Re: [asterisk-users] Selecting a specific line from Zap/g

ignorpat is your friend

On 9/30/07, Tzafrir Cohen <tzafrir.cohen@xorcom.com> wrote:

On Sun, Sep 30, 2007 at 02:34:01AM -0700, bilal ghayyad wrote:
> Dear List;
>
> How can I place a call via Zap/g1 (group) but need to
> determine the line (FXO port)
> that will go via it?

Simply don't use groups. Use channels directly. To dial via the specific
Zaptel channel NN, use Zap/NN

Am I missing anything?

--
               Tzafrir Cohen
icq#16849755              jabber: tzafrir.cohen@xorcom.com
+972-50-7952406           mailto:tzafrir.cohen@xorcom.com
http://www.xorcom.com   iax:guest@local.xorcom.com/tzafrir

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Re: [asterisk-users] What's the deal with ATAcomm?

Andrew Kohlsmith wrote:
> On Saturday 29 September 2007 18:43:59 Andrew Joakimsen wrote:
>> That's horrible. I don't buy too many IP phones these days, but can
>> anyone suggest a place better than the scumbags at VoIP supply?
>
> I don't know about you, but I've had nothing but very good results with
> VOIPSupply. I didnt do huge business with them, but I have purchased new and
> refurb polycoms from them without so much as an ounce of pain.
>
> -A.

I've bought more than $10k worth of equipment from voipsupply.com across
the globe and they've always treated me very professionally. All their
shipments always arrived on time and were well packed and documented.

Just my 2 cents,
Vahan

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[asterisk-users] mISDN NPI setting with b410p

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hi,

I've just received the following mail:

=======================================================================

SETUP Q.931 Message
- - CALLED PARTY NUMBER
- -- NUMBERING PLAN IDENTIFICATION (Octet 3)
- --- 0000 (Unknown)

Full Octet = 80'h, you have set to 81'h. TYPE OF NUMBER should also be
unknown which I have been told you have set, but have not seen it with
my own eyes yet.

Anyway, can you set NUMBERING PLAN IDENTIFICATION = 0000 and retest.

=======================================================================

Does anyone know how or where I would look to set this?

I've found a couple of 0x80's and 0x81's in i4l_mISDN.c but it doesn't
seem to be used.

mISDNdebugtool -v
Received packet from 127.0.0.1:32812 (vers:1 protocol:TE type:D_TX
id:00000200 plen:34)
1191205421.801091696: 00 81 00 00 08 01 01 05 04 03 80 90 a3 18 01 83 6c
09 01 c3 38 37 32 38 34 35 30 70 05 81 36 35 30 30

This is driving me crazy. Is there anyone who can help?

- --
Kind Regards,

Matt Riddell
Director
_______________________________________________

http://www.venturevoip.com (Great new VoIP end to end solution)
http://www.venturevoip.com/news.php (Daily Asterisk News - html)
http://feeds.venturevoip.com/AsteriskNews (Daily Asterisk News - rss)
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.7 (MingW32)
Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org

iD8DBQFHAGLUDQNt8rg0Kp4RAgh6AJ98NQauv8Ze60lNhjRYgL4PNe66AwCdGZIv
gMwh8FIagmPFT20ohzV+M7I=
=diDo
-----END PGP SIGNATURE-----

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[asterisk-users] Astcc does'nt count all calls

Hi:
Iam using astcc on my asterisk server,sometimes astcc does'nt count calls by not writing them into mysql ,example: Not subtracting the call cost from the face value of the entered card number and by not writing it into cdrs table.This problem occured sometimes and not always.
Is it possible that this problem occured due to launching mysql by this command :
/etc/init.d/mysqld start --force

Can any body help me please.

Thanks in advance;
wassim
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Re: [asterisk-users] What's the deal with ATAcomm?

On Saturday 29 September 2007 18:43:59 Andrew Joakimsen wrote:
> That's horrible. I don't buy too many IP phones these days, but can
> anyone suggest a place better than the scumbags at VoIP supply?

I don't know about you, but I've had nothing but very good results with
VOIPSupply. I didnt do huge business with them, but I have purchased new and
refurb polycoms from them without so much as an ounce of pain.

-A.

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Re: [asterisk-users] What's the deal with ATAcomm?

Andrew Joakimsen wrote:
> That's horrible. I don't buy too many IP phones these days, but can
> anyone suggest a place better than the scumbags at VoIP supply?

Reading this raised my eyebrows.

I'm not sure what the content will do for your readers regarding the
company; I am absolutely sure what it does for my opinion of your sense
of fair play.

Name-calling is the oldest, cheapest trick in the book.

b.

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Re: [asterisk-users] Non-USASCII chars in sip.conf?

On Fri, Sep 28, 2007 at 03:40:09PM +0200, Per Jessen wrote:
> This must have been asked before, but googling didn't help much.
> How do I define a callerid that contains non-USASCII characters? E.g. ä,
> ö, ü, å, ø, æ etc. ?

Use UTF-8 Encoding.

--
Stefan Tichy ( asterisk at pi4tel dot de )

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Re: [asterisk-users] Which Asterisk version to use?

"Jim Canfield" <jcanfield@tshmail.com> wrote in message
news:46FBEFF0.2080107@tshmail.com...
Eric B. wrote:

site and got to chapter 4 or 5 and decided to take a break. Which is when
I
found AsteriskNow and TriBox and then started wondering if it was really
necessary / worthwhile to figure out all the intricacies of the application
if someones have already created the appliance version of it. In which
case, I was very confused as to the difference btwn AsteriskNow and TriBox.

Thanks!

Last week I posed a similar question to the list as a "noob".
Specifically, I was curious why every one was so adverse to GUI
implementations. Like you, I entered the asterisk world quite
idealistic
and oblivious to what is actually required to create a functional system
(still am). I spent the good part of last week trying to make heads or
tails of the AsteriskNOW distro, but finally gave up in favor of a plain
jane Debian install with asterisk and wish I would have never wasted so
much time trying to figure out how the users.conf worked.

<quote>
[TK]D-Fender - The users.conf is a flaming piece of sh**!
<\quote>

I actually thought that was a bit harsh when I read it...turns out to be
quite accurate. Long story short, I'm learning to be quite comfortable
in
the CLI and finding myself more productive in nano (yes..nano) than I was
in the GUI.

Good luck!

Thanks for the advice everyone. Will continue reading TFOT and get started!

Eric

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Re: [asterisk-users] Asterisk Dropping Calls (Richard Young)

Hi,

 

Remove

 

usecallingpres=yes

busydetect=yes

 

 

from your zapata.conf file. and the restart asterisk. Hopefully you will not faced drop call issues.

 

 

Regards,

Vidura Senadeera.

 

 

Message: 3
Date: Mon, 24 Sep 2007 12:29:40 +0100
From: "Richard Young" < Richard.Young@intrintech.com>
Subject: [asterisk-users] Asterisk Dropping Calls
To: <asterisk-users@lists.digium.com>
Message-ID:
       < 2EA3B7946768DE4CB0F7F81404A472BF14FA1B@intsbs01.Intrintech.local>
Content-Type: text/plain; charset="us-ascii"

Hello,

I am having an issue whereby calls are being dropped randomly. I have an
ISDN 30 E1 line going into a Wildcard TE220 (4th Gen). My Asterisk
install is based on Trixbox 2.0. However, I have updated the source code
to the following. The Asterisk release is asterisk-1.2.20. Zaptel
release is zaptel-1.2.18. And libpri release is libpri-1.2.4.

I have include an extract from the Asterisk log file below that shows
SIP/781 dropping a call when bridged to Zap/3-1. I have also included my
zaptel and zapata conf files.

I have researched the various messages displayed in the log file but
couldn't see anything that would point definitively to why calls are
being dropped.

Has anyone experienced anything similar or can anyone give me a few
ideas on where to start looking for the cause of the drop-outs?

Many thanks.



