Thursday, January 17, 2008

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

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

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

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

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
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>

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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

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

--
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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
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Phone: +43 699 116 07 923
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Web: http://www.kamleitner.com

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

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.

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
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>

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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] Polycom Remotely Cancel Call Forward

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.

--
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http://www.btwtech.com/


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

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

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

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

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

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

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

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?

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

Re: [asterisk-users] SIP Proxy Issues



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>
 
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

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

Thanks,
Steve Totaro

[asterisk-users] PostgreSQL query results truncated 255 characters

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

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>
 
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

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

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

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?



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?

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?

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?

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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[asterisk-users] Voicemail Callback

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

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
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>

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

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

> -----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

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

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

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?

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

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?

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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>>>
>>>
>>>
>> _______________________________________________
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>> asterisk-users mailing list
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>>

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>>
>>
>
>
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> asterisk-users mailing list
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>

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>
>


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Re: [asterisk-users] AEL includes?

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 --
> >>
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> >>

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> >>
> >>
> >
> >
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> >
>
>
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--
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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

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?

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
>>
>> _______________________________________________
>> -- 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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>

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Wednesday, January 16, 2008

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

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

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://tools.search.yahoo.com/newsearch/category.php?category=shopping
>
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>

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

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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Find them fast with Yahoo! Search.   http://tools.search.yahoo.com/newsearch/category.php?category=shopping

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

> 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

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

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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>
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> To UNSUBSCRIBE or update options visit:
>

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>

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

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?

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

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'

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

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



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

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

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

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

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

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.