[asterisk-users] PBX Extension Help

[Saeed Akhtar]

hi all,

I have a little problem. I'm trying to configure a2billing
(asterisk2billing) with asterisk. Everything done successfully but when I
try to call following error occur
   "WARNING[9690]: pbx.c:3170 pbx_extension_helper: No application
'DeadAGI,a2billing.php' for extension (a2billing, 456,3)
and it hang ups the call. Can someone please tell me why this error
occuring. My extension.conf script is as following.


[a2billing]
; CallingCard application
exten => _X.,1,Answer
exten => _X.,2,Wait,2
exten => _X.,3,DeadAGI,a2billing.php
exten => _X.,4,Wait,2
exten => _X.,5,Hangup

[did]
; CallingCard application
exten => _X.,1,DeadAGI(a2billing.php|1|did)

while a2billing.php is places in /agi-bin/a2billing.php


Regards,

Saeed Akhtar
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Re: [asterisk-users] PBX Extension Help

[Alex Balashov]

When passing arguments to applications you must use parentheses.

Try:

  exten => _X.,3,DeadAGI(a2billing.php)

You can omit parentheses when calling applications with no arguments,  
e.g.


   exten => s,1,Answer

... but not when there are parameters.

--
Sent from mobile device

On Jan 1, 2010, at 7:01 AM, Saeed Akhtar   
wrote:



hi all,

I have a little problem. I'm trying to configure a2billing  
(asterisk2billing) with asterisk. Everything done successfully but  
when I try to call following error occur
   "WARNING[9690]: pbx.c:3170 pbx_extension_helper: No application  
'DeadAGI,a2billing.php' for extension (a2billing, 456,3)
and it hang ups the call. Can someone please tell me why this error  
occuring. My extension.conf script is as following.


[a2billing]
; CallingCard application
exten => _X.,1,Answer
exten => _X.,2,Wait,2
exten => _X.,3,DeadAGI,a2billing.php
exten => _X.,4,Wait,2
exten => _X.,5,Hangup

[did]
; CallingCard application
exten => _X.,1,DeadAGI(a2billing.php|1|did)
while a2billing.php is places in /agi-bin/a2billing.php


Regards,

Saeed Akhtar

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[asterisk-users] Happy New year 2010

[Patterson]

An great an happy new year 2010 for all of you ...

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Re: [asterisk-users] Monitoring SIP & Skype connections

[Administrator TOOTAI]

Myles Wakeham a écrit :
> [...]  Are there tools or 
> add-ons available for this that will email me when a SIP registration 
> goes offline?
>
> Any suggestions for this would be greatly appreciated.
>
>   
Hi Myles,

first, best wishes to the list for this new 2010 year.

To answer your question, you can run a cron job each x minutes 
-supposing that you qualify your provider- launching a script like

#!/bin/bash

isOffLine=`/usr/sbin/asterisk -rx 'sip show peers'| grep  
| grep OK`

if [ $isOffLine = "" ]; then
# start what you want, for instance do a sip reload
fi

exit 0

-- 
Daniel

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[asterisk-users] SIP Listen Multiple Ports

[Shariq Khan]

Is there any way to listen SIP on multiple ports on asterisk. Is is possible
to define in sip.conf in the following way.

sip.conf

[general]
port = 5060
port = 5090


Regards,
Shariq Khan
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Re: [asterisk-users] PBX Extension Help

[Warren Selby]

Also, shouldn't the .php script be located in "/var/lib/asterisk/agi-bin"?

On Fri, Jan 1, 2010 at 7:31 AM, Alex Balashov wrote:

> When passing arguments to applications you must use parentheses.
>
> Try:
>
>   exten => _X.,3,DeadAGI(a2billing.php)
>
> You can omit parentheses when calling applications with no arguments, e.g.
>
>exten => s,1,Answer
>
> ... but not when there are parameters.
>
> --
> Sent from mobile device
>
> On Jan 1, 2010, at 7:01 AM, Saeed Akhtar 
> wrote:
>
> hi all,
>
> I have a little problem. I'm trying to configure a2billing
> (asterisk2billing) with asterisk. Everything done successfully but when I
> try to call following error occur
>"WARNING[9690]: pbx.c:3170 pbx_extension_helper: No application
> 'DeadAGI,a2billing.php' for extension (a2billing, 456,3)
> and it hang ups the call. Can someone please tell me why this error
> occuring. My extension.conf script is as following.
>
> [a2billing]
> ; CallingCard application
> exten => _X.,1,Answer
> exten => _X.,2,Wait,2
> exten => _X.,3,DeadAGI,a2billing.php
> exten => _X.,4,Wait,2
> exten => _X.,5,Hangup
>
> [did]
> ; CallingCard application
> exten => _X.,1,DeadAGI(a2billing.php|1|did)
>
> while a2billing.php is places in /agi-bin/a2billing.php
>
>
> Regards,
>
> Saeed Akhtar
>
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>



-- 
Thanks,
--Warren Selby
http://www.selbytech.com
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Re: [asterisk-users] SIP Listen Multiple Ports

[Warren Selby]

On Fri, Jan 1, 2010 at 10:34 AM, Shariq Khan wrote:

> Is there any way to listen SIP on multiple ports on asterisk. Is is
> possible to define in sip.conf in the following way.
>
> sip.conf
>
> [general]
> port = 5060
> port = 5090
>
>
>
>
Depending on the version of asterisk you are using, you can set the port
that asterisk binds to using the following commands in sip.conf:

1.6.x:
udpbindaddr = x.x.x.x:5061

1.4.x:
bindport = 5061


-- 
Thanks,
--Warren Selby
http://www.selbytech.com
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Re: [asterisk-users] PBX Extension Help

[Alex Balashov]

Fact.

On 01/01/2010 01:06 PM, Warren Selby wrote:

> Also, shouldn't the .php script be located in "/var/lib/asterisk/agi-bin"?
>
> On Fri, Jan 1, 2010 at 7:31 AM, Alex Balashov  > wrote:
>
> When passing arguments to applications you must use parentheses.
>
> Try:
>
>exten => _X.,3,DeadAGI(a2billing.php)
>
> You can omit parentheses when calling applications with no
> arguments, e.g.
>
> exten => s,1,Answer
>
> ... but not when there are parameters.
>
> --
> Sent from mobile device
>
> On Jan 1, 2010, at 7:01 AM, Saeed Akhtar  > wrote:
>
>> hi all,
>>
>> I have a little problem. I'm trying to configure a2billing
>> (asterisk2billing) with asterisk. Everything done successfully but
>> when I try to call following error occur
>> "WARNING[9690]: pbx.c:3170 pbx_extension_helper: No application
>> 'DeadAGI,a2billing.php' for extension (a2billing, 456,3)
>> and it hang ups the call. Can someone please tell me why this
>> error occuring. My extension.conf script is as following.
>> [a2billing]
>> ; CallingCard application
>> exten =>  _X.,1,Answer
>> exten =>  _X.,2,Wait,2
>> exten =>  _X.,3,DeadAGI,a2billing.php
>> exten =>  _X.,4,Wait,2
>> exten =>  _X.,5,Hangup
>>
>> [did]
>> ; CallingCard application
>> exten =>  _X.,1,DeadAGI(a2billing.php|1|did)
>> while a2billing.php is places in /agi-bin/a2billing.php
>>
>>
>> Regards,
>>
>> Saeed Akhtar
>>
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>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>> 
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>
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> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
>
> --
> Thanks,
> --Warren Selby
> http://www.selbytech.com
>
>
>
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-- 
Alex Balashov - Principal
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct  : (+1) (678) 954-0671

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Re: [asterisk-users] PBX Extension Help

[Gergo Csibra]

Friday, January 1, 2010, 7:12:54 PM, Alex wrote:

> On 01/01/2010 01:06 PM, Warren Selby wrote:

>> Also, shouldn't the .php script be located in "/var/lib/asterisk/agi-bin"?

> Fact.

And on a live channel must use AGI instead of DeadAGI.

And man should not topposting on a maillist...

-- 
Best regards,
 Gergomailto:csi...@gmail.com


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Re: [asterisk-users] SIP Listen Multiple Ports

[Shariq Khan]

I m using Asterisk 1.4 , can i bind multiple ports to SIP at a time

like

bindport = 5060,5061 OR

bindport = 5060
bindport = 5090

I want, asterisk to listen SIP on multiple ports. so that users where SIP
port 5060 blocked, can easily register to asterisk by using an alternate
port.

Shariq Khan

On Fri, Jan 1, 2010 at 11:12 PM, Warren Selby  wrote:

> On Fri, Jan 1, 2010 at 10:34 AM, Shariq Khan wrote:
>
>> Is there any way to listen SIP on multiple ports on asterisk. Is is
>> possible to define in sip.conf in the following way.
>>
>> sip.conf
>>
>> [general]
>> port = 5060
>> port = 5090
>>
>>
>>
>>
> Depending on the version of asterisk you are using, you can set the port
> that asterisk binds to using the following commands in sip.conf:
>
> 1.6.x:
> udpbindaddr = x.x.x.x:5061
>
> 1.4.x:
> bindport = 5061
>
>
> --
> Thanks,
> --Warren Selby
> http://www.selbytech.com
>
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Re: [asterisk-users] PBX Extension Help

[Shariq Khan]

Better to post your problem on Asterisk2Billing Forum
http://forum.asterisk2billing.org/

Warren Selby mentioned right that your
agi is missing.