/var/log/asterisk/full:



Channel 0/3, span 1 got hangup request, cause 16
Sep 18 16:01:03 DEBUG[32377] channel.c: Didn't get a frame from channel:
Zap/3-1
Sep 18 16:01:03 DEBUG[32377] channel.c: Bridge stops bridging channels
SIP/781-b6e1b590 and Zap/3-1
Sep 18 16:01:03 DEBUG[32377] chan_zap.c: Set option AUDIO MODE, value:
ON(1) on Zap/3-1
Sep 18 16:01:03 DEBUG[32377] chan_zap.c: Hangup: channel: 3 index = 0,
normal = 15, callwait = -1, thirdcall = -1
Sep 18 16:01:03 DEBUG[32377] chan_zap.c: Not yet hungup...  Calling
hangup once with icause, and clearing call
Sep 18 16:01:03 DEBUG[32377] chan_zap.c: disabled echo cancellation on
channel 3
Sep 18 16:01:03 DEBUG[32377] chan_zap.c: Set option TDD MODE, value:
OFF(0) on Zap/3-1
Sep 18 16:01:03 DEBUG[32377] chan_zap.c: Updated conferencing on 3, with
0 conference users
Sep 18 16:01:03 DEBUG[32377] chan_zap.c: Set option AUDIO MODE, value:
OFF(0) on Zap/3-1
Sep 18 16:01:03 DEBUG[32377] chan_zap.c: disabled echo cancellation on
channel 3
Sep 18 16:01:03 VERBOSE[32377] logger.c:     -- Hungup 'Zap/3-1'
Sep 18 16:01:03 DEBUG[32377] app_dial.c: Exiting with DIALSTATUS=ANSWER.

Sep 18 16:01:03 VERBOSE[32377] logger.c:   == Spawn extension
(macro-dialout-trunk, s, 20) exited non-zero on 'SIP/781-b6e1b590' in
macro 'dialout-trunk'
Sep 18 16:01:03 VERBOSE[32377] logger.c:   == Spawn extension
(macro-dialout-trunk, s, 20) exited non-zero on 'SIP/781-b6e1b590'
Sep 18 16:01:03 VERBOSE[32377] logger.c:     -- Executing
Macro("SIP/781-b6e1b590", "hangupcall") in new stack
Sep 18 16:01:03 VERBOSE[32377] logger.c:     -- Executing
ResetCDR("SIP/781-b6e1b590", "w") in new stack
Sep 18 16:01:03 DEBUG[32377] cdr_addon_mysql.c: cdr_mysql: inserting a
CDR record.
Sep 18 16:01:03 DEBUG[32377] cdr_addon_mysql.c: cdr_mysql: SQL command
as follows: INSERT INTO cdr
(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,dura
tion,billsec,disposition,amaflags,accountcode) VALUES ('2007-09-18
15:58:30','02072900400','02072900400','08704440730','from-internal',
'SIP/781-b6e1b590','Zap/3-1','ResetCDR','w',153,150,'ANSWERED',3,'')
Sep 18 16:01:03 DEBUG[32377] app_macro.c: Executed application: ResetCDR

Sep 18 16:01:03 VERBOSE[32377] logger.c:     -- Executing
NoCDR("SIP/781-b6e1b590", "") in new stack
Sep 18 16:01:03 NOTICE[32377] cdr.c: CDR on channel 'SIP/781-b6e1b590'
not posted
Sep 18 16:01:03 NOTICE[32377] cdr.c: CDR on channel 'SIP/781-b6e1b590'
lacks end
Sep 18 16:01:03 DEBUG[32377] app_macro.c: Executed application: NoCDR
Sep 18 16:01:03 DEBUG[32377] pbx.c: Expression result is '1'
Sep 18 16:01:03 VERBOSE[32377] logger.c:     -- Executing
GotoIf("SIP/781-b6e1b590", "1?skiprg") in new stack
Sep 18 16:01:03 VERBOSE[32377] logger.c:     -- Goto
(macro-hangupcall,s,6)
Sep 18 16:01:03 DEBUG[32377] app_macro.c: Executed application: GotoIf
Sep 18 16:01:03 DEBUG[32377] pbx.c: Expression result is '1'
Sep 18 16:01:03 VERBOSE[32377] logger.c:     -- Executing
GotoIf("SIP/781-b6e1b590", "1?theend") in new stack
Sep 18 16:01:03 VERBOSE[32377] logger.c:     -- Goto
(macro-hangupcall,s,9)
Sep 18 16:01:03 DEBUG[32377] app_macro.c: Executed application: GotoIf
Sep 18 16:01:03 VERBOSE[32377] logger.c:     -- Executing
Wait("SIP/781-b6e1b590", "5") in new stack
Sep 18 16:01:03 DEBUG[13856] chan_sip.c: Setting NAT on RTP to 524288
Sep 18 16:01:03 DEBUG[13856] chan_sip.c: Stopping retransmission on
'3c27fa213827-z22macsy3qgz@snom360-0004132394E9' of Response 1: Match
Found
Sep 18 16:01:03 DEBUG[13856] chan_sip.c: Setting NAT on RTP to 524288



My zaptel.conf is as follows:



# Autogenerated by /usr/local/sbin/genzaptelconf -- do not hand edit
# Zaptel Configuration File
#
# This file is parsed by the Zaptel Configurator, ztcfg
#

# It must be in the module loading order


# Span 1: WCT1/0 "Wildcard TE22xP Card 0"
# channel 1, WCT1, unhandled for now
# channel 2, WCT1, unhandled for now
# channel 3, WCT1, unhandled for now
# channel 4, WCT1, unhandled for now
# channel 5, WCT1, unhandled for now
# channel 6, WCT1, unhandled for now
# channel 7, WCT1, unhandled for now
# channel 8, WCT1, unhandled for now
# channel 9, WCT1, unhandled for now
# channel 10, WCT1, unhandled for now
# channel 11, WCT1, unhandled for now
# channel 12, WCT1, unhandled for now
# channel 13, WCT1, unhandled for now
# channel 14, WCT1, unhandled for now
# channel 15, WCT1, unhandled for now
# channel 16, WCT1, unhandled for now
# channel 17, WCT1, unhandled for now
# channel 18, WCT1, unhandled for now
# channel 19, WCT1, unhandled for now
# channel 20, WCT1, unhandled for now
# channel 21, WCT1, unhandled for now
# channel 22, WCT1, unhandled for now
# channel 23, WCT1, unhandled for now
# channel 24, WCT1, unhandled for now
# channel 25, WCT1, unhandled for now
# channel 26, WCT1, unhandled for now
# channel 27, WCT1, unhandled for now
# channel 28, WCT1, unhandled for now
# channel 29, WCT1, unhandled for now
# channel 30, WCT1, unhandled for now
# channel 31, WCT1, unhandled for now

# Global data

loadzone   = uk
defaultzone     = uk

span=1,1,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16



My zapata.conf file looks like this:



;
; Zapata telephony interface
;
; Configuration file

[trunkgroups]

[channels]
language=en
usecallerid=yes
hidecallerid=no
callwaiting=no
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=800
musiconhold=default
rxgain=0.0
txgain=0.0
immediate=no
overlapdial=yes
callgroup=1
pickupgroup=1
pridialplan=unknown
faxdetect=incoming
prilocaldialplan=unknown

group=0
context=from-zaptel
callerid=asreceived
switchtype = euroisdn
signalling = pri_cpe
channel => 1-15,17-31

;Include genzaptelconf configs
;#include zapata-auto.conf

;Include AMP configs
;#include zapata_additional.conf


Kind Regards,

Richard Young
Intrintech Limited
Richard.Young@intrintech.com
111 Cannon Street
London
EC4N 5AR
Phone: 0845 644 2918

[asterisk-users] Asterisk 1.4, h.323, OpenCom 1010

Hi,

there is an OpenCom 1010 (Software Version 4) acting as Gatekeeper
and configured for h.323 connections.

Asterisk 1.4.11 with ooh323 from addons 1.4.2 detects the
gatekeeper, but until now phonecalls are not possible.
To be accurate one test call was successfull, but I did not manage
to get it working again.

