[Spce-user] user=phone and 181 forward missing

[Julian Seifert]

Hi Andrew,

well you actually pointed me in the right direction. I looked through the proxy.conf and found the
section for calls directed to "SIP Peering" :) (Route to PSTN)

The add_uri_param came in handy and we added 
        add_uri_param("user=phone"); 
just before: 
        route(ROUTE_OUTBOUND);

and now I see:
INVITE sip:095614049411 at lab.enviatel.de;user=phone SIP/2.0'

leaving our server - so it seeeems to have sufficed to add it at that point :) 


> If you have a Sipwise PRO customer you are welcome to open a ticket to
> expedite the resolution.

Well we are a sipwise pro customer and I'll probably still open a ticket
for that as we will have the same tests with our pro system in a couple of weeks.

> Do you have a link to specification that requires to add the
> 181-forwarding response? 

It's in the "Testspezifikation für NGN Interconnection Kompatibilitätstests " of AKNN("Arbeitskreis für technische und betriebliche Fragen der Nummerierung und der Netzzusammenschaltung")  in germany. It's a working group for all kinds of telecommunication
specifications and guidelines. 
 
http://www.aknn.de/fileadmin/uploads/oeffentlich/Testspezifikation_Interconnection_V_1-0-0.pdf

It's mentioned in test "SS_cfu_003" and it's checked for a 181. It's the same spec that requires us 
to provide the "user=phone" parameter.

kind regards,

  Julian

________________________________________
Von: Andrew Pogrebennyk [apogrebennyk at sipwise.com]
Gesendet: Freitag, 16. Oktober 2015 16:47
An: Julian Seifert; spce-user at lists.sipwise.com
Betreff: Re: [Spce-user] user=phone and 181 forward missing

Julian, see my reply below.

On 10/15/2015 09:30 PM, Julian Seifert wrote:
> we are running 4.0.1 as lab system and during some quite exhaustive tests we noticed the following.
> The first thing is in our invites the "user=phone" part is missing it appears at about 4 places in the kamailio.cfg
> of the loadbalancer but it apparently does not make it into any outgoing requests. Or is there any knob
> in the webinterface to configure that?
>
> It should look like this(taking it from kamailio-proxy.log from an incoming call from a peer)
> M=INVITE R=sip:+4995614049417;npdi;rn=+4995614049417 at lab.dacor.de;transport=udp;user=phone F=sip:+4934141400306 at lab.enviatel.de;user=phone
>
> but it looks like this (how we send it to our peer)
> M=INVITE R=sip:034141400309 at lab.dacor.de F=sip:095614049412 at lab.dacor.de

There is no knob to enable that. It depends on where you need to add
user=phone parameter, only on calls to the peer or on calls to
registered subscribers too. It should be relatively easy to implement it
for the former case in the proxy.cfg.customtt.tt2 file.
If you have a Sipwise PRO customer you are welcome to open a ticket to
expedite the resolution.

> The second is when you configure an unconditional forward for a subscriber there is no "181-forwarding" notice
> sent to the caller.
>
> Any ideas on either topic how to fix it to be the way we need it to be?

Do you have a link to specification that requires to add the
181-forwarding response? It was never requested by anybody yet, as far
as I remember. Again, it is easy to script it in proxy.cfg.customtt.tt2
file, but this can cause real interoperability issues (e.g. many
endpoints would ignore the real reply from the callee because to_tag !=
tag of generated 180 response). So, waiting for more information.

Andrew