| Summary: | ASTERISK-12165: Attended transfer no audio | ||
| Reporter: | jaybeepee (jaybeepee) | Labels: | |
| Date Opened: | 2008-06-10 04:11:59 | Date Closed: | 2011-06-07 14:03:07 |
| Priority: | Major | Regression? | No |
| Status: | Closed/Complete | Components: | Channels/chan_sip/General |
| Versions: | Frequency of Occurrence | ||
| Related Issues: | |||
| Environment: | Attachments: | ||
| Description: | When doing an attended transfer with Asterisk (Asterisk 1.6.0-beta9) the audio is lost. It is only regained by putting the user on hold and then unholding again. | ||
| Comments: | By: Joshua C. Colp (jcolp) 2008-06-10 07:35:41 This issue report is not complete at all. Please provide complete console output along with how the transfer is being performed, and if SIP is involved an rtp debug to go with it. By: Tilghman Lesher (tilghman) 2008-06-20 18:21:14 Suspended, due to lack of response. If you can provide the requested information, please reopen this issue or request assistance from a bug marshal on #asterisk-bugs on Freenode, if necessary. | ||