Is it a good idea to use ooh323 in this situation? Should I try some
other h.323 module? Is there anyone who has this combination working?

Tanks in advance

--
Stefan Tichy ( asterisk at pi4tel dot de )

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Re: [asterisk-users] Selecting a specific line from Zap/g

On Sun, Sep 30, 2007 at 02:34:01AM -0700, bilal ghayyad wrote:
> Dear List;
>
> How can I place a call via Zap/g1 (group) but need to
> determine the line (FXO port)
> that will go via it?

Simply don't use groups. Use channels directly. To dial via the specific
Zaptel channel NN, use Zap/NN

Am I missing anything?

--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.cohen@xorcom.com
+972-50-7952406 mailto:tzafrir.cohen@xorcom.com

http://www.xorcom.com

iax:guest@local.xorcom.com/tzafrir

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[asterisk-users] Selecting a specific line from Zap/g

Dear List;

How can I place a call via Zap/g1 (group) but need to
determine the line (FXO port)
that will go via it?

Also, how it will be possible to assign an dedicated
line (connected to FXO) to an
button on the Polycom IP Phone or Broadtel IP Phone,
so if user select that button
then he will be sure that his outside call will be via
that specific line.

Regards
Bilal

____________________________________________________________________________________
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Re: [asterisk-users] echo problems

Also many people using softphone turn's on mic boost in windows xp which also makes echo if it is set to very loud .

On 30/09/2007, Philipp Kempgen < philipp.kempgen@amooma.de> wrote:

http://linux.sgms-centre.com/misc/netiquette.php#threading
http://linux.sgms-centre.com/misc/netiquette.php#toppost
SCNR

Regards,
  Philipp Kempgen

--
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
    Let's use IT to solve problems and not to create new ones.
          Asterisk? -> http://www.das-asterisk-buch.de
              My pick of the month: rfc 2822 3.6.5

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Re: [asterisk-users] echo problems

http://linux.sgms-centre.com/misc/netiquette.php#threading
http://linux.sgms-centre.com/misc/netiquette.php#toppost
SCNR

Regards,
Philipp Kempgen

--
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de

Let's use IT to solve problems and not to create new ones.
Asterisk? -> http://www.das-asterisk-buch.de

My pick of the month: rfc 2822 3.6.5

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[asterisk-users] echo problems

Normally I use

DID provider - asterisk - sip terminator ( both DID provider and sip terminator use g729, so asterisk just passes the call between the 2 of them ).

All works wonderfull, no echo etc.

But we would like to offer the service with softphones.

However users experience echo, be it with gsm, ulaw or alaw.

Someone would like to share from his experience for  solving this echo problem?

If ta detailed answer is too much, just telling me where to look for would be great. I searched voip-info.org  but I wasn't able to find something which fits my exact problem, since I don't use TDM cards....

Best regards,

Apa

Michiel van Baak <michiel@vanbaak.info> wrote:

On 16:34, Sat 29 Sep 07, Doug wrote:
> At 04:57 9/29/2007, Michiel van Baak, wrote:
> >On 21:47, Fri 28 Sep 07, Doug wrote:
> >> At 20:53 9/28/2007, Tzafrir Cohen wrote:
> >> >On Fri, Sep 28, 2007 at 01:28:18PM -0500, Doug wrote:
> >> >> >How do you do that when your single network connection is gone?
> >> >>
> >> >> Any suggestions on dual-wan routers? We can't get this
> >> >> stupid Twin-Wan to work:
> >> >>
> >> >> http://www.xincom.com/twinwan.php
> >> >
> >> >A PC?
> >>
> >> OS? App? ;^)
> >
> >We use OpenBSD with carp+pfsync for this.
> >There are some good tutorials on the interweb on how to do
> >this.
>
> Which tube do I go through to get to the InterWeb?

You can use my doc to setup redundant firewalling:
http://michiel.vanbaak.info/page/soekrisobsdcarp.htm

Read the PF faq for information on how to loadbalance 2
internet connections:
http://www.openbsd.org/faq/pf/pools.html

--

Michiel van Baak
michiel@vanbaak.eu
http://michiel.vanbaak.eu
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"Why is it drug addicts and computer afficionados are both called users?"


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Re: [asterisk-users] Asterisk Redundancy

On 16:34, Sat 29 Sep 07, Doug wrote:
> At 04:57 9/29/2007, Michiel van Baak, wrote:
> >On 21:47, Fri 28 Sep 07, Doug wrote:
> >> At 20:53 9/28/2007, Tzafrir Cohen wrote:
> >> >On Fri, Sep 28, 2007 at 01:28:18PM -0500, Doug wrote:
> >> >> >How do you do that when your single network connection is gone?
> >> >>
> >> >> Any suggestions on dual-wan routers? We can't get this
> >> >> stupid Twin-Wan to work:
> >> >>
> >> >> http://www.xincom.com/twinwan.php
> >> >
> >> >A PC?
> >>
> >> OS? App? ;^)
> >
> >We use OpenBSD with carp+pfsync for this.
> >There are some good tutorials on the interweb on how to do
> >this.
>
> Which tube do I go through to get to the InterWeb?

You can use my doc to setup redundant firewalling:
http://michiel.vanbaak.info/page/soekrisobsdcarp.htm

Read the PF faq for information on how to loadbalance 2
internet connections:
http://www.openbsd.org/faq/pf/pools.html

--

Michiel van Baak
michiel@vanbaak.eu
http://michiel.vanbaak.eu
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x71C946BD

"Why is it drug addicts and computer afficionados are both called users?"

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Re: [asterisk-users] What's the deal with ATAcomm?

That's horrible. I don't buy too many IP phones these days, but can
anyone suggest a place better than the scumbags at VoIP supply?

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Re: [asterisk-users] Asterisk Redundancy

At 04:57 9/29/2007, Michiel van Baak, wrote:
>On 21:47, Fri 28 Sep 07, Doug wrote:
>> At 20:53 9/28/2007, Tzafrir Cohen wrote:
>> >On Fri, Sep 28, 2007 at 01:28:18PM -0500, Doug wrote:
>> >> >How do you do that when your single network connection is gone?
>> >>
>> >> Any suggestions on dual-wan routers? We can't get this
>> >> stupid Twin-Wan to work:
>> >>
>> >> http://www.xincom.com/twinwan.php

>> >
>> >A PC?
>>
>> OS? App? ;^)
>
>We use OpenBSD with carp+pfsync for this.
>There are some good tutorials on the interweb on how to do
>this.

Which tube do I go through to get to the InterWeb?

>
>--
>
>Michiel van Baak
>michiel@vanbaak.eu
>http://michiel.vanbaak.eu

>GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x71C946BD

>
>"Why is it drug addicts and computer afficionados are both called users?"
>
>
>_______________________________________________
>
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>

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Re: [asterisk-users] FAX detection not working

>>> On 9/29/2007 at 3:27 PM, Lee Howard <faxguy@howardsilvan.com> wrote:
> Joe Acquisto wrote:
>
>>As I understand it, I must have faxdetect = incoming to enable detection of
> the fax tone.
>>Then, I must have a [fax] context to pickup the line and send it to whatever
> extension the FAX device is on.
>>
>
> It's a "fax" extension in the context where the call is at... not a fax
> context in the dialplan.
>
> Lee.
>

I don't follow. Sorry.

Now might be a good time to post this, since Tzafrir asked, it looks very much like bits I have seen on the net. I did see what appeared to be the analog_fax part when checking at CLI.

So, I would surmise it detected the FAX and is trying to deal with it, but the number derived via LDAPget is hosed? It just ends up hanging up and not dialing any extension.

