[asterisk-dev] Channel problems

[voip at gmx-topmail.de]

Hi,

> ----- voip at gmx-topmail.de wrote:
> > But this isn't the solution. Because I set an AbsoluteTimeout that had
> > been ignored by asterisk... Also AGI is a part of Asterisk and if I
> > send a Dial-Command via AGI. Since it had been executed its up to
> > asterisk to finish the call. So this really is a stupid answer to a
> > major bug.
> AbsoluteTimeout will have no effect if there is no media (audio/video)
> flowing for the call.

rtptimeout also doesn't have an effect:

-----
Jul 22 01:36:02 DEBUG[11595] chan_sip.c: Outgoing Call for 700554333296999
Jul 22 01:36:02 VERBOSE[11595] logger.c:     -- Called 700554333296999 at 217.111.25.154
Jul 22 01:36:06 VERBOSE[11595] logger.c:     -- SIP/217.111.25.154-08201fe8 is making progress passing it to SIP/mysipid-08215dc8
Jul 22 01:36:07 DEBUG[11595] chan_sip.c: Oooh, format changed to 2
Jul 22 01:36:07 NOTICE[11595] rtp.c: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 217.111.25.154
Jul 22 01:36:21 VERBOSE[11595] logger.c:     -- SIP/217.111.25.154-08201fe8 answered SIP/mysipid-08215dc8
Jul 22 01:36:21 VERBOSE[11595] logger.c:     -- Attempting native bridge of SIP/mysipid-08215dc8 and SIP/217.111.25.154-08201fe8
Jul 22 02:36:02 DEBUG[11595] channel.c: Nobody there, continuing...
Jul 22 02:36:02 DEBUG[11595] channel.c: Nobody there, continuing...
Jul 22 02:36:02 DEBUG[11595] channel.c: Nobody there, continuing...
Jul 22 02:36:02 DEBUG[11595] channel.c: Nobody there, continuing...
Jul 22 02:36:02 DEBUG[11595] channel.c: Nobody there, continuing...
Jul 22 02:36:02 DEBUG[11595] channel.c: Nobody there, continuing...
Jul 22 02:36:02 DEBUG[11595] channel.c: Nobody there, continuing...
Jul 22 02:36:02 DEBUG[11595] channel.c: Nobody there, continuing...
Jul 22 02:36:02 DEBUG[11595] channel.c: Nobody there, continuing...
Jul 22 02:36:02 DEBUG[11595] channel.c: Nobody there, continuing...
Jul 22 02:36:02 DEBUG[11595] channel.c: Nobody there, continuing...

---

rtptimeout is set to 60, absolutetimeout to 3600. Asterisk version is now 1.2.10 ... so I still think this is a major bug... After this call noone was able to register at asterisk for 6 hours.

what to do?

Thanks
-- 


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