[asterisk-users] Go from *100* to just 100

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. -- _ -- Bandwidth and Colocation Provided by

[asterisk-users] SIP Registrations

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

Re: [asterisk-users] Use modprobe to find E1/T1 jumper setting onPRI card

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:

Re: [asterisk-users] can't get libpri/PRI to work, missing PRI commands

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

Re: [asterisk-users] Go from *100* to just 100

${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:

Re: [asterisk-users] Go from *100* to just 100

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 directory.

[asterisk-users] Kernel Panic When restarting the server

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

[asterisk-users] Asterisk 1.6.2.10 Internal timing

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,

Re: [asterisk-users] Kernel Panic When restarting the server

Hi, Mostly this Problem is Hardware issue . check your server Hardwares. On Thu, Sep 30, 2010 at 3:39 PM, Danny Dias ing.diasda...@gmail.com 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

Re: [asterisk-users] Kernel Panic When restarting the server

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 versions:

[asterisk-users] Asterisk 1.6.2.13 Audio Prompts Stopping

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

Re: [asterisk-users] Go from *100* to just 100

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}. -- _ -- Bandwidth and Colocation

Re: [asterisk-users] Kernel Panic When restarting the server

- Danny Dias ing.diasda...@gmail.com 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

Re: [asterisk-users] Asterisk 1.6.2.10 Internal timing

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

[asterisk-users] channel.c: Got a FRAME_CONTROL (8) frame on channel DAHDI

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

Re: [asterisk-users] Asterisk 1.6.2.13 Audio Prompts Stopping

_ 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

Re: [asterisk-users] Kernel Panic When restarting the server

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 tnel...@rockbochs.com - Danny Dias ing.diasda...@gmail.com wrote: I'm

[asterisk-users] Unscheduled service outage for various Asterisk community services

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

[asterisk-users] Intercom with Dial() works, but not with Page()

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}) The phone

Re: [asterisk-users] Intercom with Dial() works, but not with Page()

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

Re: [asterisk-users] Intercom with Dial() works, but not with Page()

_ 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

Re: [asterisk-users] Kernel Panic When restarting the server

- Danny Dias ing.diasda...@gmail.com 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

[asterisk-users] Unable to load fax modules

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]

Re: [asterisk-users] Unable to load fax modules

On Thu, Sep 30, 2010 at 10:51 AM, khalid touati khalidtou...@gmail.com 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

Re: [asterisk-users] Unable to load fax modules

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 load

Re: [asterisk-users] Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available

On Fri, 2010-02-26 at 15:21 +0800, Zhang Shukun wrote: 2010/2/26 Tilghman Lesher tles...@digium.com: On Friday 26 February 2010 00:09:55 Warren Selby wrote: On Thu, Feb 25, 2010 at 11:26 PM, Zhang Shukun bit...@gmail.com wrote: [Feb 26 13:18:08] WARNING[16077]: config.c:2025 find_engine:

Re: [asterisk-users] Unable to load fax modules

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

Re: [asterisk-users] Unable to load fax modules

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 64

[asterisk-users] Same extension on multiple servers confusion

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...@hosta

Re: [asterisk-users] Unable to load fax modules

On Thu, Sep 30, 2010 at 11:46 AM, khalid touati khalidtou...@gmail.com 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

[asterisk-users] Friday 12 Noon EDT: VoIP Abuse Project

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

Re: [asterisk-users] Unable to load fax modules

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.

[asterisk-users] a2billing

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 --

Re: [asterisk-users] a2billing

_ 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

Re: [asterisk-users] a2billing

[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 =

Re: [asterisk-users] a2billing

_ From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Thu, 30 Sep 2010 14:59:38 -0500 Subject: Re: [asterisk-users] a2billing _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Flavio Miranda Sent: