Hi,<br />
<br />
I managed to get it semi working (After successfully upgrading my servers from prehistoric versions).<br />
<br />
Our client PBX uses the RURI by default, and I changed it to use the "To:" section of the invite<br />
<br />
What I have noticed now is that the 1st failover works as expected, but the 2nd failover ends up with the RURI of the 1st failover in its "To:"<br />
<br />
<strong>Examples:</strong><br />
########################################## ORIGINAL ###################################################<br />
<br />
INVITE sip:01xxxxxxx9@2.2.2.2;user=phone SIP/2.0'<br />
Via: SIP/2.0/UDP 1.1.1.1:2049;branch=z9hG4bK-li82baliyyct;rport'<br />
From: "27xxxxxxxx2" <sip:test@2.2.2.2>;tag=5xiqh2lj6c'<br />
To: <sip:01xxxxxxx9@2.2.2.2;user=phone>'<br />
Call-ID: 313536303935383439353536393239-davtmohvrxwe'<br />
CSeq: 1 INVITE'<br />
Max-Forwards: 70'<br />
User-Agent: '<br />
Contact: <sip:test@1.1.1.1:2049;line=beoci8wp>;reg-id=1'<br />
X-Serialnumber: 0xxx1xxxxxx8'<br />
P-Key-Flags: keys="3"'<br />
Accept: application/sdp'<br />
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO, UPDATE'<br />
Allow-Events: talk, hold, refer, call-info'<br />
Supported: timer, 100rel, replaces, from-change'<br />
Session-Expires: 3600'<br />
Min-SE: 90'<br />
Content-Type: application/sdp'<br />
Content-Length: 427'<br />
<br />
######################################### 1st Failover ##################################################<br />
<br />
INVITE sip:Tenant-Failover1@ClientIP1:5060 SIP/2.0'<br />
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bKka512a6U;rport'<br />
From: <sip:27xxxxxxxx2@exampledomain.com>;tag=4532BC27-5D0A53F4000EA445-83A95700'<br />
To: <sip:01xxxxxxx9@exampledomain.com>'<br />
CSeq: 10 INVITE'<br />
Call-ID: 313536303935373933363238313932-lxxe06we83v8_b2b-1'<br />
Route: <sip:127.0.0.1:5060;lr;lr>'<br />
Max-Forwards: 70'<br />
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, UPDATE'<br />
Supported: 100rel, replaces, from-change'<br />
P-NGCP-Caller-Info: <sip:Tenant@exampledomain.com>;ip=127.0.0.1;port=5080;primary=27xxxxxxxx3'<br />
P-NGCP-Callee-Info: <sip:Tenant-Failover1@exampledomain.com>;ip=127.0.0.1;port=5060;primary=27xxxxxxxx4'<br />
P-D-Uri: sip:127.0.0.1:5060;lr'<br />
Content-Type: application/sdp'<br />
Contact: <sip:127.0.0.1:5080;transport=udp>'<br />
Content-Length: 455'<br />
<br />
######################################### 2nd Failover ##################################################<br />
<br />
INVITE sip:Tenant-Failover2@ClientIP2:5060 SIP/2.0'<br />
Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bKka512a6U;rport'<br />
From: <sip:27xxxxxxxx2@exampledomain.com>;tag=4532BC27-5D0A53F4000EA445-83A95700'<br />
To: <sip:Tenant-Failover1@exampledomain.com>'<br />
CSeq: 10 INVITE'<br />
Call-ID: 313536303936363439373236333036-f8f5lkiwc31n_b2b-1'<br />
Route: <sip:127.0.0.1:5060;lr;lr>'<br />
Max-Forwards: 70'<br />
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, UPDATE'<br />
Supported: 100rel, replaces, from-change'<br />
P-NGCP-Caller-Info: <sip:Tenant-Failover1@exampledomain.com>;ip=127.0.0.1;port=5080;primary=27xxxxxxxx3'<br />
P-NGCP-Callee-Info: <sip:Tenant-Failover2@exampledomain.com>;ip=127.0.0.1;port=5060;primary=27xxxxxxxx4'<br />
P-D-Uri: sip:127.0.0.1:5060;lr'<br />
Content-Type: application/sdp'<br />
Contact: <sip:127.0.0.1:5080;transport=udp>'<br />
Content-Length: 455'<br />
<br />
<br />
As you can see in the 2nd Failover, the "To:" is the RURI of the 1st failover.<br />
<br />
Is there a way that I can carry it over? Maybe change the RURI at the first failover? Do I need to change something in the template files? I have enabled the "e164_to_ruri" but it doesnt make a difference to the invite. <br />
<p class="MsoNormal">
<span lang="EN-ZA" style="font-size:10.0pt;color:gray;
mso-ansi-language:EN-ZA">Kind Regards,</span>
</p>
<p class="MsoNormal">
<span lang="EN-ZA" style="color:#1F497D;mso-ansi-language:
EN-ZA"> </span>
</p>
<p class="MsoNormal">
