc.), and we've stopped trying
to use suffixes to indicate bus type and instead just use a different
model number.

This why the TE122 (which replaced the TE120P) no longer has a 'P'
suffix; the PCI-Express version is a different model number entirely.
With that said, for some reason our marketing department decided to
change the *prefix* for PCI-Express analog cards from TDM to AEX, but
they still follow the rest of the model naming scheme (no suffix letter
and no different model numbers that indicate included optional modules).

--
Kevin P. Fleming
Director of Software Technologies
Digium, Inc. - "The Genuine Asterisk Experience" (TM)

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Re: [asterisk-users] WARNING[31046]: chan_sip.c:4978 process_sdp: Unable to lookup host in c= line, 'IN IP4 100101'

2008-01-16T11:03:00.000-08:00

Did you look at the trace I send you in email? Because in each request
there are two IN IP lines I think Asterisk should only interpret the
first one,

On Jan 16, 2008 2:40 AM, Johansson Olle E <oej@edvina.net> wrote:
>
> 16 jan 2008 kl. 04.43 skrev Andrew Joakimsen:
>
> > Well can you offer some explanation why T.38 faxing worked for months
> > and then one day stopped working?
> You are asking the wrong forum. Your device is clearly sending a bad
> SDP. Ask the vendor of that device.
>
> /O
> >
> >
> > Using both Linksys & Audiocodes (yuck) ATA. The first second of the
> > fax tone is heard and then the T.38 switchover is attempted and the
> > call drops with said error.
> >
>
> >
> >
> > On Jan 15, 2008 6:25 PM, Mark Michelson <mmichelson@digium.com> wrote:
> >>
> >> Andrew Joakimsen wrote:
> >>> Anyone else have issues with T.38 where the call drops after T.38 is
> >>> attempted to be negotiated, with a message like the below?
> >>>
> >>> WARNING[31046]: chan_sip.c:4978 process_sdp: Unable to lookup host
> >>> in
> >>> c= line, 'IN IP4 100101'
> >>
> >> The problem is that 100101 is neither a valid IPv4 address nor a
> >> fully-qualified
> >> domain name.
> >>
> >> _______________________________________________
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> >> asterisk-users mailing list
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> >>

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> >>
> >
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>
> ---
> * Olle E Johansson - oej@edvina.net
> * Cell phone +46 70 593 68 51, Office +46 8 96 40 20, Sweden
>
>
>
>
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Re: [asterisk-users] Can DB() use SQLite instead of BerkeleyDB?

2008-01-16T10:10:00.000-08:00

On Wednesday 16 January 2008 10:02:12 Vincent wrote:
> Before I bother calling a PHP script through AGI just to read a number
> and rewrite the CID name... I was wondering if Asterisk could be
> configured so that DB() uses a SQL server instead of the usual
> BerkeleyDB?

No, it cannot. You could use func_odbc to formulate your own queries,
though.

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Re: [asterisk-users] [IAX] Up-to-date list of soft- and hardphones?

2008-01-16T10:08:00.000-08:00

On Wed, 16 Jan 2008, Vincent wrote:

> Hello
>
> There's a lot of information on VoIP at www.voip-info.org ...
> but there's also a lot of outdated information there as well :-/
>
> Since SIP is a pain to use when NAT is involved, especially when both
> the Asterisk server and the remote phones are behind NAT... I'd like
> to try IAX to see how it works and if it solves the issue.
>
> I'd like to start with a softphone (Windows only), and then, if tests
> prove successfully, buy a hardphone. What would be your
> recommendations?

IDEFISK or Zoiper as it's called now.

However, you'll need to do similar things to your asterisk box & router if
it's behind NAT for IAX as you do for SIP. (You will need a static IP
address on the NAT router and port-forward 4569 to the asterisk box, just
as you'd port-forward 5060 and 10000-20000 for SIP)

And a SIP phone behind a NAT router is also solvable if it supports STUN.

I know that SIP behind NAT isn't perfect, but with care, it's very usable
and workable. I have many installations doing just this, as I'm sure many
others on the list have too.

Gordon

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Re: [asterisk-users] [IAX] Up-to-date list of soft- and hardphones?

2008-01-16T09:43:00.000-08:00

On Wed, 16 Jan 2008, Vincent wrote:

Diax is probably the smallest Windows softphone.

I like my Digium Iaxy and a POTS instrument.

Thanks in advance,
------------------------------------------------------------------------
Steve Edwards sedwards@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000

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Re: [asterisk-users] Zap Issues

2008-01-16T09:32:00.000-08:00

Jeremy Mann wrote:
> Using Asterisk-1.4.17, Zaptel-1.4.8, libpri-1.4.3
>
> Upgraded this morning, now PRI channels are unstable as hell. After about 5 minutes all asterisk commands on the console refuse to respond, attached is the debug log right before and after the "lock-up", IT occurred between 9:18 and 9:20 AM at 9:20 I restarted asterisk.
>
> Box is debian w/ asterisk built from scratch.
>
> My setup is asterisk as a man-in-the-middle, Span 1 goes to Telco, Span 2 to Nortel MICS. PRI is not the problem as it's plugged into the Nortel directly for now and we have no problems.
>
> Nothing in dmesg indicates any errors.
>
> Any clue how I go about debugging this?

