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: Thursday, September 30, 2010 2:44 PM
To: Asterisk Asterisk
Subject: [asterisk-users] a2billing
Hi all,
I am trying to integrate a2b with asterisk 1.6, 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 a2billing.php completed, returning 0
-- Auto fallthrough, channel 'SIP/3000-b5ba6e80' status is 'UNKNOWN
Anybody know what could be happen?
Att,
Flavio Roberto Miranda
MSN:flaviormira...@hotmail.com
Skype: flaviormiranda
It appears that the AGI is completing successfully and you have a dialplan
issue after that. Please post the dialplan snippet.
Att,
Flavio Roberto Miranda
MSN:flaviormira...@hotmail.com
Skype: flaviormiranda
Asterisk is doing "exactly what you told it to do". You dial 3000, it drops
into a2billing, gives you the "auto fall-through" message because you don't
have any further "dialplan" such as a hangup (h) context.
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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 => 3005,1,DIAL(SIP/3005,10)exten => 3005,2,VoiceMail(3005u)
exten => 1001,1,DIAL(SIP/1001,10)exten => 1001,2,VoiceMail(1001u)
exten => 3999,1,VoiceMailMain($(CALLERID(num))
[external]
;exten => _0NXXX,1,DIAL(SIP/10.201.201.254/${EXTEN},20,rt);exten =>
_0NXXX,1,AGI,a2billing.phpexten => _X.,1,AGI(a2billing.php,2)
Att,
Flavio Roberto Miranda
MSN:flaviormira...@hotmail.com
Skype: flaviormiranda
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: Thursday, September 30, 2010
2:44 PM
To: Asterisk Asterisk
Subject: [asterisk-users]
a2billing
Hi all,
I am trying to integrate a2b with asterisk 1.6, 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
a2billing.php completed, returning 0
-- Auto fallthrough, channel
'SIP/3000-b5ba6e80' status is 'UNKNOWN
Anybody know what could be happen?
Att,
Flavio Roberto Miranda
MSN:flaviormira...@hotmail.com
Skype: flaviormiranda
It
appears that the AGI is completing successfully and you have a dialplan issue
after that. Please post the dialplan snippet.
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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 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 a2billing.php completed, returning 0
-- Auto fallthrough, channel 'SIP/3000-b5ba6e80' status is 'UNKNOWN
Anybody know what could be happen?
Att,
Flavio Roberto Miranda
MSN:flaviormira...@hotmail.com
Skype: flaviormiranda
It appears that the AGI is completing successfully and you have a dialplan
issue after that. Please post the dialplan snippet.
--
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[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
-- AGI Script a2billing.php completed, returning 0--
Auto fallthrough, channel 'SIP/3000-b5ba6e80' status is 'UNKNOWN
Anybody know what could be happen?
Att,
Flavio Roberto Miranda
MSN:flaviormira...@hotmail.com
Skype: flaviormiranda
--
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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. David: i see what you mean, you're right it's there,thank you for your help! On Thu, Sep 30, 2010 at 12:09 PM, David Backeberg wrote: > 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 rebuilding with manager support that would be great if you could give > me > > a link to know about that, cause i've never done it ! > > If you're building from source, do a make menuconfig > > then you get the ability to do select/deselct on the individual > applications. > > One is called app_fax > That contains the built-in fax support. > > It requires that you have already pre-built SpanDSP. > > And also, it looks like you cannot NOT compile in manager support, so > my original idea seems wrong. > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- Abdullah -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 features and announcements are scheduled after that segment, including an Astricon update and maybe even something about a mysterious book I have heard about. Call in from 11:40 AM SIP:200...@login.zipdx.com - prefer g722 and accept g711 - Skype:vuc.me - Call widget on the site during conference hours. See you there! /r -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unable to load fax modules
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 rebuilding with manager support that would be great if you could give me > a link to know about that, cause i've never done it ! If you're building from source, do a make menuconfig then you get the ability to do select/deselct on the individual applications. One is called app_fax That contains the built-in fax support. It requires that you have already pre-built SpanDSP. And also, it looks like you cannot NOT compile in manager support, so my original idea seems wrong. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 I call (1...@hostb) what I see on the softphone on host B is "100-lo...@hostb is contacting you" and the call gets routed on the local calls context instead of the incoming call context. I expected to see "100-lo...