similar to: Avoiding IAX destroy deadlock

Hi there, I am having a problem. It looks like this: Feb 16 15:01:10 WARNING[11122]: chan_iax2.c:5546 socket_read: Call rejected by XXX.XXX.XXX.XXX: No authority found Feb 16 15:01:10 NOTICE[11122]: chan_iax2.c:1375 iax2_destroy: Avoiding IAX destroy deadlock -- Hungup 'IAX2/user/1' Even I have entry in iax.conf for this user as a friend, and * server of this user is already

Hi asterisk users, I have a inssue with incoming calls with wcfxo card, while receiving a call, I?ve configured my dialplan to forward the call to all mi home voip extensions and that works just fine, but while in the call, after a few seconds, the pbx starts the simple switch once more and keeps ringing the voip extensions log as follows:

Hello, I have Asterisk running on a RH9 box; Everything seems to be working as it should, except for Festival. Every time that Festival is called from Asterisk, Asterisk silently shuts down. Festival doesn't give any error indication and Asterisk just plain dies without a peep. Festival was installed per the Wiki, using source and patched with festival-1.4.3-diff; it works fine at the

We have a running Asterisk on a small server (RH 9.0) und were able to make calls via SIP. Since the quality was rather poor (one of us has only a 150/75 kbps DSL connection, iLBC did not work(?)) we tried to setup clients (FireFly) and Asterisk for connections via IAX. In some point seems to be a mistake. Asterisk says: -- Executing Dial("IAX2[83022777@83022777]/2",

Hi, I'm playing with the agent/queue system. Everything work well with v1.0.3. but I want the 'Action: Agents' in the manager API that is only on the CVS version. So i switched to, but now the Queue/Agent system barely work. (my agent don't get the call) Where I can get a 'stable' CVS version? Or maybe, how I can solve my Queue/Problem? here is the detail: 1. I can

Hi: I unseccessfuly tried to place a call from one IAX behind a nat through another IAX with a real IP. I got the following error message on Asterisk Console: Executing Dial("OSS/dsp", "IAX2/wassim/01)") in new stack -- Called wassim/01) Call rejected by 195.112.214.98: No such context/extensionad: -- IAX2/wassim-1 is circuit-busy Aug 8 10:13:38 NOTICE[21759]:

Hi, I am not new to asterisk but this is the first time im using iax protocol. I have always used sip before, but i have heard its better to use iax if i want communication between 2 asterisk servers. I just registered asterisk server1 with asterisk server2 and tried to call server1 from 2 but the call does not pass thru. i dont see any messages on the recieving asterisk cli but the caller

Hi, I'm trying to set up a call-back system using auto-dialout files. I want the call to be terminated when a specific timeout (defined in the .call file) is detected. Both parties should then be hangup. The problem is that the timeout is never detected... How to solve this? Thank you, Pierre .call file ---------- Channel: IAX2/:@xxx.xxx.xxx.xxx/0111111111 Callerid: 111111111

I tried to submit this in R, but not sure if it worked. When running R as sudo, using ctrl-c dumps me to the command line. Hitting exit to exit the terminal window results in R taking 100% of resources. I am using R-2.1.0 on Fedora Core 3. Thanks. Manuel

Hi, I am very closed to implement the IAX2 version in DIAX, but still some issues which I don't know how to handle, maybe someone from this list can help me. Trying to register with the * server as in version 1, I get the following in the * console: NOTICE[1150495040]: File chan_iax2.c, Line 2919 (register_verify): Inappropriate authentication received and in the client: Registration

Hi i was trying to call 17009978275 which is my Fwd line on my notebook from Asterisk and i keep getting this message on the console. -- Executing Dial("Zap/2-1", "IAX2/@iaxtel.com/17009978275@iaxtel") in new stack -- Called @iaxtel.com/17009978275@iaxtel WARNING[1150495040]: File chan_iax2.c, Line 4547 (socket_read): I don't know how to authenticate rob to

An Asterisk box at customer site shows these messages pretty regularly. This causes one way voice, the called party cannot hear the calling party. Apr 7 11:59:44 WARNING[18406] channel.c: Avoided initial deadlock for '0x817b790', 10 retries! Apr 7 14:47:46 WARNING[18406] channel.c: Avoided initial deadlock for '0x81a4380', 10 retries! Apr 7 14:58:53 WARNING[18406] channel.c:

An Asterisk box at customer site shows these messages pretty regularly. This causes one way voice, the called party cannot hear the calling party. Apr 7 11:59:44 WARNING[18406] channel.c: Avoided initial deadlock for '0x817b790', 10 retries! Apr 7 14:47:46 WARNING[18406] channel.c: Avoided initial deadlock for '0x81a4380', 10 retries! Apr 7 14:58:53 WARNING[18406] channel.c:

For some reason we are seeing "Avoiding deadlock for channel" in our Asterisk logs, the logs are getting filled up with an amazing speed around 12000 lines a second, and all of them are "Avoiding deadlock". What could be the potential reason for this to be happening? The Asterisk is used as auto dialler, therefore different channel types are involved SIP, DAHDI, Local's.

Hi guys, First of all, I know that this server must be upgraded asap, I'm just wondering if anyone of you has already faced this problem and , if so, would the upgrade solve my problems... CAPI version 0.6 Asterisk 1.2.5 AGI scripts are being used Main problems: -Dropped Calls - ps aux | grep asterisk shows that asterisk (that is started with safe_asterisk) is generating multiple

Hi, My asterisk sometimes stop responding to iax calls. In the log, I've found this: Mar 29 13:35:45 DEBUG[8386] channel.c: Avoiding initial deadlock for 'IAX2/trunkjstpcn-3' Mar 29 13:35:45 DEBUG[13002] chan_sip.c: update_call_counter(1409) - decrement call limit counter Mar 29 13:35:45 DEBUG[8386] channel.c: Avoiding initial deadlock for 'IAX2/trunkjstpcn-3' Mar 29

Hello list, today I experienced the following for the first time : [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c: Avoiding deadlock for channel '0x114af2c0' [Jan 14 11:26:18] DEBUG[27654] channel.c:

Hi I have 3FXO trunks called ZAP-25,ZAP-26 and ZAP-28 and T1 Channnel bank I get this deadlock problem when 2 incoming call from FXO(Here ZAP-28 and then ZAP-26) wants to dial same channel (Here ZAP-1). In this senario ZAP-1 first answer ZAP-28 and thne ZAP-26 wants to call ZAP-1 but it time out and goto voicemail after that ZAP-1 try to reach ZAP-26 call by puting ZAP-28 on HOLD During

I have looked all over the site(s) for help. But heres the problem. Im missing something. In coming works fine from FreeWorld via IAX. But when Dialing out i get: May 13 13:42:01 WARNING[1150495040]: chan_iax2.c:5256 socket_read: I don't know how to authenticate iaxtel to 65.39.205.121 my IAX.conf if as follows [general] port=5036 register => ######:xxxxxxxxxxxxx@iax2.fwdnet.net

Hi, I'm using the following setup : Alice ---- IPPhone ------<LAN>----- Media gateway ----<PSTN> ------- Phone ---- Bob For certain calls, users complains about echo : they can ear their own voice in their handset, though media gateway echo cancel is turned on. I'm wondering how this echo cancelation engine is supposed to work. My understanding of echo is that most probably,