Opensips 2.3.2

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Opensips 2.3.2

OpenSIPS - Users mailing list
Hello there

I'm having an issue with placing external calls.
I can register softphones and place calls between them fine, but have failed in
the next step which is to hook up to my SBC(working) to call the outside world.
On Opensips my interfaces (with fictional IP's are:

192.168.54.55 ENS160
(159.99.8.111 is my Public IP that points to LAN 192.168.54.55 for my
softphones to register via)
172.23.100.55 ENS224(this communicates with an interface on my SBC
172.23.100.250)

so  a call path will be:  159.99.8.111->192.168.54.55->172.23.100.55->172.
23.100.250->carrier

Im sure it has to do with the interface presented to my SBC which currently
shows as extn111@192.168.54.55.
the call hits my SBC but no further, or my routing set up.
I want to force the call to be 111@172.23.100.55 but not sure how to.
I have experimented with domains and routing but as a new Opensipper im
guessing a lot of it and trying to read online too but not much out there that
relates.
If someone can offer some examples of Dynamic routing or anything that would
help that would be great.

Many Thanks


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Re: Opensips 2.3.2

Bogdan-Andrei Iancu-2
Hi,

I'm 100% sure I understood your scenario, but what you need to do is an
interface exchange for the calls from users to the SBC. You can do this
using the force_send_socket() function
(http://www.opensips.org/Documentation/Script-CoreFunctions-2-3#toc16)
before doing the relay.

Regards,

Bogdan-Andrei Iancu
   OpenSIPS Founder and Developer
   http://www.opensips-solutions.com

On 11/21/2017 03:46 PM, mas_66--- via Users wrote:

> Hello there
>
> I'm having an issue with placing external calls.
> I can register softphones and place calls between them fine, but have failed in
> the next step which is to hook up to my SBC(working) to call the outside world.
> On Opensips my interfaces (with fictional IP's are:
>
> 192.168.54.55 ENS160
> (159.99.8.111 is my Public IP that points to LAN 192.168.54.55 for my
> softphones to register via)
> 172.23.100.55 ENS224(this communicates with an interface on my SBC
> 172.23.100.250)
>
> so  a call path will be:  159.99.8.111->192.168.54.55->172.23.100.55->172.
> 23.100.250->carrier
>
> Im sure it has to do with the interface presented to my SBC which currently
> shows as extn111@192.168.54.55.
> the call hits my SBC but no further, or my routing set up.
> I want to force the call to be 111@172.23.100.55 but not sure how to.
> I have experimented with domains and routing but as a new Opensipper im
> guessing a lot of it and trying to read online too but not much out there that
> relates.
> If someone can offer some examples of Dynamic routing or anything that would
> help that would be great.
>
> Many Thanks
>
>
> _______________________________________________
> Users mailing list
> [hidden email]
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users


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Re: Opensips 2.3.2

OpenSIPS - Users mailing list
Hi Bogdan
I tried your suggestion but unfortunately i am still presenting the incorrect
domain when i try to dial 02088253053. So i am still trying to dial out From:
"6691"<sip:6691@159.99.8.111> instead of
"6691"<sip:6691@172.23.100.55>. below is the udp stream if it helps.

Thanks
Mas



U 172.23.100.250:5060 -> 172.23.100.55:5060
OPTIONS sip:172.23.100.55:5060 SIP/2.0.
Via: SIP/2.0/UDP 172.23.100.250:5060;branch=z9hG4bK00Bc649aa94943d13d3.
From: <sip:172.23.100.250>;tag=gK007fa50f.
To: <sip:172.23.100.55>.
Call-ID: 35425181_134052342@172.23.100.250.
CSeq: 500409 OPTIONS.
Max-Forwards: 1.
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,UPDATE,
OPTIONS,MESSAGE,PUBLISH.
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-
relay, multipart/mixed.
Contact: <sip:172.23.100.250:5060>.
Content-Length: 0.
.


U 172.23.100.55:5060 -> 172.23.100.250:5060
SIP/2.0 484 Address Incomplete.
Via: SIP/2.0/UDP 172.23.100.250:5060;branch=z9hG4bK00Bc649aa94943d13d3.
From: <sip:172.23.100.250>;tag=gK007fa50f.
To: <sip:172.23.100.55>;tag=5588ad0e2fdda81c4b33ac6770683a01.1b10.
Call-ID: 35425181_134052342@172.23.100.250.
CSeq: 500409 OPTIONS.
Server: OpenSIPS (2.3.2 (x86_64/linux)).
Content-Length: 0.
.


