[Spce-user] Incident [#SRY04139] Carrying original destination when forwarding

Fri Jun 21 02:54:19 EDT 2019

Hi,<br />
<br />
<br />
>> Our client PBX uses the RURI by default, and I changed it to use the<br />
>> "To:" section of the invite<br />
<br />
<br />
>Why that? It should route calls via R-URI not To header.<br />
<br />
The "outbound_to_user" option only modifies the To header and not the R-URI, even if "e164_to_r-uri" is enabled. So the only way for me to get it working on the client PBX was with the To header<br />
<br />
>><br />
>> What I have noticed now is that the 1st failover works as expected,<br />
>> but the 2nd failover ends up with the RURI of the 1st failover in its "To:"<br />
>><br />
<br />
>Is this a peering failover or a CF scenario ?<br />
<br />
Its a peering failover scenario<br />
 
<hr />
<br />
Date: 2019-06-21 08:47:53 AM<br />
Subject: FW: [Spce-user] Incident [#SRY04139] Carrying original destination when forwarding<br />
From: morne at tenacit.net<br />
To: linksilent at tenacit.net<br />
<br />
<br />
<br />
-----Original Message-----<br />
From: Spce-user <spce-user-bounces at lists.sipwise.com> On Behalf Of Daniel Grotti<br />
Sent: Thursday, June 20, 2019 11:48 AM<br />
To: spce-user at lists.sipwise.com<br />
Subject: Re: [Spce-user] Incident [#SRY04139] Carrying original destination when forwarding<br />
<br />
Hi,<br />
<br />
<br />
On 6/20/19 10:44 AM, morne at tenacit.net wrote:<br />
> Hi,<br />
><br />
> I managed to get it semi working (After successfully upgrading my<br />
> servers from prehistoric versions).<br />
><br />
> Our client PBX uses the RURI by default, and I changed it to use the<br />
> "To:" section of the invite<br />
<br />
<br />
Why that? It should route calls via R-URI not To header.<br />
<br />
<br />
><br />
> What I have noticed now is that the 1st failover works as expected,<br />
> but the 2nd failover ends up with the RURI of the 1st failover in its "To:"<br />
><br />
<br />
Is this a peering failover or a CF scenario ?<br />
<br />
<br />
<br />
> *Examples:*<br />
> ########################################## ORIGINAL<br />
> ###################################################<br />
><br />
> INVITE sip:01xxxxxxx9 at 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 at 2.2.2.2>;tag=5xiqh2lj6c'<br />
> To: <sip:01xxxxxxx9 at 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 at 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,<br />
> 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 />
><br />
> INVITE sip:Tenant-Failover1 at ClientIP1:5060 SIP/2.0'<br />
> Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bKka512a6U;rport'<br />
> From:<br />
> <sip:27xxxxxxxx2 at exampledomain.com>;tag=4532BC27-5D0A53F4000EA445-83A95700'<br />
> To: <sip:01xxxxxxx9 at 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,<br />
> MESSAGE, UPDATE'<br />
> Supported: 100rel, replaces, from-change'<br />
> P-NGCP-Caller-Info:<br />
> <sip:Tenant at exampledomain.com>;ip=127.0.0.1;port=5080;primary=27xxxxxxxx3'<br />
> P-NGCP-Callee-Info:<br />
> <sip:Tenant-Failover1 at 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 />
><br />
> INVITE sip:Tenant-Failover2 at ClientIP2:5060 SIP/2.0'<br />
> Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bKka512a6U;rport'<br />
> From:<br />
> <sip:27xxxxxxxx2 at exampledomain.com>;tag=4532BC27-5D0A53F4000EA445-83A95700'<br />
> To: <sip:Tenant-Failover1 at 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,<br />
> MESSAGE, UPDATE'<br />
> Supported: 100rel, replaces, from-change'<br />
> P-NGCP-Caller-Info:<br />
> <sip:Tenant-Failover1 at exampledomain.com>;ip=127.0.0.1;port=5080;primary=27xxxxxxxx3'<br />
> P-NGCP-Callee-Info:<br />
> <sip:Tenant-Failover2 at 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<br />
> failover.<br />
><br />
> Is there a way that I can carry it over? Maybe change the RURI at the<br />
> first failover? Do I need to change something in the template files? I<br />
> have enabled the "e164_to_ruri" but it doesnt make a difference to the<br />
> invite.<br />
<br />
<br />
<br />
You can try to pay with the preferences "outbound_to_user".<br />
<br />
<br />
<br />
><br />
> Kind Regards,<br />
><br />
> *Morne du Plessis*<br />
><br />
> Senior Network/Voice Engineer / Department Manager<br />
><br />
> *TenacIT***<br />
><br />
> *Strategic IT Consulting *•*Advanced Networking *•<br />
><br />
> *Custom Development *•*Hosting *•*Syspro Support *<br />
><br />
> Tel: 041 10 10 100<br />
><br />
> Web: http://www.tenacit.net <http://www.tenacit.net/><br />
><br />
> PBefore printing this email please think about the environment<br />
><br />
> ----------------------------------------------------------------------<br />
> --<br />
><br />
> Date: 2018-10-22 12:02:13 PM<br />
> Subject: Re: [Spce-user] Incident [#SRY04139] Carrying original<br />
> destination when forwarding<br />
> From: apogrebennyk at sipwise.com<br />
> To: Spce-user at lists.sipwise.com<br />
> Cc: linksilent at app.tenacit.net<br />
><br />
> Hi,<br />
><br />
> On 10/18/2018 01:57 PM, linksilent at 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 "Original<br />
> (Forwarding) called user" or "Received To header" - on the call<br />
> receiving side (user/domain).<br />
><br />
> This should help.<br />
> Andrew<br />
><br />
> _______________________________________________<br />
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> Spce-user at lists.sipwise.com<br />
> https://lists.sipwise.com/listinfo/spce-user<br />
><br />
<br />
<br />
<br />
Daniel Grotti<br />
<br />
Head of Customer Support Sipwise GmbH<br />
e: dgrotti at sipwise.com Europaring F15<br />
t: +43(0)130120332 A-2345 Brunn Am Gebirge<br />
w: www.sipwise.com FN: 305595f FG: LG Wiener Neustadt<br />
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