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@lists.freeswitch.org <mailto:freeswitch-users@lists.freeswitch.org>> wrote:
---------- Forwarded message ---------- From: Tamer Higazi <th982a@googlemail.com <mailto:th982a@googlemail.com>> To: freeswitch-users@lists.freeswitch.org <mailto: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 <mailto:sip%3A117xxxxxx@sip12.e-fon.ch>>;tag=51a3e1d6 SIP Display info: "117xxxxxx" SIP from address: sip:117xxxxxx@sip12.e-fon.ch <mailto:sip%3A117xxxxxx@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@sip12.e-fon.ch <mailto:sip%3A0764823425@sip12.e-fon.ch>> SIP to address: sip:076xxxxxxx@sip12.e-fon.ch <mailto:sip%3A076xxxxxxx@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@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 <mailto:sip%3A076xxxxxxx@sip12.e-fon.ch>>;tag=56dedfd3319cd1b9fedc1197b499329c.26ea SIP to address: sip:076xxxxxxx@sip12.e-fon.ch <mailto:sip%3A076xxxxxxx@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@lists.freeswitch.org <mailto:freeswitch-users@lists.freeswitch.org>> To: freeswitch-users@lists.freeswitch.org <mailto: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 _________________________________________________________________________
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