U 172.23.100.55:5060 -> 172.23.100.250:5060
INVITE sip:02088253053@172.23.100.250 SIP/2.0.
Record-Route: <sip:172.23.100.55;r2=on;lr;did=bb3.90ad9f94>.
Record-Route: <sip:192.168.54.55;r2=on;lr;did=bb3.90ad9f94>.
Via: SIP/2.0/UDP 172.23.100.55:5060;branch=z9hG4bK2f3e.20349925.0.
Via: SIP/2.0/UDP 192.168.200.162:56511;received=55.59.210.100;branch=z9hG4bK-
524287-1---3b2989732e03e469;rport=12328.
Max-Forwards: 69.
Contact: <sip:6691@55.59.210.100:12328;rinstance=f9fd12e599850665>.
To: <sip:02088253053@172.23.100.250>.
From: "6691"<sip:6691@159.99.8.111>;tag=1a24473b.
Call-ID: 88254ZjZhZjYwYmI1N2MyYzNjODkzYTNmYzRhMzg2MWFiYTI.
CSeq: 1 INVITE.
Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, OPTIONS,
MESSAGE.
Content-Type: application/sdp.
Supported: replaces.
User-Agent: X-Lite release 5.0.3 stamp 88254.
Content-Length: 340.
P-hint: outbound.
.
v=0.
o=- 13156428154656607 1 IN IP4 192.168.200.162.
s=X-Lite release 5.0.3 stamp 88254.
c=IN IP4 192.168.200.162.
t=0 0.
m=audio 59698 RTP/AVP 9 8 120 0 84 101.
a=rtpmap:120 opus/48000/2.
a=fmtp:120 useinbandfec=1; usedtx=1; maxaveragebitrate=64000.
a=rtpmap:84 speex/16000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=sendrecv.


U 172.23.100.250:5060 -> 172.23.100.55:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP 172.23.100.55:5060;branch=z9hG4bK2f3e.20349925.0.
Via: SIP/2.0/UDP 192.168.200.162:56511;received=55.59.210.100;branch=z9hG4bK-
524287-1---3b2989732e03e469;rport=12328.
From: "6691"<sip:6691@159.99.8.111>;tag=1a24473b.
To: <sip:02088253053@172.23.100.250>;tag=gK04ead159.
Call-ID: 88254ZjZhZjYwYmI1N2MyYzNjODkzYTNmYzRhMzg2MWFiYTI.
CSeq: 1 INVITE.
Content-Length: 0.
.


U 172.23.100.250:5060 -> 172.23.100.55:5060
SIP/2.0 488 Not Acceptable Here.
Via: SIP/2.0/UDP 172.23.100.55:5060;branch=z9hG4bK2f3e.20349925.0.
Via: SIP/2.0/UDP 192.168.200.162:56511;received=55.59.210.100;branch=z9hG4bK-
524287-1---3b2989732e03e469;rport=12328.
From: "6691"<sip:6691@159.99.8.111>;tag=1a24473b.
To: <sip:02088253053@172.23.100.250>;tag=gK04ead159.
Call-ID: 88254ZjZhZjYwYmI1N2MyYzNjODkzYTNmYzRhMzg2MWFiYTI.
CSeq: 1 INVITE.
Content-Length: 0.
.


U 172.23.100.55:5060 -> 172.23.100.250:5060
ACK sip:02088253053@172.23.100.250 SIP/2.0.
Via: SIP/2.0/UDP 172.23.100.55:5060;branch=z9hG4bK2f3e.20349925.0.
From: "6691"<sip:6691@159.99.8.111>;tag=1a24473b.
Call-ID: 88254ZjZhZjYwYmI1N2MyYzNjODkzYTNmYzRhMzg2MWFiYTI.
To: <sip:02088253053@172.23.100.250>;tag=gK04ead159.
CSeq: 1 ACK.
Max-Forwards: 70.
User-Agent: OpenSIPS (2.3.2 (x86_64/linux)).
Content-Length: 0.

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Re: Opensips 2.3.2

OpenSIPS - Users mailing list
Hi Everyone 
Any idea on how to change the FROM domain on a call?
I have more than one NIC and my calls fail because the sending domain to my Carrier is incorrect 
Thanks



Sent from Samsung Mobile on O2

-------- Original message --------
From: mas_66--- via Users <[hidden email]>
Date: 29/11/2017 11:55 (GMT+00:00)
To: [hidden email], OpenSIPS users mailling list <[hidden email]>
Subject: Re: [OpenSIPS-Users] Opensips 2.3.2

Hi Bogdan
I tried your suggestion but unfortunately i am still presenting the incorrect
domain when i try to dial 02088253053. So i am still trying to dial out From:
"6691"<sip:6691@159.99.8.111> instead of
"6691"<sip:6691@172.23.100.55>. below is the udp stream if it helps.

Thanks
Mas



U 172.23.100.250:5060 -> 172.23.100.55:5060
OPTIONS sip:172.23.100.55:5060 SIP/2.0.
Via: SIP/2.0/UDP 172.23.100.250:5060;branch=z9hG4bK00Bc649aa94943d13d3.
From: <sip:172.23.100.250>;tag=gK007fa50f.
To: <sip:172.23.100.55>.
Call-ID: 35425181_134052342@172.23.100.250.
CSeq: 500409 OPTIONS.
Max-Forwards: 1.
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,UPDATE,
OPTIONS,MESSAGE,PUBLISH.
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-
relay, multipart/mixed.
Contact: <sip:172.23.100.250:5060>.
Content-Length: 0.
.


