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From: da...@debsinc.com
To: asterisk-users@lists.digium.com
Date: Thu, 30 Sep 2010 14:59:38 -0500
Subject: Re: [asterisk-users] a2billing
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From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Flavio Miranda
Sent: Thursd
[ramais]
include => internalinclude => externalinclude => conference
[internal]
exten => 3000,1,DIAL(SIP/3000,10)exten => 3000,2,VoiceMail(3000,u)
exten => 3003,1,DIAL(SIP/3003,30)exten => 3003,2,VoiceMail(3003,u)
exten => 3004,1,DIAL(SIP/3004,10)exten => 3004,2,VoiceMail(3004,u)
exten => 3005,1
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From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Flavio Miranda
Sent: Thursday, September 30, 2010 2:44 PM
To: Asterisk Asterisk
Subject: [asterisk-users] a2billing
Hi all,
I am trying to integrate a2b with asterisk 1
Hi all,
I am trying to integrate a2b with asterisk 1.6, but ,when i try to do external
call, I receive this mensage:
-- Executing [01221341...@ramais:1] AGI("SIP/3000-b5ba6e80", "a2billing.php,2")
in new stack-- Launched AGI Script /var/lib/asterisk/agi-bin/a2billing.php
-- AGI Script
Hi Guys,
Sorry Kevin for that it was not on purpose (i didn't pay attention to what
"reply" is putting as emails).
actually I feel so dump, i didn't pay attention at all when i was
downloading, but thanks a lot. i did install the right version and it's
showing up info about modules, so it's fine.
By popular request, we've convinced someone from the VoIP Abuse
Project to join us tomorrow at noon on VUC. I think many of you will
be interested in this topic, so please come by, join in and ask
questions.
http://vuc.me for all connection info and links to VoIP Abuse Project
A couple of other f
On Thu, Sep 30, 2010 at 11:46 AM, khalid touati wrote:
> thanks for replies,
> I am using Asterisk 1.6.2.11
> and components res_fax-1.4_1.2.1-x86_64 and
> res_fax_digium-1.4_1.2.1-barcelona_64.
> (amd 64 bit machine)
> actually I am not aware that there is version which include fax.
> for rebuild
Hi,
I have the same extension registered with multiple softphones on
multiple servers, i.e.
100-lo...@hosta
100-lo...@hostb
and on both hostA and hostB I have the extension in extension.conf
exten => 100,1,Answer()
exten => 100,n,Dial(100-local)
When from softphone registered as 100-lo...@host
On 09/30/2010 10:46 AM, khalid touati wrote:
> thanks for replies,
Please do not send personal replies to messages on the mailing list.
Reply to the mailing list. Thanks.
> I am using Asterisk 1.6.2.11
> and components res_fax-1.4_1.2.1-x86_64 and
> res_fax_digium-1.4_1.2.1-barcelona_64.
> (amd 6
thanks for replies,
I am using Asterisk 1.6.2.11
and components res_fax-1.4_1.2.1-x86_64 and
res_fax_digium-1.4_1.2.1-barcelona_64.
(amd 64 bit machine)
David:
actually I am not aware that there is version which include fax.
for rebuilding with manager support that would be great if you could give
On Fri, 2010-02-26 at 15:21 +0800, Zhang Shukun wrote:
> 2010/2/26 Tilghman Lesher :
> > On Friday 26 February 2010 00:09:55 Warren Selby wrote:
> >> On Thu, Feb 25, 2010 at 11:26 PM, Zhang Shukun wrote:
> >> > [Feb 26 13:18:08] WARNING[16077]: config.c:2025 find_engine: Realtime
> >> > mapping fo
On 09/30/2010 09:51 AM, khalid touati wrote:
> Hi List,
> I did follow the procedure to install Free Fax for Asterisk successfully
> till i came accross this isssue: i can't load the fax module:
>
> pbx3*CLI> module load res_fax_digium.so
> Unable to load module res_fax_digium.so
> Command 'module
On Thu, Sep 30, 2010 at 10:51 AM, khalid touati wrote:
> Hi List,
> I did follow the procedure to install Free Fax for Asterisk successfully
> till i came accross this isssue: i can't load the fax module:
>
> pbx3*CLI> module load res_fax_digium.so
> Unable to load module res_fax_digium.so
> Comma
Hi List,
I did follow the procedure to install Free Fax for Asterisk successfully
till i came accross this isssue: i can't load the fax module:
pbx3*CLI> module load res_fax_digium.so
Unable to load module res_fax_digium.so
Command 'module load res_fax_digium.so' failed.
[Sep 30 10:50:12] WARNING[
- "Danny Dias" wrote:
> That solved my problem, thank you very much...but now i'm having another
> problem, when the server starts, it doesn't start asterisk automatically,
> should i change the start script?
Your system *should* start Wanpipe, DAHDI, then Asterisk (in that order). What
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Thursday, September 30, 2010 9:04 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Intercom with Dial() works, but not w
Hi!
> Can you tell me how I can get my Snom 320 auto-answer the call when I
> use the Page()-command ?