@hosta is contacting you" instead. Is this behavior something that can be avoided? I thought it would be normal to have two asterisk's in e.g. two companies serving the same extensions... Thank you, Matteo -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 bit machine) You are trying to use FAX modules for Asterisk 1.4.x with Asterisk 1.6.2.11. Did you use the FAX download selector to get links to the proper modules to use for your version of Asterisk? -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 give me a link to know about that, cause i've never done it ! Thank you guys! On Thu, Sep 30, 2010 at 11:23 AM, Kevin P. Fleming wrote: > 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 res_fax_digium.so' failed. > > [Sep 30 10:50:12] WARNING[5427]: loader.c:429 load_dynamic_module: Error > > loading module 'res_fax_digium.so': > > /usr/lib/asterisk/modules/res_fax_digium.so: undefined symbol: > manager_event > > [Sep 30 10:50:12] WARNING[5427]: loader.c:797 load_resource: Module > > 'res_fax_digium.so' could not be loaded. > > > > any help will be much appreciated!! > > It will be very hard to help you with the information you provided; at a > minimum we need to know what version of Asterisk and of the FAX modules > you tried to use. > > -- > Kevin P. Fleming > Digium, Inc. | Director of Software Technologies > 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA > skype: kpfleming | jabber: kflem...@digium.com > Check us out at www.digium.com & www.asterisk.org > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- Abdullah -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 : > > 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 for 'sippeers' found to engine 'mysql', but the engine is not > >> > available-- > >> > >> Is MySQL running and all the proper values set in the appropriate files? > > > > Does the config name in extconfig.conf right after the word "mysql" exist > > as a > > section in res_mysql.conf? > > the section in extconfig.conf is : > > > sipusers => mysql,asterisk,sip_buddies > sippeers => mysql,asterisk,sip_buddies > > > and section in res_mysql.conf is : > > > > [asterisk] > ;dbhost = 127.0.0.1 > dbname = asterisk > dbuser = root > dbpass = net263 > dbport = 3306 > ;dbsock = /tmp/mysql.sock > ;dbsock = /var/run/mysqld/mysqld.sock > requirements=createclose ; or createclose or createchar > I know this is months later but I've just been having the same problem and fixed it by adding mysql dev packages and then recompiling the asterisk addons Hope this helps Ish > > > > > -- > > Tilghman Lesher > > Digium, Inc. | Senior Software Developer > > twitter: Corydon76 | IRC: Corydon76-dig (Freenode) > > Check us out at: www.digium.com & www.asterisk.org > > > > -- > > _ > > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > -- > Best regards, > Sucan > -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 res_fax_digium.so' failed. > [Sep 30 10:50:12] WARNING[5427]: loader.c:429 load_dynamic_module: Error > loading module 'res_fax_digium.so': > /usr/lib/asterisk/modules/res_fax_digium.so: undefined symbol: manager_event > [Sep 30 10:50:12] WARNING[5427]: loader.c:797 load_resource: Module > 'res_fax_digium.so' could not be loaded. > > any help will be much appreciated!! It will be very hard to help you with the information you provided; at a minimum we need to know what version of Asterisk and of the FAX modules you tried to use. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unable to load fax modules
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 > Command 'module load res_fax_digium.so' failed. > [Sep 30 10:50:12] WARNING[5427]: loader.c:429 load_dynamic_module: Error > loading module 'res_fax_digium.so': > /usr/lib/asterisk/modules/res_fax_digium.so: undefined symbol: manager_event > [Sep 30 10:50:12] WARNING[5427]: loader.c:797 load_resource: Module > 'res_fax_digium.so' could not be loaded. > > any help will be much appreciated!! Don't know for certain, but my guess is you didn't build in AMI or manager support with your asterisk build. Rebuild with manager support and you should be able to find the undefined symbol. And if you're building from scratch, I don't know the reason why you wouldn't use the built-in app_fax.so -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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] WARNING[5427]: loader.c:429 load_dynamic_module: Error loading module 'res_fax_digium.so': /usr/lib/asterisk/modules/res_fax_digium.so: undefined symbol: manager_event [Sep 30 10:50:12] WARNING[5427]: loader.c:797 load_resource: Module 'res_fax_digium.so' could not be loaded. any help will be much appreciated!! -- Abdullah -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Kernel Panic When restarting the server
- "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 flavor of Linux are you running? If CentOS/Redhat/Fedora use 'chkconfig --list' to ensure those services are set to startup. On Debian, you can use 'rcconf'. --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 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 rings and does not auto-answer the call...