U 172.23.100.55:5060 -> 172.23.100.250:5060
SIP/2.0 484 Address Incomplete.
Via: SIP/2.0/UDP 172.23.100.250:5060;branch=z9hG4bK00Bc649aa94943d13d3.
From: <sip:172.23.100.250>;tag=gK007fa50f.
To: <sip:172.23.100.55>;tag=5588ad0e2fdda81c4b33ac6770683a01.1b10.
Call-ID: 35425181_134052342@172.23.100.250.
CSeq: 500409 OPTIONS.
Server: OpenSIPS (2.3.2 (x86_64/linux)).
Content-Length: 0.
.


U 172.23.100.55:5060 -> 172.23.100.250:5060
INVITE sip:02088253053@172.23.100.250 SIP/2.0.
Record-Route: <sip:172.23.100.55;r2=on;lr;did=bb3.90ad9f94>.
Record-Route: <sip:192.168.54.55;r2=on;lr;did=bb3.90ad9f94>.
Via: SIP/2.0/UDP 172.23.100.55:5060;branch=z9hG4bK2f3e.20349925.0.
Via: SIP/2.0/UDP 192.168.200.162:56511;received=55.59.210.100;branch=z9hG4bK-
524287-1---3b2989732e03e469;rport=12328.
Max-Forwards: 69.
Contact: <sip:6691@55.59.210.100:12328;rinstance=f9fd12e599850665>.
To: <sip:02088253053@172.23.100.250>.
From: "6691"<sip:6691@159.99.8.111>;tag=1a24473b.
Call-ID: 88254ZjZhZjYwYmI1N2MyYzNjODkzYTNmYzRhMzg2MWFiYTI.
CSeq: 1 INVITE.
Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, OPTIONS,
MESSAGE.
Content-Type: application/sdp.
Supported: replaces.
User-Agent: X-Lite release 5.0.3 stamp 88254.
Content-Length: 340.
P-hint: outbound.
.
v=0.
o=- 13156428154656607 1 IN IP4 192.168.200.162.
s=X-Lite release 5.0.3 stamp 88254.
c=IN IP4 192.168.200.162.
t=0 0.
m=audio 59698 RTP/AVP 9 8 120 0 84 101.
a=rtpmap:120 opus/48000/2.
a=fmtp:120 useinbandfec=1; usedtx=1; maxaveragebitrate=64000.
a=rtpmap:84 speex/16000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=sendrecv.


U 172.23.100.250:5060 -> 172.23.100.55:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP 172.23.100.55:5060;branch=z9hG4bK2f3e.20349925.0.
Via: SIP/2.0/UDP 192.168.200.162:56511;received=55.59.210.100;branch=z9hG4bK-
524287-1---3b2989732e03e469;rport=12328.
From: "6691"<sip:6691@159.99.8.111>;tag=1a24473b.
To: <sip:02088253053@172.23.100.250>;tag=gK04ead159.
Call-ID: 88254ZjZhZjYwYmI1N2MyYzNjODkzYTNmYzRhMzg2MWFiYTI.
CSeq: 1 INVITE.
Content-Length: 0.
.


U 172.23.100.250:5060 -> 172.23.100.55:5060
SIP/2.0 488 Not Acceptable Here.
Via: SIP/2.0/UDP 172.23.100.55:5060;branch=z9hG4bK2f3e.20349925.0.
Via: SIP/2.0/UDP 192.168.200.162:56511;received=55.59.210.100;branch=z9hG4bK-
524287-1---3b2989732e03e469;rport=12328.
From: "6691"<sip:6691@159.99.8.111>;tag=1a24473b.
To: <sip:02088253053@172.23.100.250>;tag=gK04ead159.
Call-ID: 88254ZjZhZjYwYmI1N2MyYzNjODkzYTNmYzRhMzg2MWFiYTI.
CSeq: 1 INVITE.
Content-Length: 0.
.


U 172.23.100.55:5060 -> 172.23.100.250:5060
ACK sip:02088253053@172.23.100.250 SIP/2.0.
Via: SIP/2.0/UDP 172.23.100.55:5060;branch=z9hG4bK2f3e.20349925.0.
From: "6691"<sip:6691@159.99.8.111>;tag=1a24473b.
Call-ID: 88254ZjZhZjYwYmI1N2MyYzNjODkzYTNmYzRhMzg2MWFiYTI.
To: <sip:02088253053@172.23.100.250>;tag=gK04ead159.
CSeq: 1 ACK.
Max-Forwards: 70.
User-Agent: OpenSIPS (2.3.2 (x86_64/linux)).
Content-Length: 0.

_______________________________________________
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Re: Opensips 2.3.2

Srigo Kanapathipillai
Hi, 

Did you try uac_replace_from?

