replace SIP from user and host part

[Tamer Higazi]

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 at 0:0:0:0:0:0:0:0
         [Generated Call-ID: 
bb85334b160ce6cb26ab480fd2a13236 at 0:0:0:0:0:0:0:0]
         CSeq: 2 INVITE
             Sequence Number: 2
             Method: INVITE
         From: "117xxxxxx" <sip:117xxxxxx at sip12.e-fon.ch>;tag=51a3e1d6
             SIP Display info: "117xxxxxx"
             SIP from address: sip:117xxxxxx at 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 at sip12.e-fon.ch>
             SIP to address: sip:076xxxxxxx at 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 at 217.162.xxx.xxx>;tag=gNe9eHQm44HBj
             SIP Display info: "1000"
             SIP from address: sip:1000 at 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 at sip12.e-fon.ch>;tag=56dedfd3319cd1b9fedc1197b499329c.26ea
             SIP to address: sip:076xxxxxxx at 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