similar to: channel.c:780 channel_find_locked: Avoided deadlock

REPOSTED with MORE Info and Modified Subject Line: -------------------------------------------------------- I am using one of the Minute Provider to dial out USA numbers. Now in one of my process, we need to Dial IVR and the enter DTMF digit and then it connects to the automated IVR. When I dial out the IVR directly using Xlite and VOIP Mins provider , it works perfectly. but when In try from

My Inbound calls lands , but line get disconnect in exactly 3 secs. Here goes my extension.conf setting : [from-ipkall] exten => 901835,1,Ringing ; call ringing exten => 901835,2,Wait(1) ; Wait 1 second for CID delivery from PRI exten => 901835,3,Answer ; Answer the line exten =>

== Manager 'sendcron' logged off from 127.0.0.1 vicidialnow*CLI> dial 919545090201 -- Executing AGI("OSS/dsp", "agi://127.0.0.1:4577/call_log") in new stack -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 -- Executing Dial("OSS/dsp", "SIP/19545090201 at sip203||tTor") in new stack -- Called 19545090201 at sip203 Feb 2 13:36:38

Hello! I clear remarks in Makefile: DEBUG_THREADS = -DDEBUG_THREADS -DDETECT_DEADLOCKS But same things in CLI: Aug 30 18:16:31 WARNING[22292]: channel.c:780 channel_find_locked: Avoided initial deadlock for '0x82f2fe0', 9 retries! -- Zap/32-1 is proceeding passing it to Zap/31-1 -- Zap/32-1 is ringing -- Accepting call from '2177' to '7141278' on channel

*Code:* == Manager 'sendcron' logged off from 127.0.0.1 -- Executing Playback("Local/91441425477394 at default-b9f2,1", "sip-silence") in new stack -- Playing 'sip-silence' (language 'en') -- Executing AGI("Local/91441425477394 at default-b9f2,1", "agi:// 127.0.0.1:4577/call_log") in new stack -- AGI Script

My dialer works perfectly , but whenever I dial a number manually from xlite and press a Key like 6055 for DTMF , line gets disconnected. Line gets DC as soon as I press any key from xlite What could be the issues ? I tried the SAME VOIP from another center and Its Ok there. I tried the Same dialer Xlite over Static IP, problem is there. I tried the same number from other Dialer , it works

*I am getting below CLI in my asterisk :* == Manager 'sendcron' logged off from 127.0.0.1 -- Executing AGI("SIP/cc101-b7910cc0", "agi://127.0.0.1:4577/call_log") in new stack -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 -- Executing Dial("SIP/cc101-b7910cc0", "SIP/Sama203/119545090201||tTor") in new stack --

What are these? Nov 22 09:35:23 WARNING[7127]: channel.c:787 channel_find_locked: Avoided deadlock for '0xf6c06778', 10 retries! Nov 22 09:35:24 WARNING[7127]: channel.c:787 channel_find_locked: Avoided deadlock for '0xf6c06778', 10 retries! Nov 22 09:35:24 WARNING[7127]: channel.c:787 channel_find_locked: Avoided deadlock for '0xf6c06778', 10 retries! Nov 22 09:35:25

In asterisk 1.2 branch SVN 51363 zaptel svn 1980 libpri svn 393 addons svn 332 My equipment is a Via EPIA minit-itx CN10000 1.2ghz, 1gb ram and a tdm400p (4fxo). A call comes from zap, a SIP ulaw receives the call, talks for a while and when SIP users tries to park the call, then dozens of... WARNING[3853]: channel.c:781 channel_find_locked: Avoided initial deadlock for '0x91bb840', 10

Hi i have 20 people calling agents calling when ever they calling i get this below error May 30 00:46:57 WARNING[2840]: channel.c:785 channel_find_locked: Avoided deadlock for '0x8b2f50', 10 retries! and the voice go choppy, and voice breakages iam using Latest SVN, any suggestion to come over this problem ram -------------- next part -------------- An HTML attachment was scrubbed...

I never so this error. I am using H323 with Asterisk 1.2.6 Any idea what can be the problem? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060401/2e8ad498/attachment.htm

Hi sir, I was installed Goautodial server and I have b410p BRI card. BRI card showing OK with dahdi_tool, this NT mode. whenever I am dialing from server i am not able to connect the call . in Cli below mention warning is comming . please what is the mistake with me . help me Executing [0559566768 at default:1] AGI("Console/dsp", "agi:// 127.0.0.1:4577/call_log") in new

Hi all, I have an Astribank connected to Asterisk 1.4. I'm setting up extensions and I have a problem with inbound calls to zap extensions. The phone at 65 rings once and then the line gets hung up. If I pick up the phone really fast, it works. Any suggestions? I have the following setup: [from-pstn] exten => 207582401,1,Dial(Zap/65,30) CLI shows me this: -- Accepting call from

Sir, i am using goautodial server , bri card is showing ok but when i try to call that showing below , This configuration is in doing in dubai , so kindly help me how can connet the call from this , what is my mistake is in this :::chan-dahdi.conf [channels] #include dahdi-channels.conf language=en context=default usecallerid=yes hidecallerid=yes callwaiting=yes usecallingpres=yes

I tried this http://lists.digium.com/pipermail/asterisk-users/2008-January/203615.html But I am NOT getting call in asterisk. SIP.conf file : _________________ [general] port = 5060 bindaddr = 0.0.0.0 context = default externhost=59.160.44.21 localnet=192.168.0.2/255.255.255.0 ; register SIP account on remote machine if using SIP trunks ; register => testSIPtrunk:test at 10.10.10.16:5060 ;

Hello. I have an Asterisk server (ViciDialNow) set up behind NAT. I can manage to make outbound calls, but the communication drops off after 30 seconds or so. I'd really appreciate having some assistance from the mailing list on this issue. So, I'm having an Asterisk server behind a firewall and Zoiper softphones on SIP connecting to Asterisk on the same local area network. The

All my calls are getting DECLINED when I am trying from xlite : CLI shows : May 18 00:00:32 WARNING[4617]: channel.c:2781 ast_channel_make_compatible: No pa th to translate from SIP/cc101-b790c1d8(4) to SIP/sip19-090e87d8(256) May 18 00:00:32 WARNING[4617]: app_dial.c:1628 dial_exec_full: Had to drop call because I couldn't make SIP/cc101-b790c1d8 compatible with

I am getting this Error on my Asterisk. How to solve it ? "ERROR[2654]: chan_sip.c:11355 handle_request: Missing Cseq. Dropping this SIP message, it's incomplete." -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090115/cb953962/attachment.htm

I'm looking into the dialplan specifics: tleilax:~ # tleilax:~ # cat /etc/asterisk/extensions.conf [general] static=yes writeprotect=no [globals] CONSOLE=Console/dsp ; Console interface for demo TRUNK=DAHDI/r1 ; Trunk interface TRUNKX=DAHDI/r2 ; 2nd trunk interface TRUNKIAX=IAX2/ASTtest1:test at 10.10.10.16:4569 ; IAX trunk

Hi there, Im getting this notice in CLI, but the call quality is okey, Im using digium TE406 and asterisk 1.2.4. here are the CLI actual logs: -- Executing SetAccount("Local/50015308467418@default-ca2e,2", "XXXXXX") in new stack -- Executing AGI("Local/50015308467418@default-ca2e,2", "call_log.agi|50015308467418") in new stack -- Launched AGI