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

[Mario Contreras]

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!


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