[asterisk-users] route INVITE sip:[EMAIL PROTECTED]
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: . Content-Type: application/sdp. CSeq: 34878212 INVITE. From: "0614740696" ;tag=02975-US-0223ae6e-67d6c4495. Max-Forwards: 31. To: . 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" ;tag=as25391ca7. To: . Contact: . 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
Re: [asterisk-users] route INVITE sip:[EMAIL PROTECTED]
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: . > Content-Type: application/sdp. > CSeq: 34878212 INVITE. > From: "0614740696" > ;tag=02975-US-0223ae6e-67d6c4495. > Max-Forwards: 31. > To: . > 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" ;tag=as25391ca7. > To: . > Contact: . > 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 > -- Esta mensagem (incluindo quaisquer anexos) pode conter informação confidencial para uso exclusivo do destinatário. Se não for o destinatário pretendido, não deverá usar, distribuir ou copiar este e-mail. Se recebeu esta mensagem por engano, por favor informe o emissor e elimine-a imediatamente. Obrigado. This e-mail message is intended only for individual(s) to whom it is addressed and may contain information that is privileged, confidential, proprietary, or otherwise exempt from disclosure under applicable law. If you believe you have received this message in error, please advise the sender by return e-mail and delete it from your mailbox. Thank you. ___ --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
Re: [asterisk-users] route INVITE sip:[EMAIL PROTECTED]
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: . >> Content-Type: application/sdp. >> CSeq: 34878212 INVITE. >> From: "0614740696" >> ;tag=02975-US-0223ae6e-67d6c4495. >> Max-Forwards: 31. >> To: . >> 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" ;tag=as25391ca7. >> To: . >> Contact: . >> 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
Re: [asterisk-users] route INVITE sip:[EMAIL PROTECTED]
Hi, I would suggest you to use Asterisk Application SIPGetHeader in your Dialplan for incoming calls from "plugandtel". http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+SIPGetHeader Something like * exten=>_[a-z].,1,SIPGetHeader(Var_TO=To) exten=>_[a-z].,2,Dial(SIP/${Var_TO}) Please be aware that this application as a different name in asterisk 1.4. Learn from CLI > show application ? Hope it helps, Best regards, MoutaPT * On Nov 14, 2007 3:02 PM, Marc LEURENT <[EMAIL PROTECTED]> wrote: > 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: . > >> Content-Type: application/sdp. > >> CSeq: 34878212 INVITE. > >> From: "0614740696" > >> ;user=phone>;tag=02975-US-0223ae6e-67d6c4495. > >> Max-Forwards: 31. > >> To: . > >> 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" ;tag=as25391ca7. > >> To: . > >> Contact: . > >> 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 > -- Esta mensagem (incluindo quaisquer anexos) pode conter informação confidencial para uso exclusivo do destinatário. Se não for o destinatário pretendido, não deverá usar, distribuir ou copiar este e-mail. Se recebeu esta mensagem por engano, por favor informe o emissor e elimine-a imediatamente. Obrigado. This e-mail message is intended only for individual(s) to whom it is addressed and may contain information that is privileged, confidential, proprietary, or otherwise exempt from disclosure under applicable law. If you believe you have received this message in error, please advise the sender by return e-mail and delete it from your mailbox. Thank you. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visi
Re: [asterisk-users] route INVITE sip:[EMAIL PROTECTED]
This is propably because you've registered in sip.conf without an extension in the register= configuration line. Check the syntax. If you can't fix that, use the SIP_HEADER() dialplan funcion to get the value of the To: header and use that as an extension. /Olle --- * Olle E Johansson - [EMAIL PROTECTED] * Register now for the Asterisk SIP Masterclass * Stockholm, Sweden, January 2008 ___ --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