Can you tell me how I can get my Snom 320 auto-answer the call when I use
the Page()-command ?
Kind regards,
Jonas.
While I think Philipp's reply is a workable answer, I am curious why two
seemly similar commands would produce different outcomes. Have you followed
these two snippets in CLI (with and without SIP debug) to see why the header
works with Dial() but not with Page()?
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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 way of multicasting. Philipp -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 rings and does not auto-answer the call...
Can you tell me how I can get my Snom 320 auto-answer the call when I
use the Page()-command ?
Kind regards,
Jonas.
--
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[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 this may cause. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 > - "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 while Asterisk is still > using it. It's a known problem. Put the following into > /etc/wanpipe/scripts/stop > > #!/bin/sh > /etc/init.d/asterisk stop > sleep 2 > /etc/init.d/asterisk stop > > Then chmod +x it. When Wanpipe attempts to stop, it will shut down Asterisk > first, bypassing a panic situation. > > --Tim > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 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 playback in IVR's start to play and then stops. It happes with both g729 and g711 calls. This does not happen on every call but more then 50%. This is a big issue any ideas? I need to fix this ASAP. Thanks Bryan More information please - Compile flags, O/S, etc. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 hanging up process is: DEBUG[28467] channel.c: Got a FRAME_CONTROL (8) frame on channel DAHDI/... This frame come when call already established. So, when it come, the call drops. "FRAME_CONTROL (8)" means 'Congestion' according to 'frame.h'. I have already set debug 6, verbose 6 and enabled EXTENSIVE debugging on span, but couldn't find incoming frame from telco with information of 'Congestion' on this channel. I want to debug this message. I want to know where the root of my problem. And I'm sure that it's only my problem. That's why I didn't create issue ticket on bug tracker. So default methods of debug didn't show me control frames. I have a call log: http://pastebin.mozilla-russia.org/107089 Part of the full log file at the moment when this FRAME have been got: http://pastebin.mozilla-russia.org/107090 Part of the full log file from start of the call to drop: http://pastebin.com/MphaCkiV I have from 3 to 5 call drops in hour so it's reproduce periodically : http://pastebin.com/rdnYR8dU http://pastebin.com/KUJDPd3C I'm sorry, but I can't remember what E1 card placed in server. But it could be Digium or OpenVox. Here is output of some commands: lspci: 02:00.0 Communication controller: Digium, Inc. Wildcard TE210P dual-span T1/E1/J1 card 3.3V (rev 02) Control: I/O+ Mem+ BusMaster+ SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx- Status: Cap- 66MHz- UDF- FastB2B- ParErr- DEVSEL=medium >TAbort- SERR- GSI 16 (level, low) -> IRQ 16 wct4xxp :02:00.0: Found TE2XXP at base address d010, remapped to f90d4000 wct4xxp :02:00.0: DMA memory base of size 2048 at f6829000. Read: f6829400 and Write f6829000 wct4xxp :02:00.0: Firmware Version: c01a wct4xxp :02:00.0: Burst Mode: On wct4xxp :02:00.0: FALC Framer