Summary:ASTERISK-14841: rtptimeout option doesn't work for inbound calls
Reporter:Makoto Dei (makoto)Labels:
Date Opened:2009-09-15 21:05:27Date Closed:2009-09-18 12:04:59
Versions:Frequency of
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!