similar to: Asterisk Crashes

Hi What causes deadlock? Apr 5 14:02:43 WARNING[2413] channel.c: Avoided initial deadlock for '0x82acb10', 10 retries! Apr 5 14:02:43 WARNING[2413] channel.c: Avoided initial deadlock for '0x8298160', 10 retries! Here is the portion of the log: Apr 5 14:02:42 NOTICE[23363] chan_zap.c: Got event 18 (Ring Begin)... Apr 5 14:02:42 VERBOSE[23363] logger.c: -- Executing

Anyone have any idea what's causing this or how to debug it? I'm pretty sure the root cause is with chan_sccp.so, but not sure how to prove it. I recently upgraded from CVS-head to 1.2.1 and the chan_sccp from 12-17-2005. Now, once or twice a week, I get this on the console: Jan 31 10:39:08 WARNING[10586]: channel.c:784 channel_find_locked: Avoided deadlock for '0xbf1013e0',

Anybody out there use Ser along with *? Any advantages disadvantages? Is this even a good idea?

Sorry for all the newbie questions. I really appreciate everyone's help today. Okay I've got outgoing and incoming calls working with no echo. yay! Now I'm having an issue with SIP extension to extension calling. Any time I dial another extension it goes right into voice mail. My extensions.conf is pretty small and rough but, here's what I have right now. Most of it was taken

I have a major problem with SIP redirects, or maybe just do not understand how they are supposed to work. We are using Cisco 7940s and 7960s with SIP version 6.3. Asterisk 1.2.5. A call come in to extension 944 over the IAX trunk. Extension 944 has forward all to extension 904 set on the phone. According to the dialplan. extension 904 should ring for 90 seconds, then ring another extension, and

I am running asterisk 1.4 and the Digium GUI SVN-branch-2.0-r4489. When one phone calls another, I see the following on the console (here, 6223 dials 6123) -- Executing [6123 at DLPN_DefaultDialPlan:1] Macro("IAX2/6223-10489", "stdexten|6123|SIP/6123&IAX2/6123") in new stack -- Executing [s at macro-stdexten:1] Set("IAX2/6223-10489",

I added the playback line to my [macro-stdexten] context but when I dail an extension I don't get the "please hold while I try that extension" message. It just dials the extexsion. Do I have a syntax problem somewhere ? exten => 8005,1,Macro(stdexten,8005,Zap/2) exten => 8006,1,Macro(stdexten,8006,Sip/8006) [macro-stdexten] ; ; Standard extension macro: ; ${ARG1} -

Have a question about implementing Call Rollover with my current extensions.conf configuration. [macro-stdexten] exten => s,1,Dial(${ARG2},20) ; Ring the interface, 20 seconds maximum exten => s,2,Voicemail2(u${ARG1}) ; If unavailable, send to voicemail w/ unavail announce exten => s,3,Goto(default,s,1) ; If they press #, return to start exten =>

Almost two years ago, a change between how AEL code is built into Asterisk dialplan between minor versions made clear the need to provide a sane entry point into AEL subroutines and that's how AELSub() born. With Asterisk 11 release, they way [stdexten] at extensions.conf is invoked changed from Macro to Gosub using the 'missing context feature' and this caused that any stdexten

I thought it was quite easy to implement but I cannot get one-touch recording to work. Here are the changes what I did: I restarted Asterisk after the change (because reload does not work for changes in features.conf). I press *1 on the Polycom IP600 phone to record a conversation but no new wav file appear in /var/spool/asterisk/monitor or elsewhere. Test A: Outside line calling in

Okay everyone - I'm moving away from using sipura 841 phones. I'm starting to test with Polycom IP 501 phones. We plan to upgrade our server to a dual t1 but, for now we have a digium tdm400P with 4 analog lines coming into it. So my question is will upgrading the IP phones with my existing digium tdm400 card be enough to satisfy my users ? or is it really a combo deal needing to

Hi Guys, i spent some time to figure this out (since i love how dialplan is written) but i decided to ask for your help guys. i have two asterisk servers one is 1.2 the other is 1.4, from 1.4 (pbx1) to 1.2 (pbx2) i can leave a voice mail without any pb, but from pbx2 to pbx1 it just hang up. in pbx2 extensions.conf: i am using: exten => 8029,1,Dial(IAX2/pbx1/${EXTEN},20,tTWwr) in pbx1, i

I thought it was quite easy to implement but I cannot get one-touch recording to work. Here are the changes what I did: I restarted Asterisk after the change (because reload does not work for changes in features.conf). I press *1 on the Polycom IP600 phone to record a conversation but no new wav file appear in /var/spool/asterisk/monitor or elsewhere. Any suggestions? Here is the console log:

I am having a problem with an Asterisk server, in that when it is receiving a call from another Asterisk server using an IAX2 trunk the phone rings for 10 ms and then there is a hungup from asterisk and then the phone rings again before another hangup. The funny thing is that after I really hang up on the calling phone it repeats this as if I am still trying to call. Any Ideas?

Hello mailinglist, I want to connect Asterisk with OpenBTS and make a call with a mobile phone. I use: Ubuntu 11.10 + Kernel 3.0.22 GnuRadio 3.3.0 Asterisk 1.8.13 OpenBTS 2.8 Nokia Mobile Phone OpenBTS works and I can send sms from the OpenBTS server to the mobile phone. What I also need is a call between Asterisk and OpenBTS. I have also two soft phones which works with Asterisk. And also

with the below defined in logger.conf on 11.6 cert 6 I am not getting any log message other than notice and warning in any files when doing module reload logger - queue log is the only one that says it restarts *CLI> module reload logger == Parsing '/etc/asterisk/logger.conf': Found Asterisk Queue Logger restarted built fresh box with make samples - added 2 stations, dialing from

I'm having an issue with CDR. Basically, I expect to have all "legs" of a call having the same linkedid and differing only by the sequence value. That does happen, but I'm getting null dst values after doing an attended transfer. I'm not sure if this is a bug or I'm doing something wrong. I'm running Asterisk 13.2.0. Here's the console log, step by step: First,

hi, i'm a bit puzzled because i do not get include and hint to work with the new realtime enginge (cvs-head from 2004-12-09). other things (sipfriends and "normal" extensions) work perfect with the realtime engine. the entries in the static extensions.conf file i used before where: exten => 183,hint,SIP/snom220 exten => 183,1,Macro(stdexten,443,SIP/snom220,183) exten =>

I am integrating a new server (Asterisk 1.8.2.3, DAHDI 2.4.0) with VOIP only trunks, and this server only has soft phones. When I dial an extension and the phone is not registered, I don't get any ring or progress indications, and eventually the Dial() times out and drops into voicemail (as expected). CLI output: -- Executing [s at macro-StdExten:6] Dial("IAX2/barneveld-2036",

Hello, i am trying to enable call forwarding on asterisk 1.6 with asterisk-gui If i set my stdexten as follows (with the lines i marked) everything seems like working. But if i make any change on asterisk-gui and apply it.. it recreates the macro-stdexten and deletes my configuration regarding to it. So where should i add my call-forward configuration??? Where am i making a mistake??