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

Wed Nov 14 13:14:41 CST 2007

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 <lftsy at free.fr> 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 <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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> >>
> >
> >
> >
> >
> > ------------------------------------------------------------------------
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