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

[Daniel Grotti]

Hi,


On 6/20/19 10:44 AM, morne at tenacit.net wrote:
> Hi,
> 
> I managed to get it semi working (After successfully upgrading my 
> servers from prehistoric versions).
> 
> Our client PBX uses the RURI by default, and I changed it to use the 
> "To:" section of the invite


Why that? It should route calls via R-URI not To header.


> 
> 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:"
> 

Is this a peering failover or a CF scenario ?



> *Examples:*
> ########################################## ORIGINAL 
> ###################################################
> 
> INVITE sip:01xxxxxxx9 at 2.2.2.2;user=phone SIP/2.0'
> Via: SIP/2.0/UDP 1.1.1.1:2049;branch=z9hG4bK-li82baliyyct;rport'
> From: "27xxxxxxxx2" <sip:test at 2.2.2.2>;tag=5xiqh2lj6c'
> To: <sip:01xxxxxxx9 at 2.2.2.2;user=phone>'
> Call-ID: 313536303935383439353536393239-davtmohvrxwe'
> CSeq: 1 INVITE'
> Max-Forwards: 70'
> User-Agent: '
> Contact: <sip:test at 1.1.1.1:2049;line=beoci8wp>;reg-id=1'
> X-Serialnumber: 0xxx1xxxxxx8'
> P-Key-Flags: keys="3"'
> Accept: application/sdp'
> Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, 
> PRACK, MESSAGE, INFO, UPDATE'
> Allow-Events: talk, hold, refer, call-info'
> Supported: timer, 100rel, replaces, from-change'
> Session-Expires: 3600'
> Min-SE: 90'
> Content-Type: application/sdp'
> Content-Length: 427'
> 
> ######################################### 1st Failover 
> ##################################################
> 
> INVITE sip:Tenant-Failover1 at ClientIP1:5060 SIP/2.0'
> Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bKka512a6U;rport'
> From: 
> <sip:27xxxxxxxx2 at exampledomain.com>;tag=4532BC27-5D0A53F4000EA445-83A95700'
> To: <sip:01xxxxxxx9 at exampledomain.com>'
> CSeq: 10 INVITE'
> Call-ID: 313536303935373933363238313932-lxxe06we83v8_b2b-1'
> Route: <sip:127.0.0.1:5060;lr;lr>'
> Max-Forwards: 70'
> Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, 
> MESSAGE, UPDATE'
> Supported: 100rel, replaces, from-change'
> P-NGCP-Caller-Info: 
> <sip:Tenant at exampledomain.com>;ip=127.0.0.1;port=5080;primary=27xxxxxxxx3'
> P-NGCP-Callee-Info: 
> <sip:Tenant-Failover1 at exampledomain.com>;ip=127.0.0.1;port=5060;primary=27xxxxxxxx4'
> P-D-Uri: sip:127.0.0.1:5060;lr'
> Content-Type: application/sdp'
> Contact: <sip:127.0.0.1:5080;transport=udp>'
> Content-Length: 455'
> 
> ######################################### 2nd Failover 
> ##################################################
> 
> INVITE sip:Tenant-Failover2 at ClientIP2:5060 SIP/2.0'
> Via: SIP/2.0/UDP 127.0.0.1:5080;branch=z9hG4bKka512a6U;rport'
> From: 
> <sip:27xxxxxxxx2 at exampledomain.com>;tag=4532BC27-5D0A53F4000EA445-83A95700'
> To: <sip:Tenant-Failover1 at exampledomain.com>'
> CSeq: 10 INVITE'
> Call-ID: 313536303936363439373236333036-f8f5lkiwc31n_b2b-1'
> Route: <sip:127.0.0.1:5060;lr;lr>'
> Max-Forwards: 70'
> Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, 
> MESSAGE, UPDATE'
> Supported: 100rel, replaces, from-change'
> P-NGCP-Caller-Info: 
> <sip:Tenant-Failover1 at exampledomain.com>;ip=127.0.0.1;port=5080;primary=27xxxxxxxx3'
> P-NGCP-Callee-Info: 
> <sip:Tenant-Failover2 at exampledomain.com>;ip=127.0.0.1;port=5060;primary=27xxxxxxxx4'
> P-D-Uri: sip:127.0.0.1:5060;lr'
> Content-Type: application/sdp'
> Contact: <sip:127.0.0.1:5080;transport=udp>'
> Content-Length: 455'
> 
> 
> As you can see in the 2nd Failover, the "To:" is the RURI of the 1st 
> failover.
> 
> 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.



You can try to pay with the preferences "outbound_to_user".



> 
> Kind Regards,
> 
> *Morne du Plessis*
> 
> Senior Network/Voice Engineer / Department Manager
> 
> *TenacIT***
> 
> *Strategic IT Consulting *•*Advanced Networking *•
> 
> *Custom Development *•*Hosting *•*Syspro Support *
> 
> Tel: 041 10 10 100
> 
> Web: http://www.tenacit.net <http://www.tenacit.net/>
> 
> PBefore printing this email please think about the environment
> 
> ------------------------------------------------------------------------
> 
> Date: 2018-10-22 12:02:13 PM
> Subject: Re: [Spce-user] Incident [#SRY04139] Carrying original 
> destination when forwarding
> From: apogrebennyk at sipwise.com
> To: Spce-user at lists.sipwise.com
> Cc: linksilent at app.tenacit.net
> 
> Hi,
> 
> On 10/18/2018 01:57 PM, linksilent at app.tenacit.net wrote:
>  > The part that I want to stay the same after the forward is the "To:"
>  > number before the @
> 
> as explained try setting the preference outbound_to_user to
> "Original (Forwarding) called user" or "Received To header" - on the
> call receiving side (user/domain).
> 
> This should help.
> Andrew
> 
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> 



Daniel Grotti

Head of Customer Support                     Sipwise GmbH
e: dgrotti at sipwise.com                     Europaring F15
t: +43(0)130120332                A-2345 Brunn Am Gebirge
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