Srigo

Sent from my iPhone

On 10 Dec 2017, at 11:28, mas_66 via Users <[hidden email]> wrote:

Hi Everyone 
Any idea on how to change the FROM domain on a call?
I have more than one NIC and my calls fail because the sending domain to my Carrier is incorrect 
Thanks



Sent from Samsung Mobile on O2

-------- Original message --------
From: mas_66--- via Users <[hidden email]>
Date: 29/11/2017 11:55 (GMT+00:00)
To: [hidden email], OpenSIPS users mailling list <[hidden email]>
Subject: Re: [OpenSIPS-Users] Opensips 2.3.2

Hi Bogdan
I tried your suggestion but unfortunately i am still presenting the incorrect
domain when i try to dial 02088253053. So i am still trying to dial out From:
"6691"<sip:6691@159.99.8.111> instead of
"6691"<sip:6691@172.23.100.55>. below is the udp stream if it helps.

Thanks
Mas



U 172.23.100.250:5060 -> 172.23.100.55:5060
OPTIONS sip:172.23.100.55:5060 SIP/2.0.
Via: SIP/2.0/UDP 172.23.100.250:5060;branch=z9hG4bK00Bc649aa94943d13d3.
From: <sip:172.23.100.250>;tag=gK007fa50f.
To: <sip:172.23.100.55>.
Call-ID: [hidden email].
CSeq: 500409 OPTIONS.
Max-Forwards: 1.
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,UPDATE,
OPTIONS,MESSAGE,PUBLISH.
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-
relay, multipart/mixed.
Contact: <sip:172.23.100.250:5060>.
Content-Length: 0.
.


U 172.23.100.55:5060 -> 172.23.100.250:5060
SIP/2.0 484 Address Incomplete.
Via: SIP/2.0/UDP 172.23.100.250:5060;branch=z9hG4bK00Bc649aa94943d13d3.
From: <sip:172.23.100.250>;tag=gK007fa50f.
To: <sip:172.23.100.55>;tag=5588ad0e2fdda81c4b33ac6770683a01.1b10.
Call-ID: [hidden email].
CSeq: 500409 OPTIONS.
Server: OpenSIPS (2.3.2 (x86_64/linux)).
Content-Length: 0.
.


U 172.23.100.55:5060 -> 172.23.100.250:5060
INVITE sip:02088253053@172.23.100.250 SIP/2.0.
Record-Route: <sip:172.23.100.55;r2=on;lr;did=bb3.90ad9f94>.
Record-Route: <sip:192.168.54.55;r2=on;lr;did=bb3.90ad9f94>.
Via: SIP/2.0/UDP 172.23.100.55:5060;branch=z9hG4bK2f3e.20349925.0.
Via: SIP/2.0/UDP 192.168.200.162:56511;received=55.59.210.100;branch=z9hG4bK-
524287-1---3b2989732e03e469;rport=12328.
Max-Forwards: 69.
Contact: <sip:6691@55.59.210.100:12328;rinstance=f9fd12e599850665>.
To: <sip:02088253053@172.23.100.250>.
From: "6691"<sip:6691@159.99.8.111>;tag=1a24473b.
Call-ID: 88254ZjZhZjYwYmI1N2MyYzNjODkzYTNmYzRhMzg2MWFiYTI.
CSeq: 1 INVITE.
Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, OPTIONS,
MESSAGE.
Content-Type: application/sdp.
Supported: replaces.
User-Agent: X-Lite release 5.0.3 stamp 88254.
Content-Length: 340.
P-hint: outbound.
.
v=0.
o=- 13156428154656607 1 IN IP4 192.168.200.162.
s=X-Lite release 5.0.3 stamp 88254.
c=IN IP4 192.168.200.162.
t=0 0.
m=audio 59698 RTP/AVP 9 8 120 0 84 101.
a=rtpmap:120 opus/48000/2.
a=fmtp:120 useinbandfec=1; usedtx=1; maxaveragebitrate=64000.
a=rtpmap:84 speex/16000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=sendrecv.


U 172.23.100.250:5060 -> 172.23.100.55:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP 172.23.100.55:5060;branch=z9hG4bK2f3e.20349925.0.
Via: SIP/2.0/UDP 192.168.200.162:56511;received=55.59.210.100;branch=z9hG4bK-
524287-1---3b2989732e03e469;rport=12328.
From: "6691"<sip:6691@159.99.8.111>;tag=1a24473b.
To: <sip:02088253053@172.23.100.250>;tag=gK04ead159.
Call-ID: 88254ZjZhZjYwYmI1N2MyYzNjODkzYTNmYzRhMzg2MWFiYTI.
CSeq: 1 INVITE.
Content-Length: 0.
.


U 172.23.100.250:5060 -> 172.23.100.55:5060
SIP/2.0 488 Not Acceptable Here.
Via: SIP/2.0/UDP 172.23.100.55:5060;branch=z9hG4bK2f3e.20349925.0.
Via: SIP/2.0/UDP 192.168.200.162:56511;received=55.59.210.100;branch=z9hG4bK-
524287-1---3b2989732e03e469;rport=12328.
From: "6691"<sip:6691@159.99.8.111>;tag=1a24473b.
To: <sip:02088253053@172.23.100.250>;tag=gK04ead159.
Call-ID: 88254ZjZhZjYwYmI1N2MyYzNjODkzYTNmYzRhMzg2MWFiYTI.
CSeq: 1 INVITE.
Content-Length: 0.
.