<b><span lang="EN-ZA" style="color:#79AE52;mso-ansi-language:
EN-ZA">Morne du Plessis</span></b>
</p>
<p class="MsoNormal">
<span style="font-size:10.0pt;color:gray">Senior Network/Voice Engineer / Department Manager</span>
</p>
<p class="MsoNormal">
<span style="font-size:24px;"><b><span lang="EN-ZA" style="color:#79AE52;mso-ansi-language:
EN-ZA">TenacIT</span></b></span><span style="font-size:22px;"><strong> </strong></span>
</p>
<p class="MsoNormal">
<b><span lang="EN-ZA" style="color:#1F497D;mso-ansi-language:
EN-ZA">Strategic IT Consulting </span></b><span lang="EN-ZA" style="color:#79AE52;
mso-ansi-language:EN-ZA">•</span><b><span lang="EN-ZA" style="color:#1F497D;
mso-ansi-language:EN-ZA"> Advanced Networking </span></b><span lang="EN-ZA" style="color:#79AE52;mso-ansi-language:EN-ZA">•</span>
</p>
<p class="MsoNormal">
<b><span lang="EN-ZA" style="color:#1F497D;mso-ansi-language:
EN-ZA">Custom Development </span></b><span lang="EN-ZA" style="color:#79AE52;
mso-ansi-language:EN-ZA">•</span><b><span lang="EN-ZA" style="color:#1F497D;
mso-ansi-language:EN-ZA"> Hosting </span></b><span lang="EN-ZA" style="color:
#79AE52;mso-ansi-language:EN-ZA">•</span><b><span lang="EN-ZA" style="color:#1F497D;
mso-ansi-language:EN-ZA"> Syspro Support </span></b>
</p>
<p class="MsoNormal">
<span lang="EN-ZA" style="font-size:9.0pt;color:#A6A6A6;
mso-ansi-language:EN-ZA">Tel: </span><span lang="EN-ZA" style="font-size:9.0pt;
color:gray;mso-ansi-language:EN-ZA">041 10 10 100 </span>
</p>
<p class="MsoNormal">
<span lang="EN-ZA" style="font-size:9.0pt;color:#A6A6A6;
mso-ansi-language:EN-ZA">Web: </span><span lang="EN-ZA" style="mso-ansi-language:
EN-ZA"><a href="http://www.tenacit.net/"><span style="font-size:9.0pt;
color:#BCE292">http://www.tenacit.net</span></a></span>
</p>
<span lang="EN-ZA" style="font-size:13.5pt;font-family:Webdings;mso-fareast-font-family:
"Times New Roman";mso-fareast-theme-font:minor-fareast;mso-bidi-font-family:
"Times New Roman";mso-bidi-theme-font:minor-bidi;color:green;mso-ansi-language:
EN-ZA;mso-fareast-language:EN-US;mso-bidi-language:AR-SA">P</span><span lang="EN-ZA" style="font-size:10.0pt;font-family:"Calibri",sans-serif;mso-ascii-theme-font:
minor-latin;mso-fareast-font-family:"Times New Roman";mso-fareast-theme-font:
minor-fareast;mso-hansi-theme-font:minor-latin;mso-bidi-font-family:"Times New Roman";
mso-bidi-theme-font:minor-bidi;color:navy;mso-ansi-language:EN-ZA;mso-fareast-language:
EN-US;mso-bidi-language:AR-SA"> </span><span lang="EN-ZA" style="font-size:10.0pt;
font-family:"Calibri",sans-serif;mso-ascii-theme-font:minor-latin;mso-fareast-font-family:
"Times New Roman";mso-fareast-theme-font:minor-fareast;mso-hansi-theme-font:
minor-latin;mso-bidi-font-family:"Times New Roman";mso-bidi-theme-font:minor-bidi;
color:green;mso-ansi-language:EN-ZA;mso-fareast-language:EN-US;mso-bidi-language:
AR-SA">Before printing this email please think about the environment</span><span lang="EN-ZA" style="font-size:10.0pt;font-family:"Calibri",sans-serif;mso-ascii-theme-font:
minor-latin;mso-fareast-font-family:"Times New Roman";mso-fareast-theme-font:
minor-fareast;mso-hansi-theme-font:minor-latin;mso-bidi-font-family:"Times New Roman";
mso-bidi-theme-font:minor-bidi;color:navy;mso-ansi-language:EN-ZA;mso-fareast-language:
EN-US;mso-bidi-language:AR-SA"> </span><br />
<br />
<hr />
<br />
Date: 2018-10-22 12:02:13 PM<br />
Subject: Re: [Spce-user] Incident [#SRY04139] Carrying original destination when forwarding<br />
From: apogrebennyk@sipwise.com<br />
To: Spce-user@lists.sipwise.com<br />
Cc: linksilent@app.tenacit.net<br />
<br />
Hi,<br />
<br />
On 10/18/2018 01:57 PM, linksilent@app.tenacit.net wrote:<br />
> The part that I want to stay the same after the forward is the "To:"<br />
> number before the @<br />
<br />
as explained try setting the preference outbound_to_user to<br />
"Original (Forwarding) called user" or "Received To header" - on the<br />
call receiving side (user/domain).<br />
<br />
This should help.<br />
Andrew