{begin snippet]
[ext-fax]
exten => s,1,Answer
exten => s,2,Goto(in_fax|1)
exten => in_fax,1,GotoIf($[${FAX_RX} = system]?2:analog_fax|1)
exten => in_fax,2,Macro(faxreceive)
exten => in_fax,3,system(tiff2ps -2eaz -w 8.5 -h 11 ${FAXFILE} | ps2pdf - ${FAXFILE}.pdf)
exten => in_fax,4,system(mime-construct --to ${EMAILADDR} --subject "Fax from ${CALLERID(num)} ${CALLERID(name)}" --attachment ${CALLERID(num)}.pdf --type application/pdf --file ${FAXFILE}.pdf)
exten => in_fax,5,system(rm ${FAXFILE} ${FAXFILE}.pdf)
exten => in_fax,6,Hangup
exten => analog_fax,1,GotoIf($[foo${FAX_RX} = foo]?3:2)
exten => analog_fax,2,LDAPget(DIAL=xxxxxxxxDeviceDial/${FAX_RX})
exten => analog_fax,3,Dial(${DIAL}|20|d)
exten => analog_fax,4,Hangup
exten => out_fax,1,txfax(${TXFAX_NAME}|caller)
exten => out_fax,2,Hangup
exten => h,1,Hangup()
[end snippet]

joe a.

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[asterisk-users] Big problems with TDM2400 :(

Hello Fellows!

I have a TDM2400 and I can't put it to work. Every time it receive a call the Asterisk handle it and call the SIP phone; when people pick up the fone they don't hear nothing and the caller hear the phone rings and nothing happens. In Asterisk console I can see the message answered by the SIP's phone.
I lost a lot of time trying to solve this problem without success :(.

     == Starting post polarity CID detection on channel 21
        -- Starting simple switch on 'Zap/21-1'
        -- Executing [s@entrada:1] Answer("Zap/21-1", "") in new stack
        -- Executing [s@entrada:2] Dial("Zap/21-1", "SIP/ramal01&SIP/ramal02&SIP/ramal03|30|tT|r") in new stack
        -- Called ramal01
        -- Called ramal02
        -- Called ramal03
        -- SIP/ramal03-0070e020 is ringing
        -- SIP/ramal01-006fd4f0 is ringing
        -- SIP/ramal02-00705d70 is ringing
        -- SIP/ramal01-006fd4f0 answered Zap/21-1
      == Spawn extension (entrada, s, 2) exited non-zero on 'Zap/21-1'
        -- Hungup 'Zap/21-1'


I got the following message when a enable the usecallerid=yes:

    Sep 29 16:48:31 WARNING[12369]: chan_zap.c:5961 ss_thread: DTMFCID timed out waiting for ring. Exiting simple switch
        -- Hungup 'Zap/21-1'
      == Starting post polarity CID detection on channel 21
        -- Starting simple switch on 'Zap/21-1'
    Sep 29 16:48:35 WARNING[12372]: chan_zap.c:5961 ss_thread: DTMFCID timed out waiting for ring. Exiting simple switch
        -- Hungup 'Zap/21-1'



I've tested with the Zaptel 1.2/ Asterisk 1.2 and Zaptel 1.4.5.1/Asterisk 1.4.11 and got the same problem.
Debian Etch amd64.

Thanks for any help!

Regards,

McCoy Silva

Re: [asterisk-users] FAX detection not working

Joe Acquisto wrote:

>As I understand it, I must have faxdetect = incoming to enable detection of the fax tone.
>Then, I must have a [fax] context to pickup the line and send it to whatever extension the FAX device is on.
>

It's a "fax" extension in the context where the call is at... not a fax
context in the dialplan.

Lee.

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Re: [asterisk-users] VoIP, Asterisk, What Do you really need ?

Giovanni Miano wrote:
> Hello folks,
> I was wondering, talking about VoIP, Asterisk or whatever related to it
>
> What is the Function or Service you really need to create your own
> business, simplify service issue, increase your market-cap ?
> Is it there but is it not open-source or free ?
>
> I would like collect informations to setup a "box of VoIP - Idea, what
> I need, what I can use but I cannot create"
>
> Thanks,
> John

Make an Aheeva (with the looks, bells and whistles) like product that
is opensource.

Thanks,
Steve

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Re: [asterisk-users] FAX detection not working

On Sat, Sep 29, 2007 at 08:56:56AM -0400, Joe Acquisto wrote:
> I am having a problem detecting incoming FAX. TMD22p (tdm400p 2 fxo, 2fxs)
>
> As I understand it, I must have faxdetect = incoming to enable detection of
> the fax tone. Then, I must have a [fax] context to pickup the line and send
> it to whatever extension the FAX device is on.
>
> In my case, I ask it to answer immediately and do a distinctive ring (r3) to
> alert that is its a FAX call so no one picks up the line.

Fax detection detects a tone on the line. Hence it only owrks after the
line has been answered.

>
> however, it seems the FAX tone is not being detected (I know it is being
> sent), as the normal ring tone is heard.
>
> I must be misunderstanding how this works. Or does not work.

Can you please provide your relevant dialplan snippets and relevant
parts of zapata.conf ?

--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.cohen@xorcom.com
+972-50-7952406 mailto:tzafrir.cohen@xorcom.com

http://www.xorcom.com

iax:guest@local.xorcom.com/tzafrir

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Re: [asterisk-users] FAX detection not working

This can be a partial never mind, I guess. I can see via the CLI that the call is being handled by
some FAX related routines. Just not quite the solution I expected.

joe a.

>>> On 9/29/2007 at 8:56 AM, "Joe Acquisto" <joea@j4computers.com> wrote:
> I am having a problem detecting incoming FAX. TMD22p (tdm400p 2 fxo, 2fxs)
>
> As I understand it, I must have faxdetect = incoming to enable detection of
> the fax tone.
> Then, I must have a [fax] context to pickup the line and send it to whatever
> extension the FAX device is on.
>
> In my case, I ask it to answer immediately and do a distinctive ring (r3) to
> alert that is its a FAX call so no one picks up the line.
>
> however, it seems the FAX tone is not being detected (I know it is being
> sent), as the normal ring tone is heard.
>
> I must be misunderstanding how this works. Or does not work.
>
> joe a.
>
>
> _______________________________________________
>
> Sign up now for AstriCon 2007! September 25-28th.

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[asterisk-users] FAX detection not working

I am having a problem detecting incoming FAX. TMD22p (tdm400p 2 fxo, 2fxs)

As I understand it, I must have faxdetect = incoming to enable detection of the fax tone.
Then, I must have a [fax] context to pickup the line and send it to whatever extension the FAX device is on.

In my case, I ask it to answer immediately and do a distinctive ring (r3) to alert that is its a FAX call so no one picks up the line.

however, it seems the FAX tone is not being detected (I know it is being sent), as the normal ring tone is heard.

I must be misunderstanding how this works. Or does not work.

joe a.

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Re: [asterisk-users] IAX gsm bandwith calls

How much bandwidth does speex use usually?

Tom

-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Steve Totaro
Sent: Friday, September 28, 2007 11:03 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] IAX gsm bandwith calls

Andrew Joakimsen wrote:
> On 9/26/07, Tom Moore <tmoore@issvsat.com> wrote:
>
>> If you've got a bandwidth of something that low you'll probably want to
use
>> g723.1 or g729 on this line.
>> If your lucky you'll be able to place two calls at once over this link.
>> You won't be able to do anything else though.
>>
>> Tom
>>
>>
>
> If you really want to maximize your bandwidth try LPC codec! You can
> probably squeeze 5 maybe 6 calls on there... and sound like a robot.
>
>

Speex rocks!

Thanks,
Steve

Typed using my fingers on my laptop in the Phoenix Airport waiting for
my flight home from Astricon.

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No virus found in this incoming message.
Checked by AVG Free Edition.
Version: 7.5.488 / Virus Database: 269.13.33/1034 - Release Date: 9/27/2007
5:00 PM

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Checked by AVG Free Edition.
Version: 7.5.488 / Virus Database: 269.13.33/1034 - Release Date: 9/27/2007
5:00 PM

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Re: [asterisk-users] Asterisk realtime error

Hi Renzzo,

Which linux distribution you are using?

maybe problem is due to mysql.sock

try with direct tcp connection.

1. remove "dbsock = /var/lib/mysql/mysql.sock" line
2. change mysql user host permission from "localhost" "%"
3. dump mysql permissions or restart mysql;
4. reload / restart asterisk.

