Set 'canreinvite=no' on all applicable peers? Cyprus VoIP wrote:
> Hello, > > We are trying to send faxes by T.38 protocol to a remote SIP proxy from > a local extension. The local extension sends the INVITE, Asterisk sends > the call to the Proxy the call is connected with a regular audio codec. > After a few seconds the remote proxy sends an INVITE with UDPTL and the > Asterisk sends it to the local extension and it's accepted, but (here > the problem starts) just after sending the OK with the proper SDP to the > remote Proxy, the Asterisk initiates a new INVITE to the local extension > and remote Proxy, with the normal audio codecs again. > > We set "t38pt_udptl=yes" in sip.conf and allowed all the codecs to the > local extension and remote Proxy, but it still forces the call to go > back to a voice call. > > Any idea why it happens and how to debug it? We set verbose and debug to > 20, but no "internal" info is provided to get a clear understanding on > Asterisk's "thoughts" during that process. > > Thank you in advance for your assistance, > > Andreas > > _______________________________________________ > -- 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 -- Alex Balashov - Principal Evariste Systems Web : http://www.evaristesys.com/ Tel : (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 _______________________________________________ -- 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
