[Spce-user] Problems with my sip peering since i upgraded

[Mario Contreras]

Ok, I have seen what happend. I think my provider need 100rel in 
supported header.

I've created 
/etc/ngcp-config/templates/etc/sems/etc/ngcp.sbcprofile.conf.customtt.tt2 and 
add Require to header_list like you say in other email, and it's working 
now.

I'm curious, has it been a change in spce? Another thing. I have 
realized I hadn't troubles with asterisk... why??

Thanks again!

El 10/10/2012 14:24, Mario Contreras escribió:
> Hi all,
>
> I'm having some issues since I have upgraded today to 2.6. I expect I
> can explain correctly:
>
> My peer has the ip 1.1.1.1, and my domain is sipdomain.com.
>
> Yesterday, when my sip provider made an invite the field 'to:' was like
> this:
> INVITE sip:7870#922333333 at 1.1.1.1:5060 SIP/2.0
> To: <sip:922333333 at mydomain.com:5060>
>
> I had to add 7870# because my provider ask me for it with a rewrite rule.
>
> But, since I upgraded the invites are like this:
>
> INVITE sip:7870#922333333 at 1.1.1.1:5060;transport=udp SIP/2.0
> To: <sip:7870#922333333 at 1.1.1.1>
>
> Can it originate troubles? The calls are terminating with a 504 server
> timeout like you can see in the image. I know the peering must answer
> with a ringing signal and after that with a 200 ok and it doesn't. In
> fact, the call starts(with wireshark I can hear the tone and voices),
> but my ce sees it like failed.
>
> I don't now if it is related. Anyway, if there is rtp traffic sipwise
> shouldn't see the call like failed/missed, right? Can you help me? Thanks!
>
>