The best way is to start going through versions and figuring out which
version it broke at.

Some other things worth checking:

What versions of Zaptel/libpri/Asterisk did you upgrade from?

When you upgraded, did you recompile them in the correct order (Zaptel
1st, then libpri, then Asterisk)?

Matthew Fredrickson

>
> ----
>
> [Jan 16 09:18:41] DEBUG[10183] chan_zap.c: Unlinking slave 1 from 47
> [Jan 16 09:18:41] DEBUG[10183] chan_zap.c: Removed 12 from conference 9/47
> [Jan 16 09:18:41] DEBUG[10183] chan_zap.c: Removed 57 from conference 9/1
> [Jan 16 09:18:41] DEBUG[10183] chan_zap.c: Set option AUDIO MODE, value: ON(1) on Zap/1-1
> [Jan 16 09:18:56] DEBUG[10107] chan_zap.c: Unlinking slave 26 from 3
> [Jan 16 09:18:56] DEBUG[10107] chan_zap.c: Removed 36 from conference 9/3
> [Jan 16 09:18:56] DEBUG[10107] chan_zap.c: Removed 14 from conference 9/26
> [Jan 16 09:18:56] DEBUG[10107] chan_zap.c: Set option AUDIO MODE, value: ON(1) on Zap/26-1
> [Jan 16 09:18:56] DEBUG[10107] chan_zap.c: Not yet hungup... Calling hangup once with icause, and clearing call
> [Jan 16 09:18:56] DEBUG[10107] chan_zap.c: Set option AUDIO MODE, value: OFF(0) on Zap/26-1
> [Jan 16 09:18:56] DEBUG[10107] chan_zap.c: Set option AUDIO MODE, value: ON(1) on Zap/3-1
> [Jan 16 09:20:24] DEBUG[8430] chan_zap.c: Ring requested on channel 0/23 already in use or previously requested on span 2. Attempting to renegotiating chann
> el.
> [Jan 16 09:20:24] DEBUG[8430] chan_zap.c: Found empty available channel 0/21
> [Jan 16 09:22:24] DEBUG[8430] chan_zap.c: Ring requested on channel 0/23 already in use or previously requested on span 2. Attempting to renegotiating chann
> el.
> [Jan 16 09:22:24] DEBUG[8430] chan_zap.c: Found empty available channel 0/20
> [Jan 16 09:22:31] DEBUG[8430] chan_zap.c: Ring requested on channel 0/23 already in use or previously requested on span 2. Attempting to renegotiating chann
> el.
> [Jan 16 09:22:31] DEBUG[8430] chan_zap.c: Found empty available channel 0/19
> [Jan 16 09:23:07] DEBUG[8430] chan_zap.c: Ring requested on channel 0/23 already in use or previously requested on span 2. Attempting to renegotiating chann
> el.
> [Jan 16 09:23:07] DEBUG[8430] chan_zap.c: Found empty available channel 0/18
>
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[asterisk-users] [IAX] Up-to-date list of soft- and hardphones?

2008-01-16T09:25:00.000-08:00

Hello

There's a lot of information on VoIP at www.voip-info.org ...
but there's also a lot of outdated information there as well :-/

Since SIP is a pain to use when NAT is involved, especially when both
the Asterisk server and the remote phones are behind NAT... I'd like
to try IAX to see how it works and if it solves the issue.

I'd like to start with a softphone (Windows only), and then, if tests
prove successfully, buy a hardphone. What would be your
recommendations?

Thank you.

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Re: [asterisk-users] IAX Trunk between two Asterisks: Authority, and Call Rejected

2008-01-16T08:35:00.000-08:00

On Jan 16, 2008 8:46 AM, bilal ghayyad <bilmar_gh@yahoo.com> wrote:

Hi All;

I did an IP Trunk using IAX between two Asterisk
boxes, now Asterisk A can send a call for B but B
refuse it. The IAX type was configured to be "friend"
in the iax.con for Asterisk A and B, is there any
thing else need to be done to let B accept the call
from A?

Also, I used an static IP address for the host when I
configured the iax client in the iax.conf file.

Any help?
Regards
Bilal

I used to see this problem when I used to use IAX2. Sometimes it would just go away. I seem to remember using insecure=very to get it working but I may be wrong.

Anyways, post the relevant parts of your IAX2 confs from both boxes and someone might be able to spot something right off the bat.

