[Asterisk-Users] Calls from IAX2 trunk start again when hung
Hi all, 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. Regards, Mike ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Calls from IAX2 trunk start again when hung
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. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Calls from IAX2 trunk start again when hung
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 Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
