| Summary: | ASTERISK-09398: asterisk destroy SIP dialog before auth BUY | ||
| Reporter: | Igor Goncharovsky (igorg) | Labels: | |
| Date Opened: | 2007-05-08 01:26:30 | Date Closed: | 2007-07-11 19:58:56 |
| Priority: | Minor | Regression? | No |
| Status: | Closed/Complete | Components: | Channels/chan_sip/Interoperability |
| Versions: | Frequency of Occurrence | ||
| Related Issues: | |||
| Environment: | Attachments: | ( 0) sip-bye.txt | |
| Description: | Situation is so: I have asterisk with registred Linksys and asterisk registred to operator. I make call to PSTN phone, PSTN user pick up receiver. After that I hang up, and PSTN user hear nothing, only silence, no hangup tone. Seems that asterisk sent BYE message to SIP server and do not authenticate it. PS: I have changed IPs and phones in debug to fake. | ||
| Comments: | By: Olle Johansson (oej) 2007-05-09 08:41:34 Ok, this is a bug that needs to be checked. By: Olle Johansson (oej) 2007-05-16 05:23:49 Are both of these Asterisk servers unmodified? Which versions? By: Olle Johansson (oej) 2007-05-16 05:38:04 Please test with latest 1.4 from svn, I've committed a change which I believe will change this. THanks. Rev 64602 By: Olle Johansson (oej) 2007-05-16 06:01:45 Ok, this is fixed in 1.2, 1.4 and trunk. Found another issue while working on this. If you still have issues, please re-open this report. THanks. | ||