Thanks,
Steve Totaro

Re: [asterisk-users] Zap Issues

2008-01-16T08:25:00.000-08:00

On Jan 16, 2008 10:25 AM, Jeremy Mann <jmann@txhmg.com> wrote:

Using Asterisk-1.4.17, Zaptel-1.4.8, libpri-1.4.3

Upgraded this morning, now PRI channels are unstable as hell. After about 5 minutes all asterisk commands on the console refuse to respond, attached is the debug log right before and after the "lock-up", IT occurred between 9:18 and 9:20 AM at 9:20 I restarted asterisk.

Box is debian w/ asterisk built from scratch.

My setup is asterisk as a man-in-the-middle, Span 1 goes to Telco, Span 2 to Nortel MICS. PRI is not the problem as it's plugged into the Nortel directly for now and we have no problems.

Nothing in dmesg indicates any errors.

Any clue how I go about debugging this?

----

[Jan 16 09:18:41] DEBUG[10183] chan_zap.c: Unlinking slave 1 from 47

[Jan 16 09:18:41] DEBUG[10183] chan_zap.c: Removed 12 from conference 9/47

[Jan 16 09:18:41] DEBUG[10183] chan_zap.c: Removed 57 from conference 9/1

[Jan 16 09:18:41] DEBUG[10183] chan_zap.c: Set option AUDIO MODE, value: ON(1) on Zap/1-1

[Jan 16 09:18:56] DEBUG[10107] chan_zap.c: Unlinking slave 26 from 3

[Jan 16 09:18:56] DEBUG[10107] chan_zap.c: Removed 36 from conference 9/3

[Jan 16 09:18:56] DEBUG[10107] chan_zap.c: Removed 14 from conference 9/26

[Jan 16 09:18:56] DEBUG[10107] chan_zap.c: Set option AUDIO MODE, value: ON(1) on Zap/26-1

[Jan 16 09:18:56] DEBUG[10107] chan_zap.c: Not yet hungup... Calling hangup once with icause, and clearing call

[Jan 16 09:18:56] DEBUG[10107] chan_zap.c: Set option AUDIO MODE, value: OFF(0) on Zap/26-1

[Jan 16 09:18:56] DEBUG[10107] chan_zap.c: Set option AUDIO MODE, value: ON(1) on Zap/3-1

[Jan 16 09:20:24] DEBUG[8430] chan_zap.c: Ring requested on channel 0/23 already in use or previously requested on span 2. Attempting to renegotiating chann

el.

[Jan 16 09:20:24] DEBUG[8430] chan_zap.c: Found empty available channel 0/21

[Jan 16 09:22:24] DEBUG[8430] chan_zap.c: Ring requested on channel 0/23 already in use or previously requested on span 2. Attempting to renegotiating chann

el.

[Jan 16 09:22:24] DEBUG[8430] chan_zap.c: Found empty available channel 0/20

[Jan 16 09:22:31] DEBUG[8430] chan_zap.c: Ring requested on channel 0/23 already in use or previously requested on span 2. Attempting to renegotiating chann

el.

[Jan 16 09:22:31] DEBUG[8430] chan_zap.c: Found empty available channel 0/19

[Jan 16 09:23:07] DEBUG[8430] chan_zap.c: Ring requested on channel 0/23 already in use or previously requested on span 2. Attempting to renegotiating chann

el.

[Jan 16 09:23:07] DEBUG[8430] chan_zap.c: Found empty available channel 0/1

Sorry I cannot be of help with your issue since I stick with 1.2.x until I stop seeing my bugs folder filling up daily and seeing posts like this and the very long running post about "Unstable Releases".

I would suggest rolling back to whatever version worked and opening an issue on bugtracker.

I am curious though, why did you upgrade? Were you having problems with the version you upgraded from?

Thanks,
Steve Totaro

[asterisk-users] Can DB() use SQLite instead of BerkeleyDB?

2008-01-16T08:02:00.000-08:00

Hello

Before I bother calling a PHP script through AGI just to read a number
and rewrite the CID name... I was wondering if Asterisk could be
configured so that DB() uses a SQL server instead of the usual
BerkeleyDB?

============
;rewrite CIDNAME if found in DB

exten => cid,1,Set(CALLERIDNAME=${IF($["${CALLERID(name)}"=""]?<No CID
name>:${CALLERID(name)})})

exten =>
cid,n,Set(CALLERIDNAME=${IF($["${DB(cidname/${CALLERIDNUM})}"=""]?${CALLERIDNAME}:${DB(cidname/${CALLERIDNUM})})})
============

Thank you.

