| Summary: | ASTERISK-08044: Client phones nuked with "481 Call leg/transaction does not exist" | ||
| Reporter: | Juan Pablo Abuyeres (jpabuyer) | Labels: | |
| Date Opened: | 2006-11-01 20:09:07.000-0600 | Date Closed: | 2006-11-07 07:25:50.000-0600 |
| Priority: | Minor | Regression? | No |
| Status: | Closed/Complete | Components: | Channels/chan_sip/Interoperability |
| Versions: | Frequency of Occurrence | ||
| Related Issues: | |||
| Environment: | Attachments: | ( 0) chan_sip_ugly_workaround-1.4.0.patch ( 1) debug2.txt | |
| Description: | I'll have to explain with an example as I don't know the internals of SIP. When a SIP client makes a call, if there is no answer and the client hangs up, then a loop of SIP packets starts between the SIP client and the Asterisk server, until the phone is unplugged or shutdown. I've tried it with polycom (several firmware versions) and x-lite. Polycom phones gets nuked because of the cpu usage involved. (I think this is really a 'major' bug, at least for me, because my clients have to reboot their phones several times a day). This behavior started from asterisk-1.2.13 to latest asterisk-trunk. Version 1.2.12.1 was fine. The debug file I am attaching is from the latest trunk. | ||
| Comments: | By: Juan Pablo Abuyeres (jpabuyer) 2006-11-01 20:16:48.000-0600 If I comment a line in chan_sip.c (applying the attached patch), then the problem goes away. I am certain this is not the right solution. I am only attaching this patch to tell the developers where the bug _might_ be. By: Olle Johansson (oej) 2006-11-07 07:02:01.000-0600 I committed a fix to 1.2 svn, please test latest 1.2 svn. Thanks. By: Olle Johansson (oej) 2006-11-07 07:25:36.000-0600 Fixed in 1.2, 1.4 and svn trunk | ||