Hi, I'm trying to integrate my asterisk PBX to an Avaya PBX. I am using chan_ooh323 from asterisk-addons. I am able to make a call from SIP Phone -> Asterisk -> Avaya -> Station (phone) and vice versa. I am also able to make a call from SIP Phone -> Asterisk -> Avaya -> PSTN. However I face problems when I make DID calls from the PSTN. The DID calls are made through analog DID lines to the TN753 on the Avaya. When I make the call, I can hear ringing on the caller phone (PSTN) and the SIP Phone rings. But when I pick up the SIP Phone, the caller phone remains in ringing mode. On the SIP Phone, I hear random sound. I did a packet capture and on the Q.931 setup information header, under Progress Indicator, the call is not end-to-end ISDN. So it seems that the SIP answer message is not being communicated properly to the Avaya PBX. Can this be the cause of the problem? Has anyone encountered this problem and what is your solution? Thanks in advance. Regards, Steve
try a answer() before the dial(sip/xxx) and if you are using originate try local/.... and start whit and answer() 2009/1/22 Steven J. Douglas <stevend at moij.biz>> Hi, > > I'm trying to integrate my asterisk PBX to an Avaya PBX. I am using > chan_ooh323 from asterisk-addons. > > I am able to make a call from SIP Phone -> Asterisk -> Avaya -> Station > (phone) and vice versa. > I am also able to make a call from SIP Phone -> Asterisk -> Avaya -> PSTN. > > However I face problems when I make DID calls from the PSTN. The DID > calls are made through analog DID lines to the TN753 on the Avaya. When > I make the call, I can hear ringing on the caller phone (PSTN) and the > SIP Phone rings. But when I pick up the SIP Phone, the caller phone > remains in ringing mode. On the SIP Phone, I hear random sound. > > I did a packet capture and on the Q.931 setup information header, under > Progress Indicator, the call is not end-to-end ISDN. So it seems that > the SIP answer message is not being communicated properly to the Avaya > PBX. Can this be the cause of the problem? > > Has anyone encountered this problem and what is your solution? > > Thanks in advance. > > Regards, > Steve > > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your (")_(")signature to help him gain world domination. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090122/68a458f1/attachment.htm