Regards

Nasir Iqbal
http://www.ictinnovations.com

On Wed, 2007-09-26 at 23:25 -0500, RENZZO SOTOMAYOR wrote:
> Peder, I have all the permissions in mysql user. I can query my
> database from the local box.
> Mik Cheez, yes, it is. mysql.sock is in /var/lib/mysql/
> Asterisk and Mysql are in the same PC
> I still have the same error and don't know what to do.
> help plz!
>
> thanks in advance,
> Renzzo
>
>
>
> Mik Cheez wrote:
> >Is your mysql.sock actually in /var/lib/mysql/ ?
>
>
> Peder wrote:
> >Could be a mysql permission issue. Try this from the local box:
> >
> >mysql -u root -p
> ><enter asterisk as the password>
> >use asterisk;
> >select * from sip_buddies;
> >select * from iax_buddies;
> >
> >If you get that far and can see the entries in iax_buddies and
> >sip_buddies, you know it isn't a permissions issue. If you can't,
> then
> >you know where to look.
>
>
> RENZZO SOTOMAYOR wrote:
> > Hi! I am proving Asterisk 1.2.24 in realtime with MySQL 5.0.27 using
> > Idefisk softphones. I followed the steps of "how to" of voip-org but
> > always have this error:
> >
> > Sep 25 20:29:07 WARNING[12000]: res_config_mysql.c:360 update_mysql:
> > MySQL RealTime: Failed to query database. Check debug for more
> info.
> > Sep 25 20:29:07 WARNING[12000]: res_config_mysql.c:360 update_mysql:
> > MySQL RealTime: Failed to query database. Check debug for more info.
> > Sep 25 20:29:07 WARNING[12000]: res_config_mysql.c:360
> update_mysql:
> > MySQL RealTime: Failed to query database. Check debug for more info.
> > Sep 25 20:29:07 NOTICE[12000]: chan_iax2.c:5252 register_verify:
> Host
> > 127.0.0.1 <http://127.0.0.1/> failed MD5 authentication for '101'
> > (9a43a82001dfa49d84e8facb765f7
> d
> > e2 != 31610d29241e861816b83998501ee223)
> >
> > I configure extconfig.conf as:
> > [settings]
> > iaxusers => mysql,asterisk,iax_buddies
> > iaxpeers => mysql,asterisk,iax_buddies
> > sipusers => mysql,asterisk,sip_buddies
> > sippeers => mysql,asterisk,sip_buddies
> >
> > res_mysql.conf as:
> > [general]
> > dbhost = localhost
> > dbname = asterisk
> > dbuser = root
> > dbpass = asterisk
> > dbport = 3306
> > dbsock = /var/lib/mysql/mysql.sock
> >
> > My table as:
> > CREATE TABLE iax_buddies (
> > name varchar(30) primary key NOT NULL,
> > username varchar(30),
> > type varchar(6) NOT NULL,
> > secret varchar(50),
> > callerid varchar(100),
> > context varchar(100),
> > host varchar(31) NOT NULL default 'dynamic',
> > disallow varchar(100),
> > allow varchar(100)
> > );
> >
> > I'm running asterisk on Fedora 6. Plz help
> >
> > thanks in advance
> >
> > Renzzo
> >
> >
> >
> ------------------------------------------------------------------------
> >
> > _______________________________________________
> >
> > Sign up now for AstriCon 2007! September 25-28th.
>

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>
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Re: [asterisk-users] Asterisk Redundancy

On 21:47, Fri 28 Sep 07, Doug wrote:
> At 20:53 9/28/2007, Tzafrir Cohen wrote:
> >On Fri, Sep 28, 2007 at 01:28:18PM -0500, Doug wrote:
> >> >How do you do that when your single network connection is gone?
> >>
> >> Any suggestions on dual-wan routers? We can't get this
> >> stupid Twin-Wan to work:
> >>
> >> http://www.xincom.com/twinwan.php
> >
> >A PC?
>
> OS? App? ;^)

We use OpenBSD with carp+pfsync for this.
There are some good tutorials on the interweb on how to do
this.

--

Michiel van Baak
michiel@vanbaak.eu
http://michiel.vanbaak.eu
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x71C946BD

"Why is it drug addicts and computer afficionados are both called users?"

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[asterisk-users] VoIP, Asterisk, What Do you really need ?

Hello folks,
I was wondering, talking about VoIP, Asterisk or whatever related to it

What is the Function or Service you really need to create your own business, simplify service issue, increase your market-cap ?
Is it there but is it not open-source or free ?

I would like collect informations to setup a "box of VoIP - Idea, what I need, what I can use but I cannot create"

Thanks,
John

Re: [asterisk-users] Proximity Detection: Motorola Q + Bluetooth + Asterisk

On 9/28/07, Chuck Bunn <chuck.bunn@networkdoc.com> wrote:
> Hi,
>
> Can anyone tell me if the Motorola Q has its Bluetooth always on like
> the IPhone? I want to use the Motorola Q in a Proximity Detection setup
> like that described on nerdvittles.com. I know the Treo 650 does not
> work well since the display must be on for the bluetooth to be on and
> this eats power.
>
> Thanks
>
> Chuck Bunn
>

I don't want to install a bluetooth dongle on a server just to test :(

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Re: [asterisk-users] IAX gsm bandwith calls

On 9/26/07, Tom Moore <tmoore@issvsat.com> wrote:
>
>
> If you've got a bandwidth of something that low you'll probably want to use
> g723.1 or g729 on this line.
> If your lucky you'll be able to place two calls at once over this link.
> You won't be able to do anything else though.
>
> Tom
>

If you really want to maximize your bandwidth try LPC codec! You can
probably squeeze 5 maybe 6 calls on there... and sound like a robot.

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Re: [asterisk-users] Proximity Detection: Motorola Q + Bluetooth + Asterisk

Andrew Joakimsen wrote:
> On 9/28/07, Chuck Bunn <chuck.bunn@networkdoc.com> wrote:
>
>> Hi,
>>
>> Can anyone tell me if the Motorola Q has its Bluetooth always on like
>> the IPhone? I want to use the Motorola Q in a Proximity Detection setup
>> like that described on nerdvittles.com. I know the Treo 650 does not
>> work well since the display must be on for the bluetooth to be on and
>> this eats power.
>>
>> Thanks
>>
>> Chuck Bunn
>>
>>
>
> I don't want to install a bluetooth dongle on a server just to test :(
>
>
>

I cannot imagine anything easier than a USB dongle. The bluetooth part
is simple too.

Thanks,
Steve

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Re: [asterisk-users] . (period): Wildcard match; matches one or more characters

On Fri, Sep 28, 2007 at 03:34:29PM +0200, Philipp Kempgen wrote:
> bilal ghayyad wrote:
>
> > In the outbound, I read in the documents the Wildcard
> > match "by using the . (period)", but I did not
> > understand how Wildcard will work (like what)?
>
> http://en.wikipedia.org/wiki/Wildcard_character
>
> > As I
> > know that Wildcard is a term used with the Diguim TDM
> > card (FXO and FXS), so what is the relation between
> > such cards and the matching in the dial plan?
>
> There is no relation.

No direct relation. The "Wildcard" hardware cards are simply named after
a different wildcard: '*'.

--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.cohen@xorcom.com
+972-50-7952406 mailto:tzafrir.cohen@xorcom.com

http://www.xorcom.com

iax:guest@local.xorcom.com/tzafrir

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Re: [asterisk-users] IAX gsm bandwith calls

Andrew Joakimsen wrote:
> On 9/26/07, Tom Moore <tmoore@issvsat.com> wrote:
>
>> If you've got a bandwidth of something that low you'll probably want to use
>> g723.1 or g729 on this line.
>> If your lucky you'll be able to place two calls at once over this link.
>> You won't be able to do anything else though.
>>
>> Tom
>>
>>
>
> If you really want to maximize your bandwidth try LPC codec! You can
> probably squeeze 5 maybe 6 calls on there... and sound like a robot.
>
>

Speex rocks!

