| Summary: | ASTERISK-14841: rtptimeout option doesn't work for inbound calls | ||
| Reporter: | Makoto Dei (makoto) | Labels: | |
| Date Opened: | 2009-09-15 21:05:27 | Date Closed: | 2009-09-18 12:04:59 |
| Priority: | Minor | Regression? | No |
| Status: | Closed/Complete | Components: | Channels/chan_sip/General |
| Versions: | Frequency of Occurrence | ||
| Related Issues: | |||
| Environment: | Attachments: | ||
| Description: | rtptimeout option for peers works for outbound calls, but doesn't work for inbound calls, Global rtptimeout option works for both directions. Steps: 1. Set global rtptimeout option to 0 2. Set rtptimeout option for Phone A to 30 3. Call from Phone B to Phone A 4. Unplug network cable from Phone A 5. Wait 30 seconds 6. Asterisk disconnects the call 7. Call from Phone A to Phone B 8. Unplug network cable from Phone A 9. Wait 30 seconds 10. Asterisk should disconnect the call, but it doesn't | ||
| Comments: | By: Leif Madsen (lmadsen) 2009-09-16 08:45:00 Would you mind also providing the console output with debugging enabled while you perform this test? Thanks! By: David Woolley (davidw) 2009-09-17 10:38:02 This wouldn't have worked for 1.6.0.x or earlier, as noted in ASTERISK-12361. My understanding was that the merging of sip_user into sip_peer that happened for 1.6.1.x was supposed to have removed the problem, although I haven't tested this as it hasn't been a priority recently, and I only checked that the merging of the structures had occured. By: Leif Madsen (lmadsen) 2009-09-18 12:04:59 Based on what davidw said, I'm closing this issue out. I believe you can test this against Asterisk 1.6.1.x and the issue should resolve itself (if not, then possibly 1.6.2.x). If neither of those issues resolves, then please open a new issue. Thanks! | ||