| Summary: | ASTERISK-15168: Outoing calls disconnected immediately after remote end picks up. | ||
| Reporter: | Brendan Martens (shrift) | Labels: | |
| Date Opened: | 2009-11-18 15:54:38.000-0600 | Date Closed: | 2009-11-19 09:49:52.000-0600 |
| Priority: | Blocker | Regression? | No |
| Status: | Closed/Complete | Components: | General |
| Versions: | Frequency of Occurrence | ||
| Related Issues: | |||
| Environment: | Attachments: | ( 0) asterisk-1.6.2.0-rc6_console-debug-verbose.txt ( 1) asterisk-1.6.2.0-rc6_sip-debug.txt | |
| Description: | When I initiate a call from an asterisk peer which uses my sip trunk provider the call is immediately disconnected as soon as the remote person picks up. Calls between extensions on my asterisk are fine. I don't know if this is *for sure* asterisk's problem, however going from -rc3 to -rc6 with no configuration changes causes this problem. I unfortunately did not try any candidates between those two, so I don't know where the regression occurred, but it was not present in -rc3 or earlier. ****** ADDITIONAL INFORMATION ****** There is no Product Version for 1.6.2.0 for me to select. I'm attaching a sip debug output as well as console with verbose 50 and debug 50. | ||
| Comments: | By: Brendan Martens (shrift) 2009-11-18 15:55:04.000-0600 I'm sorry I didn't know what category to make this, so I chose general. : ( By: Leif Madsen (lmadsen) 2009-11-19 09:49:51.000-0600 This appears to be a duplicate of 16238 so I'm closing this one out. The work around appears to be to use ignoresdpversion in sip.conf. | ||