Thanks,
Steve

Typed using my fingers on my laptop in the Phoenix Airport waiting for
my flight home from Astricon.

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Re: [asterisk-users] Asterisk Redundancy

At 20:53 9/28/2007, Tzafrir Cohen wrote:
>On Fri, Sep 28, 2007 at 01:28:18PM -0500, Doug wrote:
>> >How do you do that when your single network connection is gone?
>>
>> Any suggestions on dual-wan routers? We can't get this
>> stupid Twin-Wan to work:
>>
>> http://www.xincom.com/twinwan.php

>
>A PC?

OS? App? ;^)

>
>--
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>icq#16849755 jabber:tzafrir.cohen@xorcom.com
>+972-50-7952406 mailto:tzafrir.cohen@xorcom.com
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>
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Re: [asterisk-users] How to "busy out" zap channels

Andrew Joakimsen wrote:
> On 9/26/07, Brian Roy <mister.roy@gmail.com> wrote:
>
>> Anyone have a better idea? Or do they have anything like this so I'm not
>> putting it together?
>>
>>
>
> If its PRI why don't you try:
>
> exten => 0000000000,1,Set(PRI_CAUSE=27)
> exten => 0000000000,2,Hangup
>
> Or cause code 17
>
> 17 = User Busy. The number dialed is busy and cannot receive any more calls.
> 27 = Destination Out-of-Order. This is a working number, but the span
> to the destination is not active or there is a problem sending
> messages to this destination.
>
>

I am pretty sure there is no way to busy out a channel currently. You
could make it busy by using it though.

Thanks,
Steve

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Re: [asterisk-users] How to "busy out" zap channels

On 9/26/07, Brian Roy <mister.roy@gmail.com> wrote:
>
> Anyone have a better idea? Or do they have anything like this so I'm not
> putting it together?
>

If its PRI why don't you try:

exten => 0000000000,1,Set(PRI_CAUSE=27)
exten => 0000000000,2,Hangup

Or cause code 17

17 = User Busy. The number dialed is busy and cannot receive any more calls.
27 = Destination Out-of-Order. This is a working number, but the span
to the destination is not active or there is a problem sending
messages to this destination.

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Re: [asterisk-users] Asterisk Redundancy

On Fri, Sep 28, 2007 at 01:28:18PM -0500, Doug wrote:
> >How do you do that when your single network connection is gone?
>
> Any suggestions on dual-wan routers? We can't get this
> stupid Twin-Wan to work:
>
> http://www.xincom.com/twinwan.php

A PC?

--
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[asterisk-users] meetme conference using g729?

Hi,

is there a way to use g729 in meetme?

Thanks!

[asterisk-users] Asterisk realtime error

Peder, I have all the permissions in mysql user. I can query my database from the local box.
Mik Cheez, yes, it is. mysql.sock is in /var/lib/mysql/
Asterisk and Mysql are in the same PC
I still have the same error and don't know what to do.
help plz!

thanks in advance,
Renzzo



Mik Cheez wrote:
>Is your mysql.sock actually in /var/lib/mysql/ ?


Peder wrote:
>Could be a mysql permission issue.  Try this from the local box:
>
>mysql -u root -p
><enter asterisk as the password>
>use asterisk;
>select * from sip_buddies;
>select * from iax_buddies;
>
>If you get that far and can see the entries in iax_buddies and
>sip_buddies, you know it isn't a permissions issue.  If you can't, then
>you know where to look.


RENZZO SOTOMAYOR wrote:
> Hi! I am proving Asterisk 1.2.24 in realtime with MySQL 5.0.27 using
> Idefisk softphones. I followed the steps of "how to" of voip-org but
> always have this error:
>
> Sep 25 20:29:07 WARNING[12000]: res_config_mysql.c:360 update_mysql:
> MySQL RealTime: Failed to query database. Check debug for more info.
> Sep 25 20:29:07 WARNING[12000]: res_config_mysql.c:360 update_mysql:
> MySQL RealTime: Failed to query database. Check debug for more info.
> Sep 25 20:29:07 WARNING[12000]: res_config_mysql.c:360 update_mysql:
> MySQL RealTime: Failed to query database. Check debug for more info.
> Sep 25 20:29:07 NOTICE[12000]: chan_iax2.c:5252 register_verify: Host
> 127.0.0.1 <http://127.0.0.1/> failed MD5 authentication for '101'
> (9a43a82001dfa49d84e8facb765f7

d
> e2 != 31610d29241e861816b83998501ee223)
>
> I configure extconfig.conf as:
> [settings]
> iaxusers => mysql,asterisk,iax_buddies
> iaxpeers => mysql,asterisk,iax_buddies
> sipusers => mysql,asterisk,sip_buddies
> sippeers => mysql,asterisk,sip_buddies
>
> res_mysql.conf as:
> [general]
> dbhost = localhost
> dbname = asterisk
> dbuser = root
> dbpass = asterisk
> dbport = 3306
> dbsock = /var/lib/mysql/mysql.sock
>
> My table as:
> CREATE TABLE iax_buddies (
>        name varchar(30) primary key NOT NULL,
>        username varchar(30),
>        type varchar(6) NOT NULL,
>        secret varchar(50),
>        callerid varchar(100),
>        context varchar(100),
>        host varchar(31) NOT NULL default 'dynamic',
>        disallow varchar(100),
>        allow varchar(100)
> );
>
> I'm running asterisk on Fedora 6. Plz help
>
> thanks in advance
>
> Renzzo
>
>
> ------------------------------------------------------------------------
>
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Re: [asterisk-users] . (period): Wildcard match; matches one or more characters

On Sep 28, 2007, at 4:52 PM, Mojo with Horan & Company, LLC wrote:

To use the 

wildcard characters, 'X', 'N', or '.',  I had to also prefix my 

extension with '_', which enables pattern matching.

Don't forget you also have Z which if I recall its 1-9, N is 2-9 and X is 0-9

/b

Re: [asterisk-users] . (period): Wildcard match; matches one or more characters

An example similar to one that exists in many dialplans:

exten => _011.,1,Dial(Zap/g1/${EXTEN})

which would match any international number as dialed from North America
because, depending on what country you'd be calling, the number of
digits after the 011 would differ. As such, putting the period after
the 011 says 'match 011 followed by one or more digits". To use the
wildcard characters, 'X', 'N', or '.', I had to also prefix my
extension with '_', which enables pattern matching.

Mojo

bilal ghayyad wrote:
> Hi List;
>
> In the outbound, I read in the documents the Wildcard
> match "by using the . (period)", but I did not
> understand how Wildcard will work (like what)? As I
> know that Wildcard is a term used with the Diguim TDM
> card (FXO and FXS), so what is the relation between
> such cards and the matching in the dial plan?
>
> Any help?
>
> Regards
> Bilal
>
>
> ____________________________________________________________________________________
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Re: [asterisk-users] asterisk audits

On 9/27/07, Tilghman Lesher <tilghman@mail.jeffandtilghman.com> wrote:

On Wednesday 26 September 2007 18:39:31 Mark Quitoriano wrote:
> Some company asked me to do audits with there asterisk boxes. Is there a
> standard that i should be following in auditing? anyway can give me a start
> what to do with asterisk audits?

Have you considered the ethics of getting yourself hired to do something you
don't know how to do?  Worse, have you considered the ramifications of posting
to a publically archived list that you got yourself hired to do a job you're
unqualified for?

senseless post which doesn't help at all.

First of all they're not hiring me for this they just asking for a favor and i'm not familiar with this and thought want to know more about asterisk auditing.

Re: [asterisk-users] G.722: ast_channel_make_compatible failure

Ondrej Valousek wrote:

> [Sep 20 10:14:32] WARNING[30706]: chan_sip.c:2963 sip_call: No audio
> format found to offer. Cancelling call to phone3

Asterisk 1.4 does not have G.722 transcoding, only passthrough support.
It can connect G.722 channels together, and record or playback G.722
audio files, but that is all.