Configure a special identity on the SNOM that is set to auto-answer in
the phone's configuration. Or consider to use multicast instead of Page()
if your network topology doesn't stand in the
Hello list,
this works :
exten => _*XXX*,n,SIPAddHeader("Call-Info:\; answer-after=0")
exten => _*XXX*,n,Dial(SIP/${SIPACCOUNT})
The phone auto-answers the call...
this does not work :
exten => _*XXX*,n,SIPAddHeader("Call-Info:\; answer-after=0")
exten => _*XXX*,n,Page(SIP/${SIPACCOUNT})
Th
Between 9:30AM and 10:00AM CDT (GMT-5) today, the services below will
experience short outages:
downloads.digium.com
downloads.asterisk.org
bamboo.asterisk.org
packages.asterisk.org
svn.digium.com
svn.asterisk.org
issues.asterisk.org
reviewboard.asterisk.org
We apologize for any inconvenience thi
Thanks Tim
That solved my problem, thank you very much...but now i'm having another
problem, when the server starts, it doesn't start asterisk automatically,
should i change the start script?
2010/9/30 Tim Nelson
> - "Danny Dias" wrote:
> >I'm getting a KErnel Pannic every time i restart
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bryant
Zimmerman
Sent: Thursday, September 30, 2010 7:00 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Asterisk 1.6.2.13 Audio Prompts St
Hello everyone.
I have server with 2E1 PCI card, asterisk 1.4.35, dahdi 2.4.0, libpri
1.4.12-beta2. One PRI trunk looks to PSTN and take a clocksource from
telco. Another trunk looks to PBX with DECT system.
Some outgoing calls from asterisk to PSTN drops. The last message that
exists before
On 09/30/2010 12:16 PM, Jonas Kellens wrote:
Hello list,
I get the following error :
pbx_extension_helper: No application Page for extension
Apparently I have no timing source installed.
But I thought that Dahdi did not need to be installed for timing ?!
And that there is some internal timing
- "Danny Dias" wrote:
> I'm getting a KErnel Pannic every time i restart the server, what could be
> happening?
> I just make: "shutdown -r now" and the server gets Kernel Panic. I'have to go
> on site and press the power button
I'd be willing to bet Wanpipe is attempting to stop whil
> how can I go from *100* to 100 ?
>
> I know I can do something like ${EXTEN:1} but that way I only get rid of just
> one *.
${EXTEN:1:-1} removes the first and last characters of ${EXTEN}.
--
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-- Bandwidth and Colocation Pro
Version 1.6.2.13 is having issues with audio prompts dieing. When users
call in to get voicemail the prompts start and then stop about 6 to 10
seconds in. On hold music plays for 6 to 10 seconds and then stops. In meet
me conference rooms hold music will stop about 6 to 10 seconds in. Audio
pla
On Thursday 30 Sep 2010, Danny Dias wrote:
> Hello,
>
> I'm getting a KErnel Pannic every time i restart the server, what could be
> happening?
>
> I just make: "shutdown -r now" and the server gets Kernel Panic. I'have to
> go on site and press the power button
>
> Here you have my sotware version
Hi,
Mostly this Problem is Hardware issue . check your server Hardwares.
On Thu, Sep 30, 2010 at 3:39 PM, Danny Dias wrote:
> Hello,
>
> I'm getting a KErnel Pannic every time i restart the server, what could be
> happening?
>
> I just make: "shutdown -r now" and the server gets Kernel Panic. I
Hello list,
I get the following error :
pbx_extension_helper: No application Page for extension
Apparently I have no timing source installed.
But I thought that Dahdi did not need to be installed for timing ?! And
that there is some internal timing in Asterisk 1.6.2.10 ?
Kind regards,
Jona
Hello,
I'm getting a KErnel Pannic every time i restart the server, what could be
happening?
I just make: "shutdown -r now" and the server gets Kernel Panic. I'have to
go on site and press the power button
Here you have my sotware versions:
Asterisk 1.4.24.1
DAHDI Tools Version - 2.1.0.2
DAHDI
On Thu, 30 Sep 2010, Jonas Kellens wrote:
> Hello list,
>
> how can I go from *100* to 100 ?
>
> I know I can do something like ${EXTEN:1} but that way I only get rid of just
> one *.
${EXTEN:1:3}
That gives 3 characters from an offset of 1.
Read the file channelvariables.txt in the doc direct
${EXTEN:1:3}
http://astbook.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/
asterisk-CHP-5-SECT-3.html#asterisk-CHP-5-SECT-3.6.3
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas
Kellens
Sent: 3
In Asterisk, the funny thing is if a certain component is not installed
properly or the config file has a typo or something, this will be shown
up as a non-existent command in Asterisk command line interface.
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:ast
Thanks Shaun.
Unfortunately, I am still using zaptel.
Is there a similar command in zaptel?
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
> boun...@lists.digium.com] On Behalf Of Shaun Ruffell
> Sent: Thursday, 30 September 2010 1:00 AM
> To:
Hello list,
I need some light regarding the way asterisk is handling the
SIP Registration method:
I have an asterisk 1.6.0.22 and a UAC that sends REGISTER
requests without the Authentication part in the sip message. The UAC expects a
401 reply to create the cor
Hello list,
how can I go from *100* to 100 ?
I know I can do something like ${EXTEN:1} but that way I only get rid of
just one *.
Kind regards,
Jonas.
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