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 <[email protected]> > 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.
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