And dont change DeadAGI and ignore this error, as the script is still not
updated to use AGI for local channels.

Shariq Khan

On Fri, Jan 1, 2010 at 11:41 PM, Gergo Csibra  wrote:

> Friday, January 1, 2010, 7:12:54 PM, Alex wrote:
>
> > On 01/01/2010 01:06 PM, Warren Selby wrote:
>
> >> Also, shouldn't the .php script be located in
> "/var/lib/asterisk/agi-bin"?
>
> > Fact.
>
> And on a live channel must use AGI instead of DeadAGI.
>
> And man should not topposting on a maillist...
>
> --
> Best regards,
>  Gergomailto:csi...@gmail.com
>
>
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[asterisk-users] AudioCodes MP-114 2xFXS/2xFXO - FXO not working correctly

[Joseph]

I have AudioCodes 2xFXO / 2xFXS but can not make the FXO port to work 
correctly; I can dial out on one FXO port or the other FXO, but not on both.
It depends on Sorting in : Hunt Group Setting (Ascending, Descending)
If setting is set to a "Cyclic Ascending" I can dial out on FXO port every 
second call.

Can anybody share their setting how is it done in mix environment FXS/FXO in 
AudioCodes MediaPack?

-- 
Joseph

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Re: [asterisk-users] PBX Extension Help

[Steve Edwards]

Un-top-posting...

> On 01/01/2010 01:06 PM, Warren Selby wrote:

>> Also, shouldn't the .php script be located in 
>> "/var/lib/asterisk/agi-bin"?

On Fri, 1 Jan 2010, Alex Balashov wrote:

> Fact.

To be more specific, unless the full path ("starts with a slash") is 
specified, the specified path is relative to the value of astagidir set in 
asterisk.conf.

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

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Re: [asterisk-users] SIP Incoming / Inbound not working for Broadvoice (Asterisk PBX 1.6.1.6)

[Doug]

At 18:36 12/31/2009, Qurba Joog wrote:
>Thanks for the reply Doug. Do you mean I should add "nat=yes" on my 
>nat'd extensions?

Affirmative.




>On Wed, Dec 30, 2009 at 10:57 PM, Doug 
><d...@natel.net> wrote:
>At 18:22 12/30/2009, Qurba Joog wrote:
>You are correct.. I had the correct context on my current production 
>configuration I just copied from an older saved file.. So the 
>[enterbroadvoice] has a context of incoming and incoming is defined 
>in the extensions.con. But still have the same problem with incoming 
>jumping directly into Broadvoice VM. I even changed it to IAX2 extension.
>
>Any help would be appreciated
>
>
>I don't this for any of your peers:
>
>nat=yes
>
>
>
>
>
>On Fri, Dec 25, 2009 at 10:43 AM, Qurba Joog 
><qurbaj...@gmail.com> 
>wrote:
>Thanks very much for your reply Nguyen. I read and re-read the url 
>you sent.. Sorry I'm new to this but are you saying the [incoming] 
>context is wrong? Please indicate what is wrong.
>
>If you look at the SIP.conf [enter_broadvoice] I have [incoming] 
>context defined and that is the one I have called out in the extensions.conf
>
>Thanks.
>
>
>On Fri, Dec 25, 2009 at 5:20 AM, Nguyen Quang Tri 
><<http://kihote.am>kihote.am@<http://gmail.com>gmail.com>
> 
>wrote:
>Wrong context for incoming,
>you can read
><http://www.voip-info.org/wiki/view/Asterisk+config+sip.conf>http://www.voip-info.org/wiki/view/Asterisk+config+sip.conf
>and
><http://www.voip-info.org/wiki/view/IPKall>http://www.voip-info.org/wiki/view/IPKall
>
>2009/12/25 --[ UxBoD ]-- 
><ux...@splatnix.net>
>
>- "Qurba Joog" 
><qurbaj...@gmail.com> 
>wrote:
>| Hello,
>|
>| Please forgive me if I'm repeating this post. I have searched and 
>looked for similar problem with a solution but have not see a similar one.
>|
>| My outgoing SIP and other channels work fine but the 
>incoming/inbound SIP call goes straight to Broadvoice voicemail. I 
>see that Broadvoice is registered when I look at the SIP registry. I 
>have turned on SIP Debug and it is below.
>|
>| Anyone know why even when SIP has registered I do not see incoming calls?
>|
>| Thanks,
>|
>|
>| --extensions.conf
>| [global]
>|
>| [general]
>| bindport=5060
>| bindaddr = 0.0.0.0
>| deny=<http://0.0.0.0/0.0.0.0>MailScanner 
>has detected a possible fraud attempt from "0.0.0.0" claiming to be 
>MailScanner warning: numerical links are often 
>malicious:<http://0.0.0.0/0.0.0.0> 
>0.0.0.0/0.0.0.0
>| 
>externhost=<http://xyz.dyndns.org>xyz.dyndns.org
>| localnet = 
><http://192.168.1.0/255.255.255.0>MailScanner
> 
>has detected a possible fraud attempt from "192.168.1.0" claiming to 
>be MailScanner warning: numerical links are often 
>malicious:<http://192.168.1.0/255.255.255.0> 
>192.168.1.0/255.255.255.0
>
>
>| disallow=all
>| allow=ulaw
>| allow=gsm
>| delayreject=yes
>| nochecksums=no
>| allowguest=no
>| delayreject=yes
>| pedantic=no
>|
>| register => 
>703xxxy...@sip.broadvoice.com:s<http://ecurepassword:703xxxy...@sip.broadvoice.com/5000>ecurepassword:703xxxy...@sip.broadvoice.com/5000
>
>|
>| [5000]
>| type=friend
>| context=internal-phones
>| secret=xxx
>| qualify=yes
>| host=dynamic ; behind nat
>| dtmfmode=rfc2833
>|
>| [5002]
>| type=friend
>| context=internal-phones
>| secret=test
>| qualify=yes
>| host=dynamic ; behind nat
>| nat=yes
>| dtmfmode=rfc2833
>|
>| [enter_broadvoice]
>| type=peer
>| user=phone
>| 
>host=<http://sip.broadvoice.com>sip.broadvoice.com
>| 
>fromdomain=<http://sip.broadvoice.com>sip.broadvoice.com
>
>| fromuser=703XXX
>| secret=securepassword
>| username=703XXX
>| insecure=very
>| ;insecure=port,invite
>| context=incoming
>| authname=703XXX
>| dtmfmode=inband
>| dtmf=inband
>| ;Disable canreinvite if you are behind a NAT
>|
>| canreinvite=no
>|
>| extensions.conf
>|
>| [globals]
>|
>| [general]
>|
>| autofallthrough=yes
>|
>|
>| [incoming_calls]
>|
>| exten => 1703XXX,1,Dial(SIP/5000)
>|
>| [internal-phones]
>|
>| include => outgoing
>| exten => 5000,1,Dial(SIP/5000,20)
>| exten => 5002,1,Dial(SIP/5002,20)
>|
>|
>| [outgoing]
>|
>| exten => _X.,1,NoOp()
>| exten => _X.,n,Dial(SIP/enter_broadvoice/${EXTEN})
>|
>| SIP Registry

