We are using 2 different incoming trunks. The first one is alsion.com and is sending INVITE with phone number in the INVITE line whereas plugandtel put the callee number only inside the To: Section.
Marco Mouta a écrit : > Could you describe in detail how did you fall into this situation, I mean > the real example which SIP phone sends this invite? Is registered in > asterisk? it is a non-registered sip phone trying to dial a sip user at your > * box? > > If this is an issue with a specific hardware outside of your asterisk, may > be something not well configured ... describe it a bit more in detail. > > If you don't have anyworkaround for this Invite format I would use OpenSER > in front of Asterisk to handle this invites and replace to SIP URI with info > from the tag TO: ... > > Any way if you provide more details may be someone in the Mailing list is > able to help u out;) > > Best regards > MoutaPT > > On Nov 13, 2007 6:14 PM, Marc LEURENT <[EMAIL PROTECTED]> wrote: > >> Good evening! >> I was wondering one thing, >> I'm using freepbx to configure my asterisk server and I have a problem >> with some inbound calls. >> >> When I receive a call to an INVITE sip:[EMAIL PROTECTED] I an set an >> inbound route! It matches a DID number. >> >> How can I route an INVITE sip:[EMAIL PROTECTED] The number only appear in the >> To: Section. >> >> Thanks! >> >> Example: >> >> With this one, I cannot route it (there is only the number to be reached >> in the To: section) >> # >> U 217.36.112.145:5060 -> 192.168.95.235:5060 >> INVITE sip:[EMAIL PROTECTED]:5060;transport=udp SIP/2.0. >> Allow: UPDATE,REFER,INFO. >> Call-ID: [EMAIL PROTECTED] >> Contact: <sip:217.66.118.145:5060>. >> Content-Type: application/sdp. >> CSeq: 34878212 INVITE. >> From: "0614740696" >> <sip:[EMAIL PROTECTED];user=phone>;tag=02975-US-0223ae6e-67d6c4495. >> Max-Forwards: 31. >> To: <sip:[EMAIL PROTECTED];user=phone>. >> User-Agent: Cirpack/v4.41c (gw_sip). >> Via: SIP/2.0/UDP 217.36.112.145:5060;branch=z9hG4bK-744D-33B812. >> Content-Length: 303. >> . >> >> >> >> Whereas with this one I can do it! (there is a number in the INVITE) >> # >> U 87.98.202.114:5060 -> 192.168.95.235:5060 >> INVITE sip:[EMAIL PROTECTED] SIP/2.0. >> Via: SIP/2.0/UDP 87.98.202.114:5060;branch=z9hG4bK1fd2c6b4;rport. >> From: "0158136741" <sip:[EMAIL PROTECTED]>;tag=as25391ca7. >> To: <sip:[EMAIL PROTECTED]>. >> Contact: <sip:[EMAIL PROTECTED]>. >> Call-ID: [EMAIL PROTECTED] >> CSeq: 102 INVITE. >> User-Agent: Asterisk PBX. >> Max-Forwards: 70. >> Date: Tue, 13 Nov 2007 18:07:00 GMT. >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. >> Content-Type: application/sdp. >> Content-Length: 233. >> . >> >> _______________________________________________ >> --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 >> > > > > > ------------------------------------------------------------------------ > > _______________________________________________ > --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 _______________________________________________ --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
