[asterisk-dev] Channel problems

[voip at gmx-topmail.de]

> you may want to post this to mantis at bugs.digium.com so if it is a bug 
> it can be documented , tagged, and looked into


Thanks for this information. But it was not a good idea.

"1. First of all, you are using an AGI script to dial. If you dial directly with Dial(), I bet this problem won't happen.
2. if the device looses network connectivity, there is a setting in sip.conf to specifically address this - 'rtptimeout'

You have posted a bug report marked 'major' that has to do with a custom component, rather than Asterisk proper, which could also be addressed with a configuration option. This is the second time you do this. In the future, if you think you have identified an Asterisk bug, please consult somebody else either on #asterisk channel on IRC or asterisk-users mailing list. Thank you."

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.



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