| Summary: | ASTERISK-13916: One Way audio on incoming calls from SIP provider trunk | ||
| Reporter: | Jeff Phelps (blargman) | Labels: | |
| Date Opened: | 2009-04-07 17:22:20 | Date Closed: | 2011-06-07 14:00:30 |
| Priority: | Blocker | Regression? | Yes |
| Status: | Closed/Complete | Components: | Channels/chan_sip/General |
| Versions: | Frequency of Occurrence | ||
| Related Issues: | |||
| Environment: | Attachments: | ( 0) 14850.diff ( 1) 2009-04-07-core-debug-not-working-1.6.1.0-rc4.txt ( 2) 2009-04-07-core-debug-working-1.6.1-rc1.txt ( 3) 2009-04-07-sip-debug-not-working-1.6.1.0-rc4.txt ( 4) 2009-04-07-sip-debug-working-1.6.1-rc1.txt ( 5) 2009-04-14-sip-debug-not-working-1.6.1.0-rc3.txt ( 6) 2009-04-16-sip-debug-outgoing-working-1.6.1.0-rc4.txt | |
| Description: | in 1.6.1-rc1 i have perfectly working sip trunking... in rc4 i have outgoing working just fine, but incoming calls on the sip trunk are one way... i.e. They can hear me, but I can't hear them... | ||
| Comments: | By: Jeff Phelps (blargman) 2009-04-08 11:21:21 I forgot to mention that calls between inside SIP users are fine. Two way audio every time... By: Joshua C. Colp (jcolp) 2009-04-14 14:36:16 I'm going to be looking at this issue and trying to see what is going on. In the mean time it would be helpful if you could upgrade incrementally from rc1 up to rc4 so that we could try to narrow down where this regression was introduced. By: Jeff Phelps (blargman) 2009-04-14 15:26:19 confirmed that the regression starts with 1.6.1.0-rc3. It don't appear to be able to reproduce the issue in rc2 By: Joshua C. Colp (jcolp) 2009-04-14 15:38:26 Per discussion on IRC this was narrowed down to rc3, and it seems to be that the gateway does not accept the reinvite. By: Joshua C. Colp (jcolp) 2009-04-15 11:04:41 I have attached a patch which fixes a bug that your debug messages showed. I do not know if this will fix the issue but would like you to try it anyway. It can be applied by placing it in your Asterisk source directory, typing patch -p0 < 14850.diff, and recompiling/installing. Additionally after re-reading your description I see you say that outgoing calls to the gateway are working fine. Would you be able to attach a sip debug of one of these as well? By: Joshua C. Colp (jcolp) 2009-04-16 15:35:48 After a ton of discussion with the great reporter this was isolated to revision 182527 and seems to have actually exposed an issue with the version of CCM in use. It seems to require audio to flow before a reinvite can be issued. Due to the changes present in 182527 this no longer happens causing the bug to show itself. The reporter has been given a one line change for their environment. | ||