Version: 2.1 or earlier wct4xxp :02:00.0: Board ID: 00 wct4xxp :02:00.0: Reg 0: 0x36829400 wct4xxp :02:00.0: Reg 1: 0x36829000 wct4xxp :02:00.0: Reg 2: 0xd018 wct4xxp :02:00.0: Reg 3: 0x wct4xxp :02:00.0: Reg 4: 0x wct4xxp :02:00.0: Reg 5: 0xd0100014 wct4xxp :02:00.0: Reg 6: 0xc01a wct4xxp :02:00.0: Reg 7: 0x1f00 wct4xxp :02:00.0: Reg 8: 0x wct4xxp :02:00.0: Reg 9: 0x wct4xxp :02:00.0: Reg 10: 0xd0100028 IRQ 16/wct2xxp: IRQF_DISABLED is not guaranteed on shared IRQs wct4xxp :02:00.0: Found a Wildcard: Wildcard TE210P dahdi_hardware: pci::02:00.0 wct4xxp+ d161:0210 Wildcard TE210P As I think, it's really Digium TE210P. But I don't think, it's a pci card problem because call drops exist only on outgoing calls and 98% DIDs is mobile phone numbers. Any suggestions how to see this frame and who was the sender? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 timing in Asterisk 1.6.2.10 ? Kind regards, Jonas. To answer my own question : I think I need linux kernel 2.6.25 or higher to have the "internal timer". As I'm on CentOS 5.5 and kernel version 2.6.18-194, I guess I still need Dahdi. Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Kernel Panic When restarting the server
- "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 while Asterisk is still using it. It's a known problem. Put the following into /etc/wanpipe/scripts/stop #!/bin/sh /etc/init.d/asterisk stop sleep 2 /etc/init.d/asterisk stop Then chmod +x it. When Wanpipe attempts to stop, it will shut down Asterisk first, bypassing a panic situation. --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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}.
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[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 playback in IVR's start to play and then stops. It happes with both g729 and g711 calls. This does not happen on every call but more then 50%. This is a big issue any ideas? I need to fix this ASAP. Thanks Bryan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 1.4.24.1 > DAHDI Tools Version - 2.1.0.2 > DAHDI Version: 2.1.0.4 > libpri version: 1.4.10.1 > WANPIPE Release: 3.5.4 > > IS there something that i shoud check? > > Best Regards! Use chmod -x to prevent the Asterisk and Dahdi startup scripts from running. (And make sure the modules aren't being loaded by udev or anything else.) Then see if you can get it to shut down cleanly. If so, that means it's one of the Dahdi modules causing it. -- AJS -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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 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 1.4.24.1 > DAHDI Tools Version - 2.1.0.2 > DAHDI Version: 2.1.0.4 > libpri version: 1.4.10.1 > WANPIPE Release: 3.5.4 > > IS there something that i shoud check? > > Best Regards! > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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, Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 Version: 2.1.0.4 libpri version: 1.4.10.1 WANPIPE Release: 3.5.4 IS there something that i shoud check? Best Regards! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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.
Gordon
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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: 30 September 2010 08:31
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [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.
