<div><div dir="auto">If you need to set a sip-from to be the same always, you can set that in the gateway configuration:</div><div dir="auto"><br></div><div dir="auto"><pre class="syntaxhighlighter-pre" style="margin-top:0px;margin-bottom:0px;padding:0px;border:0px;outline:0px;font-size:15px;line-height:1.3;overflow-x:auto;word-break:normal;color:rgb(41,41,41)"><param name="from-user" value="cluecon"/></pre></div><div dir="auto"><br></div><div dir="auto"><br></div><div dir="auto">On Tue, 29 Oct 2019 at 15:56, Tamer Higazi via FreeSWITCH-users <<a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a>> wrote:</div></div><div><div class="gmail_quote"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><br><br><br>---------- Forwarded message ----------<br>From: Tamer Higazi <<a href="mailto:th982a@googlemail.com" target="_blank">th982a@googlemail.com</a>><br>To: <a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a><br>Cc: <br>Bcc: <br>Date: Sun, 27 Oct 2019 23:36:54 +0100<br>Subject: replace SIP from user and host part<br>Hi people,<br>
<br>
My ISP provider is very special and forces me to reset the user and host <br>
part of my external profile.<br>
<br>
At the current stage, FS sets the endpoint user ID as sip-from user, and <br>
the external ip-address as the sip-from host and the ISP gateway returns <br>
me a 404 not found.<br>
<br>
When I turn off the PBX and register to the gateway directly with a <br>
softphone, the softphone can place a call.<br>
sip-from user is the userID of the gateway login and host the proxy <br>
address of the gateway.<br>
<br>
How can I set a new sip-from-user and host prior bridging a call to the <br>
gateway?<br>
<br>
I found 2 channel variables.<br>
<br>
variable_sip_from_user - sip_from_user<br>
variable_sip_from_host - sip_from_host<br>
<br>
Is that the solution manipulating these variables?<br>
Am I capable manipulate those ?<br>
Would that be the solution ?<br>
<br>
I am a little bit new to FS.<br>
For a dialplan example I welcome thankfully any advise.<br>
<br>
<br>
wireshark dump:<br>
<br>
JITSI Softphone Call:<br>
Session Initiation Protocol (100)<br>
Status-Line: SIP/2.0 100 trying -- your call is important to us<br>
Status-Code: 100<br>
[Resent Packet: False]<br>
[Request Frame: 2481]<br>
[Response Time (ms): 13]<br>
Message Header<br>
Call-ID: bb85334b160ce6cb26ab480fd2a13236@0:0:0:0:0:0:0:0<br>
[Generated Call-ID: <br>
bb85334b160ce6cb26ab480fd2a13236@0:0:0:0:0:0:0:0]<br>
CSeq: 2 INVITE<br>
Sequence Number: 2<br>
Method: INVITE<br>
From: "117xxxxxx" <<a href="mailto:sip%3A117xxxxxx@sip12.e-fon.ch" target="_blank">sip:117xxxxxx@sip12.e-fon.ch</a>>;tag=51a3e1d6<br>
SIP Display info: "117xxxxxx"<br>
SIP from address: <a href="mailto:sip%3A117xxxxxx@sip12.e-fon.ch" target="_blank">sip:117xxxxxx@sip12.e-fon.ch</a><br>
SIP from address User Part: 117xxxxxx<br>
SIP from address Host Part: <a href="http://sip12.e-fon.ch" rel="noreferrer" target="_blank">sip12.e-fon.ch</a><br>
SIP from tag: 51a3e1d6<br>