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Re: [asterisk-users] Digium Part#'s (Was: Difference between TE121 and TE122)

2008-01-16T07:30:00.000-08:00

Igor A. Goncharovsky wrote:
> Hello!
>
> Guilherme Loch Waltrick Góes wrote:
>> What's the difference between the TE121 and TE122. I read the description on
>> Digium's site and it isn't clear to me.
>> Best regards,
>>
> The only one difference is interface: one of them have PCI and other
> have PCI-Express.
>

Here's my rant for the day:

The one thing that's bugged me about Digium's model numbers is the
difficulty in determining features just by looking at them. The one
thing that is easy to tell is the maximum number of ports/channels the
card will take. From there it gets harder. Analog is either TDM* or
AEX*. PCI or PCI express can sort of be determined by whether it ends
with a P except on the TE122?. Whether hardware echo cancellation is
built-in, available or not present can sort of be determined for digital
cards if it ends in 2 or 7 except on one port models. PCI card voltage?
Again, sometimes you can use the last digit but not on one port models.
If you want to know what a card's capabilities are you're better off
just memorizing each part number. Maybe there's a scheme I'm just not
capable of understanding here.

Rant off.

-Dave

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Re: [asterisk-users] Does host accept dns or ddns?

2008-01-16T07:27:00.000-08:00

Yes, I use it and got no problems.

On Jan 16, 2008 11:47 AM, bilal ghayyad <bilmar_gh@yahoo.com> wrote:

Hi All;

Did anyone tried to use dns name or ddns name with
host (host=abc.www.com) and it worked fine?

Regards
Bilal

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Re: [asterisk-users] cisco ip phne 7911G with asterisk

2008-01-16T07:26:00.000-08:00

Now that you have your 7911g phone up running, would you mind checking
the audio quality when leaving a voicemail for on another local asterisk
user from this phone? I have a 7911g and I hear loud audio taps from the
phone. The 7961g phone doesn't have this issue. I'm just trying to
rule out the phone.
Thanks

-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Christian
Pinedo Zamalloa
Sent: Wednesday, January 16, 2008 10:16 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] cisco ip phne 7911G with asterisk

On Tue, Jan 15, 2008 at 01:14:42PM +0000, Christian Pinedo wrote:
> hi,
>
> I'm trying to configure a Cisco IP Phone 7911G in order to work with
Asterisk. I have loaded the 8.3.3 SIP Firmware of Cisco through a DHCP
and a TFTP server. All seems ok but a file that is downloaded :
term06.default.loads (I understand that is for 7906 model) instead of
term11.default.loads (I understand that is for 7911 model). In any case
the phone reboots well.
>
> At this moment I thought that the phone should ask the
SEP<mac>.xml.cnf file but it asks CTLSEP<mac>.tlv all the time. I don't
have this file in the server and it tries to download every few seconds
whitout asking another file. According to what I have read this file
shouldn't be neccesary and, when the phone cann't obtain it, the phone
should ask SEP<mac>.xml.cnf. I don't know if I'm doing something bad or
if it could be a issue of the firmware version.
>
> I would thank some clue. Thanks,
>

It was a TFTP server issue. The classical TFTP server used in the unix
world responds to queries with bad error codes. I finally
used aTFTPD that does this well so the phone understands that there's no
CTLSEP file and then asks for SEP file.

--
Christian Pinedo Zamalloa (zako)
PGP key at: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x828D0C80
Fingerprint: 7BFF 4105 F46B 7977 BD96 348C 1007 4FF8 828D 0C80

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[asterisk-users] Zap Issues

2008-01-16T07:25:00.000-08:00

Using Asterisk-1.4.17, Zaptel-1.4.8, libpri-1.4.3

Upgraded this morning, now PRI channels are unstable as hell. After about 5 minutes all asterisk commands on the console refuse to respond, attached is the debug log right before and after the “lock-up”, IT occurred between 9:18 and 9:20 AM at 9:20 I restarted asterisk.

Box is debian w/ asterisk built from scratch.

My setup is asterisk as a man-in-the-middle, Span 1 goes to Telco, Span 2 to Nortel MICS. PRI is not the problem as it’s plugged into the Nortel directly for now and we have no problems.

Nothing in dmesg indicates any errors.

Any clue how I go about debugging this?

----

[Jan 16 09:18:41] DEBUG[10183] chan_zap.c: Unlinking slave 1 from 47

[Jan 16 09:18:41] DEBUG[10183] chan_zap.c: Removed 12 from conference 9/47

[Jan 16 09:18:41] DEBUG[10183] chan_zap.c: Removed 57 from conference 9/1

[Jan 16 09:18:41] DEBUG[10183] chan_zap.c: Set option AUDIO MODE, value: ON(1) on Zap/1-1

[Jan 16 09:18:56] DEBUG[10107] chan_zap.c: Unlinking slave 26 from 3

[Jan 16 09:18:56] DEBUG[10107] chan_zap.c: Removed 36 from conference 9/3

[Jan 16 09:18:56] DEBUG[10107] chan_zap.c: Removed 14 from conference 9/26

[Jan 16 09:18:56] DEBUG[10107] chan_zap.c: Set option AUDIO MODE, value: ON(1) on Zap/26-1