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

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[asterisk-users] odd audio problem

I am having an odd audio problem. See setup diagram below. When a call
comes in it get routed through the 1st asterisk box (currently running
1.2) through another asterisk box (running 1.4.11). All audio is good.

When I upgraded the 1st asterisk box to 1.4.11. A call comes in, relays
to the 2nd asterisk box. The AA answers the call and the audio is good.
Once the call is forwarded to an agent. The agent hears everything no
problem, but the audio returned to the callers is really bad. It sounds
like it is missing 75% of the audio.

There is no packet loss and 10 ms ping times between the two asterisk
boxes. All audio streams are g729 and there is no trans coding anywhere.

When I recorded the audio on both asterisk boxes using Mixmonitor, the
recorded files sounded good.

+---------+
| TNT MAX |
+---------+
|
| SIP G729
V
+----------------+
| Asterisk Box 1 |
+----------------+
|
| IAX2 G729
V
+--------------------+
| Asterisk 1.4 Box 2 |
+--------------------+
|
| IAX2 G729
V
Agents

Peter

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[asterisk-users] Nano syntax highlighting.

Greetings,

I know the hardcore guys will laugh, but I put together a quick .nanorc
config for asterisk. I tried to include all the applications listed on
the latest install. Please feel free to send any suggestions/updates my
why. I think this will go a long way to helping out the "new guys" when
reading configs. Try it out, I hope you like it. Perhaps we have a
regex guru out there who can make it better.

Instructions:

Just paste the contents of the link below in your .nanorc file.

http://www.voip-info.org/users/499/49499/images/1745/NanoHighlightAsterisk.txt

-jc

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Re: [asterisk-users] Asterisk Redundancy

At 08:01 9/28/2007, Per Jessen wrote:
>Douglas Garstang wrote:
>
>>>Also be sure that you have a very redundant network configuration.
>>>Too often I see people spend a great deal of time and money to get
>>>redundant servers when their switches, firewalls, routers, etc are not
>>>even capable of handling a failed network element.
>>
>> You can achieve this at the application level.
>
>How do you do that when your single network connection is gone?

Any suggestions on dual-wan routers? We can't get this
stupid Twin-Wan to work:

http://www.xincom.com/twinwan.php

>
>When considering redundancy it is essential that you have no single
>point of failure. Depending on how far you want to go, this means
>right from your dual-box asterisk setup to dual diesel-generators and
>two multi-homed datacenters.
>
>
>
>/Per Jessen, Zürich
>
>--
>http://www.spamchek.com/ - your spam is our business.
>
>
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Re: [asterisk-users] Changing contexts "on the fly"

Mojo with Horan & Company, LLC wrote:

> *6 is for *N, for people to remember (N)ight mode. In my *6 extension,
> I create a mutex in a sense, the file called 'night_mode' in /home/pbx
> -- this lets me determine if night mode is enabled via external systems,
> like those written in PHP for a webpage or something else for a shell
> script, maybe as a cron schedule that rings your desk to remind you that
> night mode is still on... It is not needed for my incoming context;
> that context uses the astdb.

Nice Mojo...

Here is something I have on one machine... Ugly but effective

; Nitemode
exten => 5551,1,System(asterisk -rx "dont include biz-day-aa in biz-aa")
exten => 5551,2,System(asterisk -rx "include biz-nite-aa in biz-aa")
exten => 5551,3,Hangup ;SetVar(__main=1)

; Daymode

exten => 5552,1,System(asterisk -rx "dont include biz-nite-aa in biz-aa")
exten => 5552,2,System(asterisk -rx "include biz-day-aa in biz-aa")
exten => 5552,3,System(asterisk -rx reload);
exten => 5552,4,Hangup ;SetVar(__main=0)

[biz-day-aa]
include => 1600
exten => s,1,Wait(1)
exten => s,2,Ringing
exten => s,3,Dial(SIP/xxx&SIP/xxx|20|r)
etc, etc, etc

[biz-nite-aa]
exten => s,1,Wait(1)
exten => s,2,Answer
exten => s,3,Background(biz/welcome)
exten => s,4,Background(biz/biz-aa)
exten => s,5,Background(silence/5)
exten => s,6,Goto(biz-nite-aa|s|1)

Here is something I did using GotoIfTime. The client has an extension
(6566) so that they can record their own greetings so we wouldn't have
to swap them out for them...

; Martin Luther King's Day
exten => s,3,GotoIfTime(14:00-23:59|*|14|jan?bizclient-aa,6566,1)
;exten => s,4,GotoIfTime(*|*|15|jan?bizclient-aa,6566,1)
exten => s,4,GotoIfTime(*|*|13-15|jul?bizclient-aa,6566,1)

; Memorial Day
exten => s,5,GotoIfTime(14:00-23:59|*|27|may?bizclient-aa,6566,1)
exten => s,6,GotoIfTime(*|*|28|may?bizclient-aa,6566,1)

; Independence Day
exten => s,7,GotoIfTime(12:00-23:59|*|3|jul?bizclient-aa,6566,1)
exten => s,8,GotoIfTime(*|*|4|jul?bizclient-aa,6566,1)

; Labor Day
exten => s,9,GotoIfTime(14:00-23:59|*|2|sep?bizclient-aa,6566,1)
exten => s,10,GotoIfTime(*|*|3|sep?bizclient-aa,6566,1)

; Columbus Day
exten => s,11,GotoIfTime(14:00-23:59|*|7|oct?bizclient-aa,6566,1)
exten => s,12,GotoIfTime(*|*|8|oct?bizclient-aa,6566,1)

; Veterans Day
exten => s,13,GotoIfTime(14:00-23:59|*|11|nov?bizclient-aa,6566,1)
exten => s,14,GotoIfTime(*|*|12|nov?bizclient-aa,6566,1)

; Thanksgiving Day
exten => s,15,GotoIfTime(14:00-23:59|*|21|nov?bizclient-aa,6566,1)
exten => s,16,GotoIfTime(*|*|22|nov?bizclient-aa,6566,1)

--
====================================================
J. Oquendo
"Excusatio non petita, accusatio manifesta"

http://pgp.mit.edu:11371/pks/lookup?op=get&search=0xF684C42E
sil . infiltrated @ net http://www.infiltrated.net

Re: [asterisk-users] Music On Hold

Tilghman Lesher wrote:
> That's true if you use mpg123 for MOH... that's the old way. The recommended
> method now is to use native file format, which is saved per channel. So every
> channel gets the message started from the beginning.
>
>
Aah - cheers for that :) I havnt updated in a while I must admit - must
get round to having a looksee :)

Wayne.

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[asterisk-users] Proximity detection versus GSM receiver

Hi,

Can anyone tell me the pros and cons of Proximity Detection using
bluetooth versus using GSM cell phone with receivers. I like the idea of
calls be transferred to my cell phone when I am away from the office
and I would like to implement such a system.

Thanks

Chuck Bunn

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[asterisk-users] Proximity Detection: Motorola Q + Bluetooth + Asterisk

Hi,

Can anyone tell me if the Motorola Q has its Bluetooth always on like
the IPhone? I want to use the Motorola Q in a Proximity Detection setup
like that described on nerdvittles.com. I know the Treo 650 does not
work well since the display must be on for the bluetooth to be on and
this eats power.

Thanks

Chuck Bunn

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Re: [asterisk-users] Changing contexts "on the fly"

Another option to you might just be easier. Does your PBX ring your
desk phone for a while and then move on to IVR/auto-attendant? If it
already does, do you have a DoNotDisturb button on your phone? That's
pretty straightforward.

