[Freeswitch-users] replace SIP from user and host part
David Villasmil
david.villasmil.work at gmail.com
Tue Oct 29 16:22:44 UTC 2019
If you need to set a sip-from to be the same always, you can set that in
the gateway configuration:
<param name="from-user" value="cluecon"/>
On Tue, 29 Oct 2019 at 15:56, Tamer Higazi via FreeSWITCH-users <
freeswitch-users at lists.freeswitch.org> wrote:
>
>
>
> ---------- Forwarded message ----------
> From: Tamer Higazi <th982a at googlemail.com>
> To: freeswitch-users at 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 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
>
>
>
>
>
>
> ---------- Forwarded message ----------
> From: Tamer Higazi via FreeSWITCH-users <
> freeswitch-users at lists.freeswitch.org>
> To: freeswitch-users at 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
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>
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--
Regards,
David Villasmil
email: david.villasmil.work at gmail.com
phone: +34669448337
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