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Hello, <br>
<br>
My provider did not send correct DID number in the INVITE packet but
i can use "To" argument<br>
<br>
<tt>INVITE
<a class="moz-txt-link-abbreviated" href="mailto:sip:gw+0a96c3d3-0b0e-4864-b9ec-759fa4422429@92.222.18.xxx:5080;transport=udp;gw=0a96c3d3-0b0e-4864-b9ec-759fa4422429">sip:gw+0a96c3d3-0b0e-4864-b9ec-759fa4422429@92.222.18.xxx:5080;transport=udp;gw=0a96c3d3-0b0e-4864-b9ec-759fa4422429</a>
SIP/2.0.<br>
Call-ID: <a class="moz-txt-link-abbreviated" href="mailto:25016-VB-188fd96b-526e3dbd4@sip.ovh.fr">25016-VB-188fd96b-526e3dbd4@sip.ovh.fr</a>.<br>
Contact: <sip:10.7.1.60:5060>.<br>
Content-Type: application/sdp.<br>
CSeq: 403749831 INVITE.<br>
From: "0967212xxx"
<a class="moz-txt-link-rfc2396E" href="mailto:sip:0967212xxx@sip.ovh.fr;user=phone"><sip:0967212xxx@sip.ovh.fr;user=phone></a>;tag=25016-VE-188fd96c-18bb43586.<br>
Max-Forwards: 27.<br>
Record-Route: <sip:91.121.129.20:5060;lr>.<br>
<b>To: <a class="moz-txt-link-rfc2396E" href="mailto:sip:0557590xxx@10.7.1.60;user=phone"><sip:0557590xxx@10.7.1.60;user=phone></a>.</b><br>
Via: SIP/2.0/UDP
91.121.129.20:5060;branch=z9hG4bK-WGZO-1fe73949-2df58378.</tt><br>
<br>
Using asterisk i can bypass the issue using something like <tt>exten
=> s,1,Goto(from-trunk,${CUT(CUT(SIP_HEADER(To),@,1),:,2)},1)</tt>
but i am unable to do the same under freeswitch.<br>
<br>
My trunk configuration seems correct, as you can see i used
auto_to_user, but the destination number remains <tt>0033972480xxx
when i call 0557590xxx.</tt><br>
<br>
2015-05-01 18:02:12.235827 [INFO] mod_dialplan_xml.c:635 Processing
0967212xxx <0967212xxx>->0033972480xxx in context public<br>
<br>
<br>
<tt><include><br>
<gateway name="0a96c3d3-0b0e-4864-b9ec-759fa4422429"><br>
<param name="username" value="0033972480xxx"/><br>
<param name="password" value="xxxxxxx"/><br>
<param name="proxy" value="sip.ovh.fr"/><br>
<param name="expire-seconds" value="800"/><br>
<param name="register" value="true"/><br>
<param name="retry-seconds" value="30"/><br>
<param name="extension" value="auto_to_user"/><br>
<param name="context" value="public"/><br>
</gateway><br>
</include></tt><br>
<br>
I tried to edit my inbound dialplan manually, it works using
<condition field="${sip_to_user}" expression="0557590xxx" >
but i prefer a proper way to do this because i will also use telcos
with normal invite packets<br>
<br>
I how i can copy $sip_to_header to destination for this specific
trunk ?<br>
<br>
Please note that i use fusionpbx.<br>
<br>
Best regards, sorry for my bad English<br>
<br>
<br>
<br>
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