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

[Marco Mouta]

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.
> .
>
> _______________________________________________
> --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.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071114/433a3ad7/attachment.htm