Summary: | ASTERISK-07472: Transfering on a local channel doesn't work properly | ||
Reporter: | jmls (jmls) | Labels: | |
Date Opened: | 2006-08-07 02:54:51 | Date Closed: | 2006-11-07 14:12:51.000-0600 |
Priority: | Minor | Regression? | No |
Status: | Closed/Complete | Components: | Channels/chan_local |
Versions: | Frequency of Occurrence | ||
Related Issues: | |||
Environment: | Attachments: | ( 0) console20060807-0850.txt ( 1) console-20060807-1805.txt ( 2) console-20060914.-0930.non.txt ( 3) console-20060914.-0930.withn.txt | |
Description: | With the following scenario: (--> are comments) Zap call comes into SIP phone A --> Zap talks to SIP A SIP A starts transfer to Local/B/n --> Zap has MOH (correct) Local/B calls SIP C --> SIP A Talks to Sip C SIP A completes transfer to SIP C --> Zap hears just silence --> SIP C now has the MOH ! If zap hangs up, asterisk core dumps ****** ADDITIONAL INFORMATION ****** note is this is related to issue ASTERISK-7469, but the segfault does not happen anymore because that has been fixed. I have attached a SIP debug and Debug 4 console output | ||
Comments: | By: jmls (jmls) 2006-08-07 02:55:18 dialplan is as follows: I have made the dialplan as simple as I can to reproduce this crash - I was trying local channels for another problem that U have, and normally have dynamic members of a queue using local channels [from-isdn] exten => _4446XX,1,goto(common|77${EXTEN:4}|1) [from-sip] include => common [common] exten => _7XXX,1,Answer() exten => _7XXX,n,Dial(SIP/${EXTEN},120,g) exten => _7XXX,n,Hangup() exten => 7709,1,Answer() exten => 7709,n,Dial(Local/7710@from-sip/n) exten => 7709,n,Hangup() By: jmls (jmls) 2006-08-07 02:55:33 SIP A (extension 7706) is a cisco 7960 with 7.4, SIP C (extension 7710) is a xlite softphone (corrected as per following note) By: Serge Vecher (serge-v) 2006-08-07 09:51:22 in your last note (0049903), by SIP B (extension 7710) you meant SIP C, right? I saw some strange errors in the logs, like avoiding initial deadlock messages or this: RTCP SR transmission error, rtcp halted Interrupted system call Can you please do a clean checkout of svn trunk? By: jmls (jmls) 2006-08-07 11:07:46 I did mean SIP C, sorry. did clean checkout. Changed extension to Local/7707 instead of Local/7710 (in the office now). Same problem. After transfer, zap hears nothing, SIP/7707 hears MOH By: Serge Vecher (serge-v) 2006-08-07 11:28:20 hmm, the output looks very different. Did you change logging settings? By: jmls (jmls) 2006-08-07 12:00:32 Yes, it looks as if I didn't do the sip debug this time. Do you want that as well ? By: jmls (jmls) 2006-08-07 12:06:35 console output with sip debug enabled uploaded By: jmls (jmls) 2006-08-10 11:23:59 anything else needed to move this forward ? By: Joshua C. Colp (jcolp) 2006-08-16 12:58:49 I've labbed this up using SIP/Local/IAX2 and it works fine, so it might be isolated to chan_zap - but unfortunately I have no hardware to test with. I'll see if I can get someone else to test though. By: jmls (jmls) 2006-08-24 17:10:38 anyone else able to confirm this ? I'll try it again with the latest svn trunk tomorrow By: jmls (jmls) 2006-09-10 13:05:29 I can confirm that this is still an issue with r42621 By: jmls (jmls) 2006-09-10 16:26:35 A bit more information: if you remove the /n things work just fine: exten => 7709,n,Dial(Local/7710@from-sip/n) ; is broken exten => 7709,n,Dial(Local/7710@from-sip) ; works for me By: Serge Vecher (serge-v) 2006-09-13 09:29:21 julian: let's see what the sip debug looks like with and without the /n option. By: jmls (jmls) 2006-09-14 03:39:11 both files uploaded as requested console-20060914.-0930.non.txt is with "No N" (this works) console-20060914.-0930.withn.txt is with "/n" (this is broken) By: jmls (jmls) 2006-10-03 09:33:10 this is still an issue with 1.4 r44240 By: jmls (jmls) 2006-10-21 02:19:08 *nudge* : file, have you thought of anything more about this ? Thanks. By: Joshua C. Colp (jcolp) 2006-11-07 14:12:51.000-0600 Fixed in 1.4 as of revision 47284 and trunk as of revision 47285. Thanks jmls! |