Re: [asterisk-users] Inquiry:Asterisk different codec schemes?

[hadi motamedi]

On Thu, Dec 31, 2009 at 12:10 PM, Kevin P. Fleming wrote:

>  hadi motamedi wrote:
>
> > Can you please let me know if we can have different codec schemes for
> > audio codec in & audio codec out ? I mean , in one application , we
> > can have our audio codec input set to G.711 a-law and our audio codec
> > output set to G.711 u-law . I am facing with an application that calls
> > for such a settings .
>
> Asterisk does not support asymmetric codec configurations.
>
> --
> Kevin P. Fleming
> Digium, Inc. | Director of Software Technologies
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
> skype: kpfleming | jabber: kpflem...@digium.com
> Check us out at www.digium.com & www.asterisk.org
>
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Sorry . I didn't get the point clearly . In the SIP Invite message , it says
"my audio endpoint is IP x.x.x.x port x, and I can use codecs A,B,C". The
remote endpoint responds with a 200 OK, saying "my audio stream is at IP
y.y.y.y port y, and I choose codec B". Can you please do me favor and let me
know if my understanding is right or not ?
Thank you
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Re: [asterisk-users] Inquiry:Asterisk sip ?

[hadi motamedi]

On Thu, Dec 31, 2009 at 6:40 AM, hadi motamedi  wrote:

> Dear All
> Please be informed that my Asterisk has sip connection to an external
> sip server but the sip outgoing call will be disconnected for some
> unknown reasons . Please find attached the debug log . Can you please
> do me favor and let me know what is the problem that causes the call
> to immediately being dropped when the called party goes offhook ?
> Thank you
>


Dear All
Please be informed that the problem came from "canreinvite=yes" settings .
It changed to "canreinvite=no" and the problem solved out.
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[asterisk-users] verifying correct loading of VPMADT032

[ramadasan]

Hi,

I have two Digium Cards
http://www.digium.com/en/products/digital/te121.php, on connected to PRI &
the other to EPABX.

We have felt problem similar to what is mentioned at,
https://issues.asterisk.org/view.php?id=15498

I have removed the VPMADT032, from the card connected to EPABX. I would
like to ensure that VPMADT032 on card connected to PRI is loading
properly.

I have just enabled wcte12xp in /etc/dahdi/modules file.

Here is the output of my dmesg command: http://pastebin.com/mcc1925e

Any help would be appreciated.

--

Sri Ramadoss M


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