[asterisk-users] route INVITE sip:s at sip.test.fr

[Marc LEURENT]

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 <lftsy at free.fr> 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:01xxxxxx at myip.com I an set an
>> inbound route! It matches a DID number.
>>
>> How can I route an INVITE sip:s at myip.com? 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:s at 192.168.95.235:5060;transport=udp SIP/2.0.
>> Allow: UPDATE,REFER,INFO.
>> Call-ID: 02975-TP-0223ae6d-6daf01263 at sip.lecom.com.
>> Contact: <sip:217.66.118.145:5060>.
>> Content-Type: application/sdp.
>> CSeq: 34878212 INVITE.
>> From: "0614740696"
>> <sip:0614740696 at sip.lecom.com;user=phone>;tag=02975-US-0223ae6e-67d6c4495.
>> Max-Forwards: 31.
>> To: <sip:0170080048 at 127.0.0.1;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:0170704626 at 192.168.95.235 SIP/2.0.
>> Via: SIP/2.0/UDP 87.98.202.114:5060;branch=z9hG4bK1fd2c6b4;rport.
>> From: "0158136741" <sip:0158136740 at 87.98.201.114>;tag=as25391ca7.
>> To: <sip:0170704626 at 192.168.95.235>.
>> Contact: <sip:0158136741 at 87.98.201.114>.
>> Call-ID: 4091f4686a9bbc4c5223fe9c6cf60a62 at 87.98.202.114.
>> 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.
>> .
>>
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> 
> 
> 
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