Rich Adamson wrote:
I'm having a weird problem. The setup is Asterisk A with a
TDM400P/4xFXO, DSL line, and Asterisk B with TDM400P/4xFXO on another
DSL line.
Both boxes are behind their own NAT. Asterisk B forwards calls from it's
four PSTN ports to Asterisk A over an IAX2 trunk, which works fine using
the GSM codec. Asterisk A dials the SIP phones on it's local segment.
The problem is that when the inbound PSTN call ends, the hangups are
detected, but for some reason, Asterisk B starts a new call all over
again, Asteriks A receives it, the SIP phones ring, but when one of them
picks up there is a dialtone, busy tone, or silence.
Is there anything I may be missing here? I can post .conf files, but I
don't think it has anything to do with those. Calls on the local PSTN
ports of Asterisk A work fine. This setup is in Spain, FYI.
Kind of sounds like an issue with detecting pstn line supervision events,
but almost impossible to guess at root cause unless you provide something
to look at.
Might try some of the cli debug commands; 'zap debug', 'iax2 debug', etc.
Look those over very closely and you're likely to spot the problem.
If not, post the results. Include * version data as well.
Hi Richard,
Thanks for the pointers, I will try those debugs and will post the results.
Best regards,
Mike
_______________________________________________
Asterisk-Users mailing list
[email protected]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users