| Summary: | ASTERISK-12657: SIP channel never gets destroyed - Tx: BYE - response 404 | ||
| Reporter: | Stefan Tichy (st) | Labels: | |
| Date Opened: | 2008-08-28 12:28:22 | Date Closed: | 2008-08-29 11:20:01 |
| Priority: | Minor | Regression? | No |
| Status: | Closed/Complete | Components: | Channels/chan_sip/Interoperability |
| Versions: | Frequency of Occurrence | ||
| Related Issues: | |||
| Environment: | Attachments: | ||
| Description: | Two calls are made from one SIP Phone and SIP REFER is used to connect the callees. Works fine so far. The problem is, that asterisk fails to destroy one of the sip channels. The history shows theese 4 lines again and again: 37. AutoDestroy 38070dc46ab0085d 38. TxReqRel BYE / 104 BYE - -UNKNOWN- 39. SchedDestroy 32000 ms 40. Rx SIP/2.0 / 104 BYE / 404 Not Found ****** ADDITIONAL INFORMATION ****** The phone is a OpenStage20 (V1 R4). The 404 response might be the result of a bug in the firmware of the phone, but asterisk should destroy the channel after some time. If the phone is unplugged the channel will disappear. | ||
| Comments: | By: Frederic LE FOLL (flefoll) 2008-08-29 03:18:38 Did the phone already clear the call after the REFER succeeded ? I had a (possibly) similar problem with blind transfers ; it is described in issue 12865 <http://bugs.digium.com/view.php?id=12865> and I proposed a patch that is in trunk but not yet in 1.4(.21.2). Maybe you can give it a try. By: Stefan Tichy (st) 2008-08-29 10:39:23 Tanks a lot, now all sip channels come to an end. By: Mark Michelson (mmichelson) 2008-08-29 11:19:58 If the patch on ASTERISK-12201 fixes the issue, then I will close this as well. Please feel free to re-open if the other patch did not completely fix the issue. | ||