---------- Forwarded message ----------
From: Tamer Higazi <th982a@googlemail.com>
To: freeswitch-users@lists.freeswitch.org
Cc:
Bcc:
Date: Sun, 27 Oct 2019 23:36:54 +0100
Subject: replace SIP from user and host part
Hi people,

My ISP provider is very special and forces me to reset the user and host
part of my external profile.

At the current stage, FS sets the endpoint user ID as sip-from user, and
the external ip-address as the sip-from host and the ISP gateway returns
me a 404 not found.

When I turn off the PBX and register to the gateway directly with a
softphone, the softphone can place a call.
sip-from user is the userID of the gateway login and host the proxy
address of the gateway.

How can I set a new sip-from-user and host prior bridging a call to the
gateway?

I found 2 channel variables.

variable_sip_from_user - sip_from_user
variable_sip_from_host - sip_from_host

Is that the solution manipulating these variables?
Am I capable manipulate those ?
Would that be the solution ?

I am a little bit new to FS.
For a dialplan example I welcome thankfully any advise.

wireshark dump:

JITSI Softphone Call:
Session Initiation Protocol (100)
     Status-Line: SIP/2.0 100 trying -- your call is important to us
         Status-Code: 100
         [Resent Packet: False]
         [Request Frame: 2481]
         [Response Time (ms): 13]
     Message Header
         Call-ID: bb85334b160ce6cb26ab480fd2a13236@0:0:0:0:0:0:0:0
         [Generated Call-ID:
bb85334b160ce6cb26ab480fd2a13236@0:0:0:0:0:0:0:0]
         CSeq: 2 INVITE
             Sequence Number: 2
             Method: INVITE
         From: "117xxxxxx" <sip:117xxxxxx@sip12.e-fon.ch>;tag=51a3e1d6
             SIP Display info: "117xxxxxx"
             SIP from address: sip:117xxxxxx@sip12.e-fon.ch
                 SIP from address User Part: 117xxxxxx
                 SIP from address Host Part: sip12.e-fon.ch
             SIP from tag: 51a3e1d6
         To: <sip:0764823425@sip12.e-fon.ch>
             SIP to address: sip:076xxxxxxx@sip12.e-fon.ch
                 SIP to address User Part: 076xxxxxxx
                 SIP to address Host Part: sip12.e-fon.ch
         Via: SIP/2.0/UDP
217.162.xxx.xxx:5060;branch=z9hG4bK-383638-321b4bfd0c24f1f48906ff6382c3b18b;rport=5060
             Transport: UDP
             Sent-by Address: 217.162.xxx.xxx
             Sent-by port: 5060
             Branch: z9hG4bK-383638-321b4bfd0c24f1f48906ff6382c3b18b
             RPort: 5060
         Content-Length: 0

(successfull placing the call)

FS Call:
Session Initiation Protocol (404)
     Status-Line: SIP/2.0 404 Not found
         Status-Code: 404
         [Resent Packet: False]
         [Request Frame: 12659]
         [Response Time (ms): 13]
     Message Header
         Via: SIP/2.0/UDP
217.162.xxx.xxx:5080;rport=5080;branch=z9hG4bKmNHFUDQav3tUj;received=217.162.xxx.xxx
             Transport: UDP
             Sent-by Address: 217.162.xxx.xxx
             Sent-by port: 5080
             RPort: 5080
             Branch: z9hG4bKmNHFUDQav3tUj
             Received: 217.162.xxx.xxx
         From: "1000" <sip:1000@217.162.xxx.xxx>;tag=gNe9eHQm44HBj
             SIP Display info: "1000"
             SIP from address: sip:1000@217.162.xxx.xxx
                 SIP from address User Part: 1000
                 SIP from address Host Part: 217.162.xxx.xxx
             SIP from tag: gNe9eHQm44HBj
         To:
<sip:076xxxxxxx@sip12.e-fon.ch>;tag=56dedfd3319cd1b9fedc1197b499329c.26ea
             SIP to address: sip:076xxxxxxx@sip12.e-fon.ch
                 SIP to address User Part: 076xxxxxxx
                 SIP to address Host Part: sip12.e-fon.ch
             SIP to tag: 56dedfd3319cd1b9fedc1197b499329c.26ea
         Call-ID: 84f02fe4-71ef-1238-b881-00012e808c2d
         [Generated Call-ID: 84f02fe4-71ef-1238-b881-00012e808c2d]
         CSeq: 11458424 INVITE
             Sequence Number: 11458424
             Method: INVITE
         Content-Length: 0

---------- Forwarded message ----------
From: Tamer Higazi via FreeSWITCH-users <freeswitch-users@lists.freeswitch.org>
To: freeswitch-users@lists.freeswitch.org
Cc:
Bcc:
Date: Tue, 29 Oct 2019 08:56:21 -0700 (PDT)
Subject: [Freeswitch-users] replace SIP from user and host part
_________________________________________________________________________

The FreeSWITCH project is sponsored by SignalWire https://signalwire.com
Enhance your FreeSWITCH install with disruptive priced SMS and PSTN services.
Build your next product on our scalable cloud platform.

Join our online community to chat in real time https://signalwire.community

Professional FreeSWITCH Services
sales@freeswitch.com
https://freeswitch.com

Official FreeSWITCH Sites
https://freeswitch.com/oss
https://freeswitch.org/confluence
https://cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
https://freeswitch.com