[Jan 16 09:18:56] DEBUG[10107] chan_zap.c: Not yet hungup... Calling hangup once with icause, and clearing call

[Jan 16 09:18:56] DEBUG[10107] chan_zap.c: Set option AUDIO MODE, value: OFF(0) on Zap/26-1

[Jan 16 09:18:56] DEBUG[10107] chan_zap.c: Set option AUDIO MODE, value: ON(1) on Zap/3-1

[Jan 16 09:20:24] DEBUG[8430] chan_zap.c: Ring requested on channel 0/23 already in use or previously requested on span 2. Attempting to renegotiating chann

el.

[Jan 16 09:20:24] DEBUG[8430] chan_zap.c: Found empty available channel 0/21

[Jan 16 09:22:24] DEBUG[8430] chan_zap.c: Ring requested on channel 0/23 already in use or previously requested on span 2. Attempting to renegotiating chann

el.

[Jan 16 09:22:24] DEBUG[8430] chan_zap.c: Found empty available channel 0/20

[Jan 16 09:22:31] DEBUG[8430] chan_zap.c: Ring requested on channel 0/23 already in use or previously requested on span 2. Attempting to renegotiating chann

el.

[Jan 16 09:22:31] DEBUG[8430] chan_zap.c: Found empty available channel 0/19

[Jan 16 09:23:07] DEBUG[8430] chan_zap.c: Ring requested on channel 0/23 already in use or previously requested on span 2. Attempting to renegotiating chann

el.

[Jan 16 09:23:07] DEBUG[8430] chan_zap.c: Found empty available channel 0/18

Re: [asterisk-users] Attended transfers manager or phone

2008-01-16T07:24:00.000-08:00

Thank you very much, that was a new angle I hadn't thought of time to
investigate a little more :). The joys of learning new things :)

- Christian

> -----Original Message-----
> From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-
> bounces@lists.digium.com] On Behalf Of Mojo with Horan & Company, LLC
> Sent: 16. januar 2008 01:06
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Attended transfers manager or phone
>
> Some phones have the auto-answer ability. So your phone could have two
> extensions, one for normal use and one for auto-answer use. Redirect or
> Originate, as you were, to the auto-answer extension on the phone. So
> the phone would already put itself offhook, and asterisk would continue
> and build up the other end of the bridge.
>
> Polycom soundpoint phones, for example, but many others have this ability.
>
> an example extension setup might be
>
> exten => 110,1,Dial(SIP/110)
>
> exten => #110,1,SipAddHeader(.......whatever your phone needs to make it
> autoanswer)
> exten => #110,2,Dial(SIP/110)
>
> Don't know about phones that allow ip control of their state, though.
>
> Moj
>
> Christian Ejlertsen wrote:
> > Well I'm sure this issue has been bean up a few time since it's one of
> the
> > only ones I can't find a real "simple" answer to.
> >
> > I'm trying to find away to do attended transfers through the manager
> > interface, for a pc switchboard / Agent client solution, but so far
> coming
> > up short.
> > The action Originate is part of the solution, but what really I want is
> the
> > phone being taken off-hook and then being able to dial the number
> without
> > having to answer the dial-back first.
> >
> > 1. One solution, though an ugly one, would be using Originate, but use a
> > phone that has some sort tcp/ip interface that allows for taking the
> phone
> > off-hook.
> >
> > 2. A Better solution would be using a phone that allows dialling and
> taking
> > the phone off-hook on-hook etc. via some tcp/ip interface.
> >
> > 3. Yet another solution, though I do not favour this one since I really
> > don't want to maintain the sip phone code, would be programming a soft
> sip
> > phone with all the bells and whistles and adding the switchboard
> > functionality to that (name searching, status email so on and so forth.
> >
> > In the end all I need is just a software or hardware phone, sip/iax,
> which
> > can be told via tcp/ip to go off-hook, on-hook, dial, transfer and
> perhaps
> > status requests. If such a phone exists that would do the trick, the
> rest is
> > manageable via the Asterisk Manager console.
> >
> > I'm guessing some people have messed with this problem before so I hope
> that
> > someone has some information about this kind of thing :)
> >
> > Thank you in advance
> > Christian
> >
> >
> > _______________________________________________
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> >
> > asterisk-users mailing list
> > To UNSUBSCRIBE or update options visit:
> >

http://lists.digium.com/mailman/listinfo/asterisk-users

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Re: [asterisk-users] Voicemail consultation problem

2008-01-16T07:22:00.000-08:00

I would suppose that the time on the asterisk system is not the time
that he is using. Other than that, you should really be collecting logs.