U 172.23.100.55:5060 -> 172.23.100.250:5060
ACK sip:02088253053@172.23.100.250 SIP/2.0.
Via: SIP/2.0/UDP 172.23.100.55:5060;branch=z9hG4bK2f3e.20349925.0.
From: "6691"<sip:6691@159.99.8.111>;tag=1a24473b.
Call-ID: 88254ZjZhZjYwYmI1N2MyYzNjODkzYTNmYzRhMzg2MWFiYTI.
To: <sip:02088253053@172.23.100.250>;tag=gK04ead159.
CSeq: 1 ACK.
Max-Forwards: 70.
User-Agent: OpenSIPS (2.3.2 (x86_64/linux)).
Content-Length: 0.

_______________________________________________
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_______________________________________________
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[hidden email]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users

_______________________________________________
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Re: Opensips 2.3.2

Razvan Crainea-3
In reply to this post by OpenSIPS - Users mailing list
Check the uac_replace_from() function[1].

[1] http://www.opensips.org/html/docs/modules/2.3.x/uac.html#idp5591536

Best regards,
Răzvan Crainea
OpenSIPS Developer
www.opensips-solutions.com
On 12/10/2017 12:28 PM, mas_66 via Users wrote:
Hi Everyone 
Any idea on how to change the FROM domain on a call?
I have more than one NIC and my calls fail because the sending domain to my Carrier is incorrect 
Thanks



Sent from Samsung Mobile on O2

-------- Original message --------
From: mas_66--- via Users [hidden email]
Date: 29/11/2017 11:55 (GMT+00:00)
To: [hidden email], OpenSIPS users mailling list [hidden email]
Subject: Re: [OpenSIPS-Users] Opensips 2.3.2

Hi Bogdan
I tried your suggestion but unfortunately i am still presenting the incorrect
domain when i try to dial 02088253053. So i am still trying to dial out From:
"6691"[hidden email] instead of
"6691"[hidden email]. below is the udp stream if it helps.

Thanks
Mas



U 172.23.100.250:5060 -> 172.23.100.55:5060
OPTIONS sip:172.23.100.55:5060 SIP/2.0.
Via: SIP/2.0/UDP 172.23.100.250:5060;branch=z9hG4bK00Bc649aa94943d13d3.
From: <sip:172.23.100.250>;tag=gK007fa50f.
To: <sip:172.23.100.55>.
Call-ID: [hidden email].
CSeq: 500409 OPTIONS.
Max-Forwards: 1.
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,UPDATE,
OPTIONS,MESSAGE,PUBLISH.
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-
relay, multipart/mixed.
Contact: <sip:172.23.100.250:5060>.
Content-Length: 0.
.


U 172.23.100.55:5060 -> 172.23.100.250:5060
SIP/2.0 484 Address Incomplete.
Via: SIP/2.0/UDP 172.23.100.250:5060;branch=z9hG4bK00Bc649aa94943d13d3.
From: <sip:172.23.100.250>;tag=gK007fa50f.
To: <sip:172.23.100.55>;tag=5588ad0e2fdda81c4b33ac6770683a01.1b10.
Call-ID: [hidden email].
CSeq: 500409 OPTIONS.
Server: OpenSIPS (2.3.2 (x86_64/linux)).
Content-Length: 0.
.


U 172.23.100.55:5060 -> 172.23.100.250:5060
INVITE [hidden email] SIP/2.0.
Record-Route: <sip:172.23.100.55;r2=on;lr;did=bb3.90ad9f94>.
Record-Route: <sip:192.168.54.55;r2=on;lr;did=bb3.90ad9f94>.
Via: SIP/2.0/UDP 172.23.100.55:5060;branch=z9hG4bK2f3e.20349925.0.
Via: SIP/2.0/UDP 192.168.200.162:56511;received=55.59.210.100;branch=z9hG4bK-
524287-1---3b2989732e03e469;rport=12328.
Max-Forwards: 69.
Contact: [hidden email].
To: [hidden email].
From: "6691"[hidden email];tag=1a24473b.
Call-ID: 88254ZjZhZjYwYmI1N2MyYzNjODkzYTNmYzRhMzg2MWFiYTI.
CSeq: 1 INVITE.
Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, OPTIONS,
MESSAGE.
Content-Type: application/sdp.
Supported: replaces.
User-Agent: X-Lite release 5.0.3 stamp 88254.
Content-Length: 340.
P-hint: outbound.
.
v=0.
o=- 13156428154656607 1 IN IP4 192.168.200.162.
s=X-Lite release 5.0.3 stamp 88254.
c=IN IP4 192.168.200.162.
t=0 0.
m=audio 59698 RTP/AVP 9 8 120 0 84 101.
a=rtpmap:120 opus/48000/2.
a=fmtp:120 useinbandfec=1; usedtx=1; maxaveragebitrate=64000.
a=rtpmap:84 speex/16000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=sendrecv.


