| Summary: | ASTERISK-07965: SIP call termination fails due to From/To field mixup when callee hangs up | ||
| Reporter: | Nicolas-Peter Pohland (npohland) | Labels: | |
| Date Opened: | 2006-10-19 16:14:51 | Date Closed: | 2006-10-25 11:45:11 |
| Priority: | Major | Regression? | No |
| Status: | Closed/Complete | Components: | Channels/chan_sip/Interoperability |
| Versions: | Frequency of Occurrence | ||
| Related Issues: | |||
| Environment: | Attachments: | ||
| Description: | Asterisk 1.4 beta 3 is setup in a LAN with several SIP clients (snom 200, Covote TER 50 and Siemens C450 IP). Establishing a basic call works fine but terminating the call does work if the callee (called party) hangsup. This is due to the fact that the To and From fields are exchanged by Asterisk and the other party does not recognized the dialog it pertains to (tags are reversed). This causes the other party to respond with a 481 Call Leg/Transaction does exist or ignore it completely and continue with the call. If the caller hangs up everything is correct. | ||
| Comments: | By: Nicolas-Peter Pohland (npohland) 2006-10-20 01:01:13 - Latest SVN checkout also has this problem. By: Serge Vecher (serge-v) 2006-10-25 10:31:37 As per bug guidelines, you need to attach a SIP debug trace illustrating the problem. Please do the following: 1) Prepare test environment (reduce the amount of unrelated traffic on the server); 2) Make sure your logger.conf has the following line: console => notice,warning,error,debug 3) restart Asterik. 4) Enable SIP transaction logging with the following CLI commands: set debug 4 set verbose 4 sip debug 5) Save complete console log to file and _attach_ said file to the bug. i.e. asterisk -Tvvvvvdddddgc | tee /tmp/sipdebug.txt By: Olle Johansson (oej) 2006-10-25 11:27:12 We are aware of this problem and working to fix it. This is a duplicate bug report, btw. By: Olle Johansson (oej) 2006-10-25 11:45:10 Duplicate of 8130 | ||