David Florella wrote:
> Hello,
>
> A user who uses my Asterisk made me part of a worry about listening to
> his voicemails. He has received 4 voicemails on January 3,
> respectively at 3H00 pm, 3H36 pm, 3H41 pm and 4H40 pm. He has received
> notifications by e-mail at these times.
> On first listen to his messages, at 8.00 pm, Asterisk has announced
> two new voicemails(15H00 and 15H36). He has erased thos voicemails.
> At 8.30pm , he has called again the Asterisk voicemail. Asterisk
> announced him two messages (15H41 and 16H40).
>
> I don't have any Asterisk logs .
>
> A person have an idea of what may have caused the fact that my user
> did not, in the first call, heard his 4 messages?
>
> Thank you.
>
> ------------------------------------------------------------------------
>
> _______________________________________________
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>

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--
Thank you and have a wonderful day,

Anthony Francis
Rockynet VOIP
(303) 444-7052 opt 2
voip@rockynet.com

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Re: [asterisk-users] cisco ip phne 7911G with asterisk

2008-01-16T07:16:00.000-08:00

On Tue, Jan 15, 2008 at 01:14:42PM +0000, Christian Pinedo wrote:
> hi,
>
> I'm trying to configure a Cisco IP Phone 7911G in order to work with Asterisk. I have loaded the 8.3.3 SIP Firmware of Cisco through a DHCP and a TFTP server. All seems ok but a file that is downloaded : term06.default.loads (I understand that is for 7906 model) instead of term11.default.loads (I understand that is for 7911 model). In any case the phone reboots well.
>
> At this moment I thought that the phone should ask the SEP<mac>.xml.cnf file but it asks CTLSEP<mac>.tlv all the time. I don't have this file in the server and it tries to download every few seconds whitout asking another file. According to what I have read this file shouldn't be neccesary and, when the phone cann't obtain it, the phone should ask SEP<mac>.xml.cnf. I don't know if I'm doing something bad or if it could be a issue of the firmware version.
>
> I would thank some clue. Thanks,
>

--
Christian Pinedo Zamalloa (zako)
PGP key at: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x828D0C80
Fingerprint: 7BFF 4105 F46B 7977 BD96 348C 1007 4FF8 828D 0C80

Re: [asterisk-users] Backup Route

2008-01-16T07:02:00.000-08:00

Change the priority of the second dial() to 4.

Regards,

On Jan 16, 2008 11:42 AM, Abdul <abdul_zu@yahoo.com > wrote:

Good Day All,

Is it possible to put backup route in asterisk dial plan? fro the example if the first carrier disconnect the call with Congestion or Circuit busy then asterisk can dial another carrier?

I did the following but it is not working as i need to dial the second one only on congestions or circuit busy.

[wellsip]
exten => _x.,1,AGI(routing.pl)
exten => _x.,2,Set(TIMEOUT(absolute)=${TMO})
exten => _x.,3,Dial(SIP/${CNUM}@${CAIP})
exten => _x.,3,Dial(SIP/${CNUM}@${CAIP2})
exten => h,1,DeadAGI(stop.pl)

--------
Regard,

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Notícias e Fórum sobre VoIP com software livre: http://www.asteriskexperts.com.br

[asterisk-users] Voicemail consultation problem

2008-01-16T06:42:00.000-08:00

Hello,

A user who uses my Asterisk made me part of a worry about listening to his voicemails. He has received 4 voicemails on January 3, respectively at 3H00 pm, 3H36 pm, 3H41 pm and 4H40 pm. He has received notifications by e-mail at these times.
On first listen to his messages, at 8.00 pm, Asterisk has announced two new voicemails(15H00 and 15H36). He has erased thos voicemails.
At 8.30pm , he has called again the Asterisk voicemail. Asterisk announced him two messages (15H41 and 16H40).

I don't have any Asterisk logs .

A person have an idea of what may have caused the fact that my user did not, in the first call, heard his 4 messages?

Thank you.

Re: [asterisk-users] Attended transfers manager or phone

2008-01-15T16:06:00.000-08:00

Some phones have the auto-answer ability. So your phone could have two
extensions, one for normal use and one for auto-answer use. Redirect or
Originate, as you were, to the auto-answer extension on the phone. So
the phone would already put itself offhook, and asterisk would continue
and build up the other end of the bridge.

Polycom soundpoint phones, for example, but many others have this ability.

an example extension setup might be

exten => 110,1,Dial(SIP/110)

exten => #110,1,SipAddHeader(.......whatever your phone needs to make it
autoanswer)
exten => #110,2,Dial(SIP/110)

Don't know about phones that allow ip control of their state, though.