The way we do the switch thing is as follows:

exten => *6,1,GotoIf($["${DB(night/enabled)}" = "1"]?2:102)
exten => *6,2,Set(oldval=${DB_DELETE(night/enabled)})
exten => *6,3,System(rm /home/pbx/night_mode)
exten => *6,4,Playback(hcllc-nightmode-off)
exten => *6,5,Hangup
exten => *6,102,Set(DB(night/enabled)="1")
exten => *6,103,System(touch /home/pbx/night_mode)
exten => *6,104,Playback(hcllc-nightmode-on)
exten => *6,105,Hangup

Then, in my incoming from PSTN context, I check like this:

...
exten => s,6,GotoIf($["${DB(night/enabled)}" = "1"]?7:107)
exten => s,7,Goto(attendant-closed,s,1)
exten => s,107,Dial(${RECEPTIONIST},15,tw)
exten => s,108,Dial(${RECEPTIONIST_AND_MOJO},10,tw)
exten => s,109,Goto(attendant-open,s,1)

*6 is for *N, for people to remember (N)ight mode. In my *6 extension,
I create a mutex in a sense, the file called 'night_mode' in /home/pbx
-- this lets me determine if night mode is enabled via external systems,
like those written in PHP for a webpage or something else for a shell
script, maybe as a cron schedule that rings your desk to remind you that
night mode is still on... It is not needed for my incoming context;
that context uses the astdb.

Mojo

Ade Vickers wrote:
> Hi folks,
>
> I've been playing around with an Asterisk server in my office for a few
> weeks now, and I've got it pretty much nailed down the way I want it, which
> is nice.
>
> One of the features I'm using is the ability to switch different contexts in
> & out of the dialplan on a schedule. So, for example, I've got the
> "official" tel number ringing my desk phone between 9.00-17.30 mon-fri; and
> out of those hours any caller gets a recorded message + sent to voicemail.
>
> However, I'm quite often working later than 17.30, and would quite like to
> be able to easily "flick a switch" which tells Asterisk that, actually, I'm
> here in the office, and I'd quite like to receive calls. Currently, I have
> to alter dialplans.conf, comment out a couple of lines & uncomment another;
> save & then re-load the dialplan.
>
> I'm guessing I've got 3 options open to me:
>
> 1) Convert from using the various .conf files, to using a "realtime" config,
> then write a small front-end to the DB so I can access the settings from a
> simple switch on my Windows desktop
> 2) Write some kind of script which I can execute on the Asterisk box which
> makes the same changes I'm currently making manually
> 3) Some other option I've not thought of...
>
>
> What's the panel's opinion on the best way to do this?
>
>
> For info:
> Asterisk 1.4.5 running on Ubuntu 7.04
> Digium-compatible AX100P card providing connection to POTS line
> (this is the one that needs controlling)
> 2 SIP extensions (Grandstream GXP2000)
> Numerous SIPGATE lines (these are configured as I like them already)
>
>
> Much appreciated in advance.
>
> Cheers,
> Ade.
>
> No virus found in this outgoing message.
> Checked by AVG Free Edition.
> Version: 7.5.488 / Virus Database: 269.13.33/1034 - Release Date: 27/09/2007
> 17:00
>
>
>
>
> _______________________________________________
>
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Re: [asterisk-users] Changing contexts "on the fly"

On 9/28/07, Tzafrir Cohen <tzafrir.cohen@xorcom.com> wrote:

On Fri, Sep 28, 2007 at 05:28:21PM +0100, Ade Vickers wrote:
> Hi folks,
>
> I've been playing around with an Asterisk server in my office for a few
> weeks now, and I've got it pretty much nailed down the way I want it, which
> is nice.
>
> One of the features I'm using is the ability to switch different contexts in
> & out of the dialplan on a schedule. So, for example, I've got the
> "official" tel number ringing my desk phone between 9.00-17.30 mon-fri; and
> out of those hours any caller gets a recorded message + sent to voicemail.
>
> However, I'm quite often working later than 17.30, and would quite like to
> be able to easily "flick a switch" which tells Asterisk that, actually, I'm
> here in the office, and I'd quite like to receive calls. Currently, I have
> to alter dialplans.conf, comment out a couple of lines & uncomment another;
> save & then re-load the dialplan.
>
> I'm guessing I've got 3 options open to me:
>
> 1) Convert from using the various .conf files, to using a "realtime" config,
> then write a small front-end to the DB so I can access the settings from a
> simple switch on my Windows desktop
> 2) Write some kind of script which I can execute on the Asterisk box which
> makes the same changes I'm currently making manually
> 3) Some other option I've not thought of...

4) Use a condional dialplan. e.g GotoIfTime or other uses of GotoIf .

Now, add a flag that allows your calls to be routed as either:
1. Default - route according to the schedule
2. Open - give me the calls, to heck with the time
3. Closed - leave me alone. Yes, I know what time it is, but I don't care.

Put this before the GotoIfTime stuff, and it can override however you'd like.

We did this, but added a few fancy things, like ClosedForHurricane mode. It allows us to record a message as to which dates patients have been rescheduled to, says the time of the last update, and a few other goodies.

Have fun with it. You can do just about anything you can dream of. Except solve the halting problem. Ah well...

Re: [asterisk-users] Changing contexts "on the fly"

On Fri, Sep 28, 2007 at 05:28:21PM +0100, Ade Vickers wrote:
> Hi folks,
>
> I've been playing around with an Asterisk server in my office for a few
> weeks now, and I've got it pretty much nailed down the way I want it, which
> is nice.
>
> One of the features I'm using is the ability to switch different contexts in
> & out of the dialplan on a schedule. So, for example, I've got the
> "official" tel number ringing my desk phone between 9.00-17.30 mon-fri; and
> out of those hours any caller gets a recorded message + sent to voicemail.
>
> However, I'm quite often working later than 17.30, and would quite like to
> be able to easily "flick a switch" which tells Asterisk that, actually, I'm
> here in the office, and I'd quite like to receive calls. Currently, I have
> to alter dialplans.conf, comment out a couple of lines & uncomment another;
> save & then re-load the dialplan.
>
> I'm guessing I've got 3 options open to me:
>
> 1) Convert from using the various .conf files, to using a "realtime" config,
> then write a small front-end to the DB so I can access the settings from a
> simple switch on my Windows desktop
> 2) Write some kind of script which I can execute on the Asterisk box which
> makes the same changes I'm currently making manually
> 3) Some other option I've not thought of...

Read the relevant data from a global varaible or from the database in
the dialplan. You can set db entries and/or global variables in various
ways.

4) Use a condional dialplan. e.g GotoIfTime or other uses of GotoIf .

In fact, GotoIfTime seems to be the exact "switch flipper" you need.

--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.cohen@xorcom.com
+972-50-7952406 mailto:tzafrir.cohen@xorcom.com

http://www.xorcom.com

iax:guest@local.xorcom.com/tzafrir

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[asterisk-users] Changing contexts "on the fly"

Hi folks,

I've been playing around with an Asterisk server in my office for a few
weeks now, and I've got it pretty much nailed down the way I want it, which
is nice.

One of the features I'm using is the ability to switch different contexts in
& out of the dialplan on a schedule. So, for example, I've got the
"official" tel number ringing my desk phone between 9.00-17.30 mon-fri; and
out of those hours any caller gets a recorded message + sent to voicemail.

However, I'm quite often working later than 17.30, and would quite like to
be able to easily "flick a switch" which tells Asterisk that, actually, I'm
here in the office, and I'd quite like to receive calls. Currently, I have
to alter dialplans.conf, comment out a couple of lines & uncomment another;
save & then re-load the dialplan.

I'm guessing I've got 3 options open to me:

1) Convert from using the various .conf files, to using a "realtime" config,
then write a small front-end to the DB so I can access the settings from a
simple switch on my Windows desktop
2) Write some kind of script which I can execute on the Asterisk box which
makes the same changes I'm currently making manually
3) Some other option I've not thought of...

What's the panel's opinion on the best way to do this?

For info:
Asterisk 1.4.5 running on Ubuntu 7.04
Digium-compatible AX100P card providing connection to POTS line
(this is the one that needs controlling)
2 SIP extensions (Grandstream GXP2000)
Numerous SIPGATE lines (these are configured as I like them already)

Much appreciated in advance.

Cheers,
Ade.

No virus found in this outgoing message.
Checked by AVG Free Edition.
Version: 7.5.488 / Virus Database: 269.13.33/1034 - Release Date: 27/09/2007
17:00

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