[Freeswitch-users] replace SIP from user and host part
Tamer Higazi
th982a at googlemail.com
Wed Oct 30 20:32:19 UTC 2019
Hi David,
Thanks for your idea.
But it didn't work :-(
What worked was in the dialplan this:
<action application="export" data="sip_cid_type=none"/>
<action application="export"
data="sip_from_uri=${sip_from_user}@${network_addr}"/>
It would be great having it in the gateway config.
Any ideas ?
best, Tamer
On 10/29/19 5:22 PM, David Villasmil wrote:
> 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
> <mailto:freeswitch-users at lists.freeswitch.org>> wrote:
>
>
>
>
> ---------- Forwarded message ----------
> From: Tamer Higazi <th982a at googlemail.com
> <mailto:th982a at googlemail.com>>
> To: freeswitch-users at lists.freeswitch.org
> <mailto: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
> <mailto:sip%3A117xxxxxx at sip12.e-fon.ch>>;tag=51a3e1d6
> SIP Display info: "117xxxxxx"
> SIP from address: sip:117xxxxxx at sip12.e-fon.ch
> <mailto:sip%3A117xxxxxx at sip12.e-fon.ch>
> SIP from address User Part: 117xxxxxx
> SIP from address Host Part: sip12.e-fon.ch
> <http://sip12.e-fon.ch>
> SIP from tag: 51a3e1d6
> To: <sip:0764823425 at sip12.e-fon.ch
> <mailto:sip%3A0764823425 at sip12.e-fon.ch>>
> SIP to address: sip:076xxxxxxx at sip12.e-fon.ch
> <mailto:sip%3A076xxxxxxx at sip12.e-fon.ch>
> SIP to address User Part: 076xxxxxxx
> SIP to address Host Part: sip12.e-fon.ch
> <http://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
> <mailto:sip%3A076xxxxxxx at sip12.e-fon.ch>>;tag=56dedfd3319cd1b9fedc1197b499329c.26ea
> SIP to address: sip:076xxxxxxx at sip12.e-fon.ch
> <mailto:sip%3A076xxxxxxx at sip12.e-fon.ch>
> SIP to address User Part: 076xxxxxxx
> SIP to address Host Part: sip12.e-fon.ch
> <http://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
> <mailto:freeswitch-users at lists.freeswitch.org>>
> To: freeswitch-users at lists.freeswitch.org
> <mailto: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
> _________________________________________________________________________
>
> 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 at freeswitch.com <mailto:sales at 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 at lists.freeswitch.org
> <mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> https://freeswitch.com
>
> --
> Regards,
>
> David Villasmil
> email: david.villasmil.work at gmail.com
> <mailto:david.villasmil.work at gmail.com>
> phone: +34669448337
More information about the FreeSWITCH-users
mailing list