Moj

Christian Ejlertsen wrote:
> Well I'm sure this issue has been bean up a few time since it's one of the
> only ones I can't find a real "simple" answer to.
>
> I'm trying to find away to do attended transfers through the manager
> interface, for a pc switchboard / Agent client solution, but so far coming
> up short.
> The action Originate is part of the solution, but what really I want is the
> phone being taken off-hook and then being able to dial the number without
> having to answer the dial-back first.
>
> 1. One solution, though an ugly one, would be using Originate, but use a
> phone that has some sort tcp/ip interface that allows for taking the phone
> off-hook.
>
> 2. A Better solution would be using a phone that allows dialling and taking
> the phone off-hook on-hook etc. via some tcp/ip interface.
>
> 3. Yet another solution, though I do not favour this one since I really
> don't want to maintain the sip phone code, would be programming a soft sip
> phone with all the bells and whistles and adding the switchboard
> functionality to that (name searching, status email so on and so forth.
>
> In the end all I need is just a software or hardware phone, sip/iax, which
> can be told via tcp/ip to go off-hook, on-hook, dial, transfer and perhaps
> status requests. If such a phone exists that would do the trick, the rest is
> manageable via the Asterisk Manager console.
>
> I'm guessing some people have messed with this problem before so I hope that
> someone has some information about this kind of thing :)
>
> Thank you in advance
> Christian
>
>
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>

http://lists.digium.com/mailman/listinfo/asterisk-users
>

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Re: [asterisk-users] CID blocking ...

2008-01-15T15:31:00.000-08:00

On Tue, 2008-01-15 at 17:44 -0500, Andrew Joakimsen wrote:
> On Jan 14, 2008 6:29 PM, Paul Hales <pdhales@optusnet.com.au> wrote:
> >
> > The 'setcallerpres' application is the one to use...
> >
>
> Only works for PRI channels (maybe plain T1) channels via Zaptel.
>

Agreed entirely.

PaulH

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Re: [asterisk-users] Channel fallback

2008-01-15T15:30:00.000-08:00

Use the chanisavail to check that the SIP channels are clear, and set
reasonable 'qualify=' settings for them....

PaulH

On Tue, 2008-01-15 at 23:20 +0100, Jaap Winius wrote:
> Hi list,
>
> My Asterisk v1.4 system now has two ISDN channels and two SIP
> channels. The idea is to make a dialplan that mostly uses the SIP
> channels for outgoing calls, but I'd like those to fall back
> automatically to ISDN if the SIP channels aren't available, possibly
> in combination with a warning issued to the caller before the call is
> actually placed.
>
> Is this possible with Asterisk? If so, how?
>
> Thanks,
>
> Jaap
>
> _______________________________________________
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>
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>

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Re: [asterisk-users] WARNING[31046]: chan_sip.c:4978 process_sdp: Unable to lookup host in c= line, 'IN IP4 100101'

2008-01-15T15:25:00.000-08:00

Andrew Joakimsen wrote:
> Anyone else have issues with T.38 where the call drops after T.38 is
> attempted to be negotiated, with a message like the below?
>
> WARNING[31046]: chan_sip.c:4978 process_sdp: Unable to lookup host in
> c= line, 'IN IP4 100101'

The problem is that 100101 is neither a valid IPv4 address nor a fully-qualified
domain name.

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Re: [asterisk-users] Discover Asterisk 1.4 :: SIP Subscriptions

2008-01-15T15:03:00.000-08:00

On Tuesday 15 January 2008 16:03:16 Andrea Spadaccini wrote:
> Russell wrote:
> > Andrea wrote:
> > > I have a small question: other than a phone (ie. SIP/something), what
> > > else can I use as "app"? Can I handle the change via some custom code?
> >
> > There are a number of things that can provide "device state" in Asterisk.
> > That includes "real" devices such as SIP endpoints, or any other channel
> > driver. However, it also includes things like monitoring the state of a
> > space in parking, or the usage of a MeetMe conference. I have also
> > written a small dialplan function which lets you create custom device
> > states. A lot of people use this for things like having a light on the
> > phone that reflects whether the agent is logged in or not.
> >
> > More information:
> >
> > http://asterisk.org/node/48325
> > http://asterisk.org/node/48360
>
> Thanks a lot for the info, I already read the first article, and it's great
> to know that DEVSTATE can be used in 1.4.
>
> But my question was different, my poor english doesn't help me. :(
>
> In your article I read
> "For example, when someone subscribes to the state of extension
> 1234, Asterisk knows to give them the state of the SIP phone SIP/myphone.
>
> exten => 1234,hint,SIP/myphone"
>
> Suppose that I want to write to a database the state of all my extensions,
> in order to display it in a web page.
>
> How could I do it using the hint mechanism?

Just create a module that subscribes to every single device and when the state
changes, your callback will get an event with the device name that changed.
You could then update your database with an SQL query (or whatever else you
like).

--
Tilghman

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Re: [asterisk-users] CID blocking ...

2008-01-15T14:44:00.000-08:00

On Jan 14, 2008 6:29 PM, Paul Hales <pdhales@optusnet.com.au> wrote:
>
> The 'setcallerpres' application is the one to use...
>

Only works for PRI channels (maybe plain T1) channels via Zaptel.