U 172.23.100.250:5060 -> 172.23.100.55:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP 172.23.100.55:5060;branch=z9hG4bK2f3e.20349925.0.
Via: SIP/2.0/UDP 192.168.200.162:56511;received=55.59.210.100;branch=z9hG4bK-
524287-1---3b2989732e03e469;rport=12328.
From: "6691"[hidden email];tag=1a24473b.
To: [hidden email];tag=gK04ead159.
Call-ID: 88254ZjZhZjYwYmI1N2MyYzNjODkzYTNmYzRhMzg2MWFiYTI.
CSeq: 1 INVITE.
Content-Length: 0.
.


U 172.23.100.250:5060 -> 172.23.100.55:5060
SIP/2.0 488 Not Acceptable Here.
Via: SIP/2.0/UDP 172.23.100.55:5060;branch=z9hG4bK2f3e.20349925.0.
Via: SIP/2.0/UDP 192.168.200.162:56511;received=55.59.210.100;branch=z9hG4bK-
524287-1---3b2989732e03e469;rport=12328.
From: "6691"[hidden email];tag=1a24473b.
To: [hidden email];tag=gK04ead159.
Call-ID: 88254ZjZhZjYwYmI1N2MyYzNjODkzYTNmYzRhMzg2MWFiYTI.
CSeq: 1 INVITE.
Content-Length: 0.
.


U 172.23.100.55:5060 -> 172.23.100.250:5060
ACK [hidden email] SIP/2.0.
Via: SIP/2.0/UDP 172.23.100.55:5060;branch=z9hG4bK2f3e.20349925.0.
From: "6691"[hidden email];tag=1a24473b.
Call-ID: 88254ZjZhZjYwYmI1N2MyYzNjODkzYTNmYzRhMzg2MWFiYTI.
To: [hidden email];tag=gK04ead159.
CSeq: 1 ACK.
Max-Forwards: 70.
User-Agent: OpenSIPS (2.3.2 (x86_64/linux)).
Content-Length: 0.

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_______________________________________________
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Re: Opensips 2.3.2

Bogdan-Andrei Iancu-2
In reply to this post by OpenSIPS - Users mailing list
Hi Mas,

Use the uac_replace_from() if you need to change the URI or display in the FROM hdr:
    http://www.opensips.org/html/docs/modules/2.3.x/uac.html#idp5583120

Regards,
Bogdan-Andrei Iancu
  OpenSIPS Founder and Developer
  http://www.opensips-solutions.com
On 12/10/2017 12:28 PM, mas_66 wrote:
Hi Everyone 
Any idea on how to change the FROM domain on a call?
I have more than one NIC and my calls fail because the sending domain to my Carrier is incorrect 
Thanks



Sent from Samsung Mobile on O2

-------- Original message --------
From: mas_66--- via Users [hidden email]
Date: 29/11/2017 11:55 (GMT+00:00)
To: [hidden email], OpenSIPS users mailling list [hidden email]
Subject: Re: [OpenSIPS-Users] Opensips 2.3.2

Hi Bogdan
I tried your suggestion but unfortunately i am still presenting the incorrect
domain when i try to dial 02088253053. So i am still trying to dial out From:
"6691"<a class="moz-txt-link-rfc2396E" href="sip:6691@159.99.8.111"><sip:6691@159.99.8.111> instead of
"6691"<a class="moz-txt-link-rfc2396E" href="sip:6691@172.23.100.55"><sip:6691@172.23.100.55>. below is the udp stream if it helps.

Thanks
Mas



U 172.23.100.250:5060 -> 172.23.100.55:5060
OPTIONS <a class="moz-txt-link-freetext" href="sip:172.23.100.55:5060">sip:172.23.100.55:5060 SIP/2.0.
Via: SIP/2.0/UDP 172.23.100.250:5060;branch=z9hG4bK00Bc649aa94943d13d3.
From: <a class="moz-txt-link-rfc2396E" href="sip:172.23.100.250"><sip:172.23.100.250>;tag=gK007fa50f.
To: <a class="moz-txt-link-rfc2396E" href="sip:172.23.100.55"><sip:172.23.100.55>.
Call-ID: [hidden email].
CSeq: 500409 OPTIONS.
Max-Forwards: 1.
Allow: INVITE,ACK,CANCEL,BYE,REGISTER,REFER,INFO,SUBSCRIBE,NOTIFY,UPDATE,
OPTIONS,MESSAGE,PUBLISH.
Accept: application/sdp, application/isup, application/dtmf, application/dtmf-
relay, multipart/mixed.
Contact: <a class="moz-txt-link-rfc2396E" href="sip:172.23.100.250:5060"><sip:172.23.100.250:5060>.
Content-Length: 0.
.


