Hi,<br><br>I would suggest you to use Asterisk Application SIPGetHeader in your Dialplan for incoming calls from "plugandtel".<br><br><a href="http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+SIPGetHeader">
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+SIPGetHeader</a><br><br>Something like<br><b><br>exten=>_[a-z].,1,SIPGetHeader(Var_TO=To)<br>exten=>_[a-z].,2,Dial(SIP/${Var_TO})<br><br>Please be aware that this application as a different name in asterisk
1.4.<br><br>Learn from CLI > show application ?<br><br>Hope it helps,<br><br>Best regards,<br>MoutaPT<br><br><br></b><br><div class="gmail_quote">On Nov 14, 2007 3:02 PM, Marc LEURENT <<a href="mailto:lftsy@free.fr">
lftsy@free.fr</a>> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">We are using 2 different incoming trunks.<br>The first one is
<a href="http://alsion.com" target="_blank">alsion.com</a> and is sending INVITE with phone number in<br>the INVITE line whereas plugandtel put the callee number only inside the<br>To: Section.<br><br><br><br>Marco Mouta a écrit :
<br><div><div></div><div class="Wj3C7c">> Could you describe in detail how did you fall into this situation, I mean<br>> the real example which SIP phone sends this invite? Is registered in<br>> asterisk? it is a non-registered sip phone trying to dial a sip user at your
<br>> * box?<br>><br>> If this is an issue with a specific hardware outside of your asterisk, may<br>> be something not well configured ... describe it a bit more in detail.<br>><br>> If you don't have anyworkaround for this Invite format I would use OpenSER
<br>> in front of Asterisk to handle this invites and replace to SIP URI with info<br>> from the tag TO: ...<br>><br>> Any way if you provide more details may be someone in the Mailing list is<br>> able to help u out;)
<br>><br>> Best regards<br>> MoutaPT<br>><br>> On Nov 13, 2007 6:14 PM, Marc LEURENT <<a href="mailto:lftsy@free.fr">lftsy@free.fr</a>> wrote:<br>><br>>> Good evening!<br>>> I was wondering one thing,
<br>>> I'm using freepbx to configure my asterisk server and I have a problem<br>>> with some inbound calls.<br>>><br>>> When I receive a call to an INVITE <a href="mailto:sip:01xxxxxx@myip.com">
sip:01xxxxxx@myip.com</a> I an set an<br>>> inbound route! It matches a DID number.<br>>><br>>> How can I route an INVITE <a href="mailto:sip:s@myip.com">sip:s@myip.com</a>? The number only appear in the
<br>>> To: Section.<br>>><br>>> Thanks!<br>>><br>>> Example:<br>>><br>>> With this one, I cannot route it (there is only the number to be reached<br>>> in the To: section)<br>
>> #<br>>> U <a href="http://217.36.112.145:5060" target="_blank">217.36.112.145:5060</a> -> <a href="http://192.168.95.235:5060" target="_blank">192.168.95.235:5060</a><br>>> INVITE sip:s@192.168.95.235
:5060;transport=udp SIP/2.0.<br>>> Allow: UPDATE,REFER,INFO.<br>>> Call-ID: <a href="mailto:02975-TP-0223ae6d-6daf01263@sip.lecom.com">02975-TP-0223ae6d-6daf01263@sip.lecom.com</a>.<br>>> Contact: <sip:
<a href="http://217.66.118.145:5060" target="_blank">217.66.118.145:5060</a>>.<br>>> Content-Type: application/sdp.<br>>> CSeq: 34878212 INVITE.<br>>> From: "0614740696"<br>>> <<a href="mailto:sip:0614740696@sip.lecom.com">
sip:0614740696@sip.lecom.com</a>;user=phone>;tag=02975-US-0223ae6e-67d6c4495.<br>>> Max-Forwards: 31.<br>>> To: <<a href="mailto:sip:0170080048@127.0.0.1">sip:0170080048@127.0.0.1</a>;user=phone>.<br>
>> User-Agent: Cirpack/v4.41c (gw_sip).<br>>> Via: SIP/2.0/UDP <a href="http://217.36.112.145:5060" target="_blank">217.36.112.145:5060</a>;branch=z9hG4bK-744D-33B812.<br>>> Content-Length: 303.<br>>> .
<br>>><br>>><br>>><br>>> Whereas with this one I can do it! (there is a number in the INVITE)<br>>> #<br>>> U <a href="http://87.98.202.114:5060" target="_blank">87.98.202.114:5060</a> ->
<a href="http://192.168.95.235:5060" target="_blank">192.168.95.235:5060</a><br>>> INVITE <a href="mailto:sip:0170704626@192.168.95.235">sip:0170704626@192.168.95.235</a> SIP/2.0.<br>>> Via: SIP/2.0/UDP <a href="http://87.98.202.114:5060" target="_blank">
87.98.202.114:5060</a>;branch=z9hG4bK1fd2c6b4;rport.<br>>> From: "0158136741" <<a href="mailto:sip:0158136740@87.98.201.114">sip:0158136740@87.98.201.114</a>>;tag=as25391ca7.<br>>> To: <<a href="mailto:sip:0170704626@192.168.95.235">
sip:0170704626@192.168.95.235</a>>.<br>>> Contact: <<a href="mailto:sip:0158136741@87.98.201.114">sip:0158136741@87.98.201.114</a>>.<br>>> Call-ID: <a href="mailto:4091f4686a9bbc4c5223fe9c6cf60a62@87.98.202.114">
4091f4686a9bbc4c5223fe9c6cf60a62@87.98.202.114</a>.<br>>> CSeq: 102 INVITE.<br>>> User-Agent: Asterisk PBX.<br>>> Max-Forwards: 70.<br>>> Date: Tue, 13 Nov 2007 18:07:00 GMT.<br>>> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
<br>>> Content-Type: application/sdp.<br>>> Content-Length: 233.<br>>> .<br>>><br>>> _______________________________________________<br>>> --Bandwidth and Colocation Provided by <a href="http://www.api-digital.com--" target="_blank">
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http://lists.digium.com/mailman/listinfo/asterisk-users</a><br>>><br>><br>><br>><br>><br></div></div>> ------------------------------------------------------------------------<br><div><div></div><div class="Wj3C7c">
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