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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 [mailto:asterisk-users- > boun...@lists.digium.com] On Behalf Of mis...@efro.us > Sent: Thursday, 30 September 2010 6:57 AM > To: asterisk-users@lists.digium.com > Subject: [asterisk-users] can't get libpri/PRI to work, missing PRI > commands > > I'm putting together a PBX using a TE420P card configured for E1s that is > connected to an Errickson MTS. successfully compiled and installed libpri > 1.4.11.3, DAHDI 2.3.0.1+2.3.0 and Asterisk 1.6.2.9. everythings seems to > be working. SIP phone to SIP phone (POLYCOM) calls work fine however, > network calls do not. When I went to debug PRI, the only command that > showed up when I did ">CLI core show help pri" was 'pri intense debug > span' which seemed strange to begin with and when I did ">CLI pri intense > debug span 1", I got something strange like "'pri set debug 2 span 1' is > not a valid command". Tried reinstalling everything but keep getting the > same result. Also, when I try to make call from SIP phone to a wireless > phone and vise versa on our GSM network, I get something like "Call to > extension rejected because the extension is not found in context > POLYCOM#" but it is definitely in the extensions.conf with a line "exten > => 1,1,Dial(dahdi/g1/#xxx)". Please help... > > Thanks > > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Use modprobe to find E1/T1 jumper setting > onPRI card > > On 09/29/2010 02:52 AM, Lee, John (Sydney) wrote: > > Do you mean that if I could define 30 channels in span 1 for > > example, then span 1 is set to E1? > > > > If not, then it is T1. > > > > You could also see this information in the "type" output from > dahdi_scan. For example before configuring a span: > > # ./dahdi_scan > [1] > active=yes > alarms=UNCONFIGURED > description=Wildcard TE122 Card 0 > name=WCT1/0 > manufacturer=Digium > devicetype=Wildcard TE122 > location=PCI Bus 15 Slot 05 > basechan=1 > totchans=24 > irq=90 > type=digital-T1 > syncsrc=0 > lbo=0 db (CSU)/0-133 feet (DSX-1) > coding_opts=B8ZS,AMI > framing_opts=ESF,D4 > coding= > framing= > > > -- > Shaun Ruffell > Digium, Inc. | Linux Kernel Developer > 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA > Check us out at: www.digium.com & www.asterisk.org > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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 correct auth request. When it receives an empty REGISTER asterisk does basically 2 things: 1. returns an 503 <--- SIP read from UDP://1.2.3.5:5060 ---> REGISTER sip:1.2.3.4;user=phone SIP/2.0 Via:SIP/2.0/UDP 1.2.3.5:5060;rport;branch=z9hG4bK62F7FE99 From:;tag=3D090004 To: Call-ID:3D0807BC@ CSeq:53246 REGISTER Max-Forwards:70 Contact: Allow:INVITE,CANCEL,ACK,REGISTER,INFO,BYE,OPTIONS,REFER,NOTIFY,SUBSCRIBE,MESSAGE Expires:60 User-Agent:OTHER UA Content-Length:0 <-> --- (12 headers 0 lines) --- <--- Transmitting (no NAT) to 1.2.3.5:5060 ---> SIP/2.0 503 Server error Via: SIP/2.0/UDP 1.2.3.5:5060;rport;branch=z9hG4bK62F7FE99;received=1.2.3.5 From: ;tag=3D090004 To: ;tag=as7bb7cbf6 Call-ID: 3D0807BC@ CSeq: 53246 REGISTER User-Agent: Asterisk 1.6.0.22 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 2. returns a 401 with the nonce <--- SIP read from UDP://1.2.3.5:5060 ---> REGISTER sip:1.2.3.4;user=phone SIP/2.0 Via:SIP/2.0/UDP 1.2.3.5:5060;rport;branch=z9hG4bK62F7FEA5 From:;tag=3D090008 To: Call-ID:3D0807BC@ CSeq:53248 REGISTER Max-Forwards:70 Contact: Allow:INVITE,CANCEL,ACK,REGISTER,INFO,BYE,OPTIONS,REFER,NOTIFY,SUBSCRIBE,MESSAGE Expires:60 User-Agent:OTHER UA Content-Length:0 <-> --- (12 headers 0 lines) --- Sending to 1.2.3.5 : 5060 (no NAT) <--- Transmitting (no NAT) to 1.2.3.5:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 1.2.3.5:5060;rport;branch=z9hG4bK62F7FEA5;received=1.2.3.5 From: ;tag=3D090008 To: ;tag=as0811b2ee Call-ID: 3D0807BC@ CSeq: 53248 REGISTER User-Agent: Asterisk 1.6.0.22 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="myrealm", nonce="6a08a09c" Content-Length: 0 The questions are: * is asterisk supposed to return the 401 to a REGISTER method which lacks the Auth Info? I saw that it returns 401 to REGISTER methods that have the wrong nonce and this behavior should be correct. * the Register method should always contain the last nonce and the auth part? * why the 503 message? Thanks in advance, Alex -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[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.
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