U 172.23.100.55:5060 -> 172.23.100.250:5060
SIP/2.0 484 Address Incomplete.
Via: SIP/2.0/UDP 172.23.100.250:5060;branch=z9hG4bK00Bc649aa94943d13d3.
From: <a class="moz-txt-link-rfc2396E" href="sip:172.23.100.250"><sip:172.23.100.250>;tag=gK007fa50f.
To: <a class="moz-txt-link-rfc2396E" href="sip:172.23.100.55"><sip:172.23.100.55>;tag=5588ad0e2fdda81c4b33ac6770683a01.1b10.
Call-ID: [hidden email].
CSeq: 500409 OPTIONS.
Server: OpenSIPS (2.3.2 (x86_64/linux)).
Content-Length: 0.
.


U 172.23.100.55:5060 -> 172.23.100.250:5060
INVITE <a class="moz-txt-link-freetext" href="sip:02088253053@172.23.100.250">sip:02088253053@172.23.100.250 SIP/2.0.
Record-Route: <a class="moz-txt-link-rfc2396E" href="sip:172.23.100.55;r2=on;lr;did=bb3.90ad9f94"><sip:172.23.100.55;r2=on;lr;did=bb3.90ad9f94>.
Record-Route: <a class="moz-txt-link-rfc2396E" href="sip:192.168.54.55;r2=on;lr;did=bb3.90ad9f94"><sip:192.168.54.55;r2=on;lr;did=bb3.90ad9f94>.
Via: SIP/2.0/UDP 172.23.100.55:5060;branch=z9hG4bK2f3e.20349925.0.
Via: SIP/2.0/UDP 192.168.200.162:56511;received=55.59.210.100;branch=z9hG4bK-
524287-1---3b2989732e03e469;rport=12328.
Max-Forwards: 69.
Contact: <a class="moz-txt-link-rfc2396E" href="sip:6691@55.59.210.100:12328;rinstance=f9fd12e599850665"><sip:6691@55.59.210.100:12328;rinstance=f9fd12e599850665>.
To: <a class="moz-txt-link-rfc2396E" href="sip:02088253053@172.23.100.250"><sip:02088253053@172.23.100.250>.
From: "6691"<a class="moz-txt-link-rfc2396E" href="sip:6691@159.99.8.111"><sip:6691@159.99.8.111>;tag=1a24473b.
Call-ID: 88254ZjZhZjYwYmI1N2MyYzNjODkzYTNmYzRhMzg2MWFiYTI.
CSeq: 1 INVITE.
Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, OPTIONS,
MESSAGE.
Content-Type: application/sdp.
Supported: replaces.
User-Agent: X-Lite release 5.0.3 stamp 88254.
Content-Length: 340.
P-hint: outbound.
.
v=0.
o=- 13156428154656607 1 IN IP4 192.168.200.162.
s=X-Lite release 5.0.3 stamp 88254.
c=IN IP4 192.168.200.162.
t=0 0.
m=audio 59698 RTP/AVP 9 8 120 0 84 101.
a=rtpmap:120 opus/48000/2.
a=fmtp:120 useinbandfec=1; usedtx=1; maxaveragebitrate=64000.
a=rtpmap:84 speex/16000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=sendrecv.


U 172.23.100.250:5060 -> 172.23.100.55:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP 172.23.100.55:5060;branch=z9hG4bK2f3e.20349925.0.
Via: SIP/2.0/UDP 192.168.200.162:56511;received=55.59.210.100;branch=z9hG4bK-
524287-1---3b2989732e03e469;rport=12328.
From: "6691"<a class="moz-txt-link-rfc2396E" href="sip:6691@159.99.8.111"><sip:6691@159.99.8.111>;tag=1a24473b.
To: <a class="moz-txt-link-rfc2396E" href="sip:02088253053@172.23.100.250"><sip:02088253053@172.23.100.250>;tag=gK04ead159.
Call-ID: 88254ZjZhZjYwYmI1N2MyYzNjODkzYTNmYzRhMzg2MWFiYTI.
CSeq: 1 INVITE.
Content-Length: 0.
.


U 172.23.100.250:5060 -> 172.23.100.55:5060
SIP/2.0 488 Not Acceptable Here.
Via: SIP/2.0/UDP 172.23.100.55:5060;branch=z9hG4bK2f3e.20349925.0.
Via: SIP/2.0/UDP 192.168.200.162:56511;received=55.59.210.100;branch=z9hG4bK-
524287-1---3b2989732e03e469;rport=12328.
From: "6691"<a class="moz-txt-link-rfc2396E" href="sip:6691@159.99.8.111"><sip:6691@159.99.8.111>;tag=1a24473b.
To: <a class="moz-txt-link-rfc2396E" href="sip:02088253053@172.23.100.250"><sip:02088253053@172.23.100.250>;tag=gK04ead159.
Call-ID: 88254ZjZhZjYwYmI1N2MyYzNjODkzYTNmYzRhMzg2MWFiYTI.
CSeq: 1 INVITE.
Content-Length: 0.
.


