I would like to migration from chan_sip to pjsip connecting an asterisk box to Avaya CM, but it is found that the SIP response are behaving differently between chan_sip and pjsip, is there any options I can use the same behavior as chan_sip? Thanks

    The Avaya is expecting the response to send to port 5060 instead of the port used for sending the request. Is this possible with pjsip?

Hi, I would like to migration from chan_sip to pjsip connecting an asterisk box to Avaya CM, but it is found that the SIP response are behaving differently between chan_sip and pjsip, is there any options I can use the same behavior as chan_sip? Thanks OPTIONS request from Avaya: <--- SIP read from TCP:x.x.31.40:29778 ---> OPTIONS sip:asterisk SIP/2.0 From: <sip:x.x.31.40>;tag=809ea16429d9e8179a05bd6969100 To: <sip:asterisk> Call-ID: 809ea16429d9e8179a05bd6969100 CSeq: 21003 OPTIONS Max-Forwards: 70 Via: SIP/2.0/TCP x.x.31.40;branch=z9hG4bK809ea16429d9e817aa05bd6969100 User-Agent: Avaya CM/R016x.03.0.124.0 Contact: <sip:invalid@x.x.31.40;transport=tcp> Route: <sip:x.x.34.155;transport=tcp;lr> Expires: 0 Content-Length: 0 SIP response using chan_sip: <--- Transmitting (no NAT) to x.x.31.40:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/TCP x.x.31.40;branch=z9hG4bK809ea16429d9e817aa05bd6969100;received=x.x.31.40 From: <sip:x.x.31.40>;tag=809ea16429d9e8179a05bd6969100 To: <sip:asterisk >;tag=as689d17f2 Call-ID: 809ea16429d9e8179a05bd6969100 CSeq: 21003 OPTIONS Server: Asterisk PBX 16.0.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: <sip:x.x.34.155:5060;transport=tcp> Accept: application/sdp Content-Length: 0 OPTIONS request from Avaya: <--- Received SIP request (439 bytes) from TCP:x.x.31.40:26243 ---> OPTIONS sip:asterisk SIP/2.0 From: <sip:x.x.31.40>;tag=0fed5a02bd9e8158a65bd6969100 To: <sip:asterisk > Call-ID: 0fed5a02bd9e8158a65bd6969100 CSeq: 27168 OPTIONS Max-Forwards: 70 Via: SIP/2.0/TCP x.x.31.40;branch=z9hG4bK0fed5a02bd9e8159a65bd6969100 User-Agent: Avaya CM/R016x.03.0.124.0 Contact: <sip:invalid@x.x.31.40;transport=tcp> Route: <sip:x.x.34.155;transport=tcp;lr> Expires: 0 Content-Length: 0 SIP response using pjsip: <--- Transmitting SIP response (522 bytes) to TCP:x.x.31.40:26243 ---> SIP/2.0 200 OK Via: SIP/2.0/TCP x.x.31.40;received=x.x.31.40;branch=z9hG4bK0fed5a02bd9e8159a65bd6969100 Call-ID: 0fed5a02bd9e8158a65bd6969100 From: <sip:x.x.31.40>;tag=0fed5a02bd9e8158a65bd6969100 To: <sip:asterisk >;tag=z9hG4bK0fed5a02bd9e8159a65bd6969100 CSeq: 27168 OPTIONS Accept: application/sdp Allow: OPTIONS, REGISTER, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE Supported: 100rel, timer Accept-Encoding: text/plain Accept-Language: en Server: Asterisk PBX 16.0.0 Content-Length: 0 The Avaya is expecting the response to send to port 5060 instead of the port used for sending the request. Is this possible with pjsip? Terence Lam