search for: ast_channel_walk_lock

Dec 1 12:08:43 WARNING[6189]: channel.c:495 ast_channel_walk_locked: Avoided deadlock for 'SIP/2502-6303', 10 retries! Dec 1 12:08:44 WARNING[6189]: channel.c:495 ast_channel_walk_locked: Avoided deadlock for 'SIP/2502-6303', 10 retries! Dec 1 12:08:44 WARNING[6189]: channel.c:495 ast_channel_walk_locked: Avoided deadlock for 'SIP/2502-6303&...

All out of the blue I get these errors? Any Ideas why Please help May 26 09:54:28 WARNING[3964]: channel.c:507 ast_channel_walk_locked: Avoided initial deadlock for 'SIP/301-b9d0', 10 retries! May 26 09:54:30 WARNING[3964]: channel.c:507 ast_channel_walk_locked: Avoided initial deadlock for 'SIP/301-b9d0', 10 retries! May 26 09:54:33 WARNING[3964]: channel.c:507 ast_channel_walk_locked: Avoided initial deadlock f...

...ore the CLI command restart now didn't even work .. had to kill the process before it worked again... myasterisk*CLI> show channels Channel (Context Extension Pri ) State Appl. Data 0 active channel(s) 65 active call(s) Jun 27 16:22:06 WARNING[20313]: channel.c:531 ast_channel_walk_locked: Avoided initial deadlock for 'SIP/mistered-ddb7', 10 retries! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050627/a7a197d1/attachment.htm

...18181950250B@vicimail.vicimarketinggroup.com> > > Content-Type: text/plain; charset="iso-8859-1" > > Hello, > > I'd suggest posting a bug if you haven't already and if you have purchased > any Digium products I would recommend calling them as well. The > ast_channel_walk_locked error is a rare and hard to diagnose problem and the > bug trackers and Digium would be the best people to help you. > > It might also help to know what kind of SIP client you are using (hard/soft > phone) and what the network you are on is like. > > MATT--- > > > -----...

...ging call. the debug says: SIP/marco-73a0 is ringing -- SIP/marco-73a0 is ringing -- SIP/marco-73a0 is ringing -- SIP/marco-73a0 is ringing -- Executing PickupChan("SIP/edevena-a940", "SIP/marco") in new stack Oct 13 12:35:48 WARNING[1675]: channel.c:513 ast_channel_walk_locked: Avoided initial deadlock for 'SIP/edevena-a940', 10 retries! -- No channel found SIP/marco. -- SIP/marco-73a0 is ringing the problems seems in ast_channel_walk_locked. Will someone help on this matter? -------------- next part -------------- An HTML attachment was scrubbed.....

...es. Calling directly from Gatekeeper is ok, so problem comes from h323 asterisk channel. I saw few people telling about similar problem begining of month, does they solve their problem? I also grab an 1-0_stable version to test if this problem is also there but I get an "undefined symbol: ast_channel_walk_locked" in app_groupcount.so module, so asterisk failed. What does it mean? Thanks for any hint. -- Dash

...odd, when I call the number (1235690251) I get this in the logs, and just dead air, no ringing or anything: main-nuvoip*CLI> Jan 4 12:46:43 NOTICE[1089849920]: chan_sip.c:7889 sip_poke_noanswer: Peer 'jimball' is now UNREACHABLE! Jan 4 12:46:46 WARNING[1159722432]: channel.c:472 ast_channel_walk_locked: Avoided initial deadlock for 'SIP/123.123.94.4-44c269e0', 10 retries! main-nuvoip*CLI> main-nuvoip*CLI> -- Executing Goto("SIP/123.123.94.4-44cc0ca0", "jimballboutiques|1235690251|1") in new stack -- Goto (jimballboutiques,1235690251,1) -- Execu...

...ive me no dialtone.. here if I dial, it successfully completes a call. 2. While trying to get the dialing to work. The MGCP channel hung. the call came in IAX.. I hung up the originating channel (IAX side) and it never reported the hangup.. now I see: Mar 16 14:12:12 WARNING[8904]: channel.c:523 ast_channel_walk_locked: Avoided deadlock for 'MGCP/aaln/1@192.168.10.5-1', 10 retries! over and over.. Show channels shows: IAX2/utxaustin01@utxaustin01/2 (cp-austin 5125551234 1 ) Ring Dial MGCP/aaln/1@192.168.10.5-1 soft hangup on the IAX side dosen't do anything soft hangup on the MG...

Hi I am getting this error after installing and configuration of asterisk. Aug 24 17:53:50 WARNING[9924]: channel.c:472 ast_channel_walk_locked: Avoided initial deadlock for 'Zap/3-1', 10 retries! I have upgraded asterisk to latest version but still receiving the same error. Can someone helpme to resolve this issue. Regards, Shafqat Hamid -------------- next part -------------- An HTML attachment was scrubbed... URL:...

Hi all, i am using CVS-HEAD-08/13/04-12:00:00-BRI-stuffed-0.1.0-RC4a with Cisco ATA-186 3.1.1 atamgcp We are used to make an special ;) blind transfer like (Flash)Number(Hangup before anyone answers or ring). Then * crashes (see below) if the man in the middle is an cisco-ata-186-mgcp If one waits until the last one rings, then hangup, everything is fine. If one waits until the last one

Has anyone here done this? I got it compiled just fine but when I make a call I do not get any audio going either way. The * box is not behind any sort of firewall or nat. My H323 client (gnomemeeting) is behind NAT but I have it set up properly to work through NAT and it will talk correctly with my other regular x86 box running H323. One odd thing I note is that when looking at the UDP traffic

For an inbound call center with 4 T1s and 30-50 agents on you would do just fine with a single, one-processor machine. We have handled more than this on a single P4 server although we use astGUIclient instead of Asterisk queues, but the load is very similar. I would recommend a Sangoma Quad T1 card because they are about 30% more efficient than Digium T1 cards. When you say that you need to