To: <<a href="mailto:sip%3A0764823425@sip12.e-fon.ch" target="_blank">sip:0764823425@sip12.e-fon.ch</a>><br>
SIP to address: <a href="mailto:sip%3A076xxxxxxx@sip12.e-fon.ch" target="_blank">sip:076xxxxxxx@sip12.e-fon.ch</a><br>
SIP to address User Part: 076xxxxxxx<br>
SIP to address Host Part: <a href="http://sip12.e-fon.ch" rel="noreferrer" target="_blank">sip12.e-fon.ch</a><br>
Via: SIP/2.0/UDP <br>
217.162.xxx.xxx:5060;branch=z9hG4bK-383638-321b4bfd0c24f1f48906ff6382c3b18b;rport=5060<br>
Transport: UDP<br>
Sent-by Address: 217.162.xxx.xxx<br>
Sent-by port: 5060<br>
Branch: z9hG4bK-383638-321b4bfd0c24f1f48906ff6382c3b18b<br>
RPort: 5060<br>
Content-Length: 0<br>
<br>
<br>
(successfull placing the call)<br>
<br>
<br>
FS Call:<br>
Session Initiation Protocol (404)<br>
Status-Line: SIP/2.0 404 Not found<br>
Status-Code: 404<br>
[Resent Packet: False]<br>
[Request Frame: 12659]<br>
[Response Time (ms): 13]<br>
Message Header<br>
Via: SIP/2.0/UDP <br>
217.162.xxx.xxx:5080;rport=5080;branch=z9hG4bKmNHFUDQav3tUj;received=217.162.xxx.xxx<br>
Transport: UDP<br>
Sent-by Address: 217.162.xxx.xxx<br>
Sent-by port: 5080<br>
RPort: 5080<br>
Branch: z9hG4bKmNHFUDQav3tUj<br>
Received: 217.162.xxx.xxx<br>
From: "1000" <sip:1000@217.162.xxx.xxx>;tag=gNe9eHQm44HBj<br>
SIP Display info: "1000"<br>
SIP from address: sip:1000@217.162.xxx.xxx<br>
SIP from address User Part: 1000<br>
SIP from address Host Part: 217.162.xxx.xxx<br>
SIP from tag: gNe9eHQm44HBj<br>
To: <br>
<<a href="mailto:sip%3A076xxxxxxx@sip12.e-fon.ch" target="_blank">sip:076xxxxxxx@sip12.e-fon.ch</a>>;tag=56dedfd3319cd1b9fedc1197b499329c.26ea<br>
SIP to address: <a href="mailto:sip%3A076xxxxxxx@sip12.e-fon.ch" target="_blank">sip:076xxxxxxx@sip12.e-fon.ch</a><br>
SIP to address User Part: 076xxxxxxx<br>
SIP to address Host Part: <a href="http://sip12.e-fon.ch" rel="noreferrer" target="_blank">sip12.e-fon.ch</a><br>
SIP to tag: 56dedfd3319cd1b9fedc1197b499329c.26ea<br>
Call-ID: 84f02fe4-71ef-1238-b881-00012e808c2d<br>
[Generated Call-ID: 84f02fe4-71ef-1238-b881-00012e808c2d]<br>
CSeq: 11458424 INVITE<br>
Sequence Number: 11458424<br>
Method: INVITE<br>
Content-Length: 0<br>
<br>
<br>
<br>
<br><br><br>---------- Forwarded message ----------<br>From: Tamer Higazi via FreeSWITCH-users <<a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a>><br>To: <a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a><br>Cc: <br>Bcc: <br>Date: Tue, 29 Oct 2019 08:56:21 -0700 (PDT)<br>Subject: [Freeswitch-users] replace SIP from user and host part<br>_________________________________________________________________________<br>
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<a href="https://freeswitch.com" rel="noreferrer" target="_blank">https://freeswitch.com</a></blockquote></div></div>-- <br><div dir="ltr" class="gmail_signature" data-smartmail="gmail_signature"><div dir="ltr"><div>Regards,</div><div><br></div>David Villasmil<div>email: <a href="mailto:david.villasmil.work@gmail.com" target="_blank">david.villasmil.work@gmail.com</a></div><div>phone: +34669448337</div></div></div>