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Re: [asterisk-users] Console app

2008-01-15T14:25:00.000-08:00

What does 'make menuselect' let you choose? Under #3, Channel Driveers,
does chan_alsa have XXX through it so you can't select it? does
chan_oss have XXX? This would indicate to you that the pieces of alsa or
oss asterisk would need are not installed properly.

Moj

Gilberto Nunes wrote:
> Hi all
>
> I build an Asterisk, with asterisk 1.4.16.1 source.
> I have notice, that the console app don't appear on CLI...
>
> Is theres some options to turn on, when I compile asterisk?
>
> Thanks...
>
>
>

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[asterisk-users] Channel fallback

2008-01-15T14:20:00.000-08:00

Hi list,

My Asterisk v1.4 system now has two ISDN channels and two SIP
channels. The idea is to make a dialplan that mostly uses the SIP
channels for outgoing calls, but I'd like those to fall back
automatically to ISDN if the SIP channels aren't available, possibly
in combination with a warning issued to the caller before the call is
actually placed.

Is this possible with Asterisk? If so, how?

Thanks,

Jaap

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[asterisk-users] chan_mobile type=

2008-01-15T14:19:00.000-08:00

What are the values for type for chan_mobile?

headset and phone ???

I get my Treo650 to pair.

hcitool scan shows the device.
hcitool con comes up empty.

I go into Asterisk cli.

mobile search shows the device (while I am waiting for a response, I see
the phone showing a connection being set up). And I am shown that I
have a device that is:

NOT available and is a headset

OOPS.

So I think I need to force Asterisk to see this as a phone? Or is there
something I need in a bluetooth config file?

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Re: [asterisk-users] Discover Asterisk 1.4 :: SIP Subscriptions

2008-01-15T14:03:00.000-08:00

Ciao Russell,

> > I have a small question: other than a phone (ie. SIP/something), what else
> > can I use as "app"? Can I handle the change via some custom code?
>
> There are a number of things that can provide "device state" in Asterisk.
> That includes "real" devices such as SIP endpoints, or any other channel
> driver. However, it also includes things like monitoring the state of a space
> in parking, or the usage of a MeetMe conference. I have also written a small
> dialplan function which lets you create custom device states. A lot of
> people use this for things like having a light on the phone that reflects
> whether the agent is logged in or not.
>
> More information:
>
> http://asterisk.org/node/48325
> http://asterisk.org/node/48360

Thanks a lot for the info, I already read the first article, and it's great to
know that DEVSTATE can be used in 1.4.

But my question was different, my poor english doesn't help me. :(

In your article I read
"For example, when someone subscribes to the state of extension
1234, Asterisk knows to give them the state of the SIP phone SIP/myphone.

exten => 1234,hint,SIP/myphone"

Suppose that I want to write to a database the state of all my extensions, in
order to display it in a web page.

How could I do it using the hint mechanism?

Thanks again,

--
Dott. Andrea Spadaccini
Multimedia Technologies Institute s.r.l.

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[asterisk-users] WARNING[31046]: chan_sip.c:4978 process_sdp: Unable to lookup host in c= line, 'IN IP4 100101'

2008-01-15T13:54:00.000-08:00

Anyone else have issues with T.38 where the call drops after T.38 is
attempted to be negotiated, with a message like the below?

WARNING[31046]: chan_sip.c:4978 process_sdp: Unable to lookup host in
c= line, 'IN IP4 100101'

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Re: [asterisk-users] Discover Asterisk 1.4 :: SIP Subscriptions

2008-01-15T13:23:00.000-08:00

Patrick wrote:
> Nice one Olle. Before I possibly waste my time trying this does this
> blinkety lights magic also work with SCCP phones?

IIRC, this feature is currently only supported in Asterisk trunk (soon to become
Asterisk 1.6).

--
Russell Bryant
Senior Software Engineer
Open Source Team Lead
Digium, Inc.

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Re: [asterisk-users] Discover Asterisk 1.4 :: SIP Subscriptions

2008-01-15T13:22:00.000-08:00

Andrea Spadaccini wrote:
> I have a small question: other than a phone (ie. SIP/something), what else can
> I use as "app"? Can I handle the change via some custom code?

There are a number of things that can provide "device state" in Asterisk. That
includes "real" devices such as SIP endpoints, or any other channel driver.
However, it also includes things like monitoring the state of a space in
parking, or the usage of a MeetMe conference. I have also written a small
dialplan function which lets you create custom device states. A lot of people
use this for things like having a light on the phone that reflects whether the
agent is logged in or not.

More information:

http://asterisk.org/node/48325
http://asterisk.org/node/48360

--
Russell Bryant
Senior Software Engineer
Open Source Team Lead
Digium, Inc.

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