U 172.23.100.55:5060 -> 172.23.100.250:5060
ACK <a class="moz-txt-link-freetext" href="sip:02088253053@172.23.100.250">sip:02088253053@172.23.100.250 SIP/2.0.
Via: SIP/2.0/UDP 172.23.100.55:5060;branch=z9hG4bK2f3e.20349925.0.
From: "6691"<a class="moz-txt-link-rfc2396E" href="sip:6691@159.99.8.111"><sip:6691@159.99.8.111>;tag=1a24473b.
Call-ID: 88254ZjZhZjYwYmI1N2MyYzNjODkzYTNmYzRhMzg2MWFiYTI.
To: <a class="moz-txt-link-rfc2396E" href="sip:02088253053@172.23.100.250"><sip:02088253053@172.23.100.250>;tag=gK04ead159.
CSeq: 1 ACK.
Max-Forwards: 70.
User-Agent: OpenSIPS (2.3.2 (x86_64/linux)).
Content-Length: 0.

_______________________________________________
Users mailing list
[hidden email]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users


_______________________________________________
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Re: Opensips 2.3.2

OpenSIPS - Users mailing list
Thanks Bogdan and Razvan

I have inserted the function in the location as below  in opensips.cfg with no
success im afraid.
Have i put it in the correct location? i want to change the FROM URI 1.1.1.1 to
2.2.2.2

thanks
route[relay] {
        # for INVITEs enable some additional helper routes
        if (is_method("INVITE")) {

                if (isflagset(NAT)) {
                        rtpproxy_offer("ro");
                }

                t_on_branch("per_branch_ops");
                t_on_reply("handle_nat");
                t_on_failure("missed_call");
        }

        if (isflagset(NAT)) {
                add_rr_param(";nat=yes");
                }

        if (!t_relay()) {
                send_reply("500","Internal Error");
        };
        exit;
}




branch_route[per_branch_ops] {
        uac_replace_from("$avp(display)","$avp(uri)");
        xlog("new branch at $ru\n");
}


onreply_route[handle_nat] {
        if (nat_uac_test("1"))
                fix_nated_contact();
        if ( isflagset(NAT) )
                rtpproxy_answer("ro");
        xlog("incoming reply\n");
}


failure_route[missed_call] {
        if (t_was_cancelled()) {
                exit;
        }

        # uncomment the following lines if you want to block client
        # redirect based on 3xx replies.
        ##if (t_check_status("3[0-9][0-9]")) {
        ##t_reply("404","Not found");
        ##      exit;
        ##}


}



local_route {
        if (is_method("BYE") && $DLG_dir=="UPSTREAM") {

                acc_db_request("200 Dialog Timeout", "acc");

        }
}


_______________________________________________
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Re: Opensips 2.3.2

Razvan Crainea-3
Yes, the call seems to be in the right place.
Are you sure that $avp(display) and $avp(uri) contain the right
information? Did you add some logs to make sure? Are there any errors in
the script? Is anything changed in the From header after forwarding the
request?

Best regards,

Răzvan Crainea
OpenSIPS Developer
www.opensips-solutions.com

On 12/12/2017 06:55 PM, mas_66--- via Users wrote:

> Thanks Bogdan and Razvan
>
> I have inserted the function in the location as below  in opensips.cfg with no
> success im afraid.
> Have i put it in the correct location? i want to change the FROM URI 1.1.1.1 to
> 2.2.2.2
>
> thanks
> route[relay] {
>          # for INVITEs enable some additional helper routes
>          if (is_method("INVITE")) {
>
>                  if (isflagset(NAT)) {
>                          rtpproxy_offer("ro");
>                  }
>
>                  t_on_branch("per_branch_ops");
>                  t_on_reply("handle_nat");
>                  t_on_failure("missed_call");
>          }
>
>          if (isflagset(NAT)) {
>                  add_rr_param(";nat=yes");
>                  }
>
>          if (!t_relay()) {
>                  send_reply("500","Internal Error");
>          };
>          exit;
> }
>
>
>
>
> branch_route[per_branch_ops] {
>          uac_replace_from("$avp(display)","$avp(uri)");
>          xlog("new branch at $ru\n");
> }
>
>
> onreply_route[handle_nat] {
>          if (nat_uac_test("1"))
>                  fix_nated_contact();
>          if ( isflagset(NAT) )
>                  rtpproxy_answer("ro");
>          xlog("incoming reply\n");
> }
>
>
> failure_route[missed_call] {
>          if (t_was_cancelled()) {
>                  exit;
>          }
>
>          # uncomment the following lines if you want to block client
>          # redirect based on 3xx replies.
>          ##if (t_check_status("3[0-9][0-9]")) {
>          ##t_reply("404","Not found");
>          ##      exit;
>          ##}
>
>
> }
>
>
>
> local_route {
>          if (is_method("BYE") && $DLG_dir=="UPSTREAM") {
>
>                  acc_db_request("200 Dialog Timeout", "acc");
>
>          }
> }
>
>
> _______________________________________________
> Users mailing list
> [hidden email]
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users


_______________________________________________
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[hidden email]
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