|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|
|Description:||rtptimeout option for peers works for outbound calls,|
but doesn't work for inbound calls,
Global rtptimeout option works for both directions.
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!