SIP Ext GW problem

[MartinB]

Yes I did both ways. Via ippstat menu/option 15

and with the command ippstat telnet d XXX t 1440

but neither work...so cannot enable mirror

However I managed to get outbound working fine. Incoming still has 1 way speech but working on it now

Outbound problem was:

I had: Outbound Calls 100 REL + Supported on my external Gateway as that was a request from them in original spec document.

So when call is established I have following happening:
Invite
100 trying
183 session description from them
PRACK from me
401 from them
183 from them
183 from them
183 from them
Cancel from me
200 ok from them
487 request terminated from them
ACK from me

Problem was in their 183 Session progress they send "Require: 100rel

The PABX then responds with a PRACK (As they asked for it in the Progress message)

Then they send 401 Unauthorized, and keep on sending 183 messages after that but pabx does not respond witk ACK as it was not authorized and then the call cuts.

So I disabled the PRACK feature on SIP trunk and call works fine.

(This morning when I checked new traces I saw they removed the Required: Rel100 request from their Session Progress in the night. )

Problem was on their side as OXE reacted exactly the way they requested but then they send 401.

[cavagnaro]

Nice work

Enviado de meu E6633 usando o Tapatalk

[MartinB]

Now I still have a problem with inbound calls.
My SDP is correct and on second invite I send the IP address of the phone which will handle the RTP but they keep connection to the GD only as that was the first IP they got invite from
Ringing: 192.168.230.233 (GD/GA)
OK: 192.168.230.233 (GD/GA)
Re-Invite: 192.168.230.92 (Phone)

I tried changing SDP settings on my SIP ext Gw parametres but it does not help...
Awaiting feedback from them....

[tgn]

what sdp parameters do you mean.
have you tried „sdp in 18x == false“. it will switch early media to false and the 1st sdp will come in 200 ok with ip address of the ip phone. but provider have to support this...

regards...

[MartinB]

tgn

Problem was their invites did not include the FQDN of the 5 different sip accounts. Instead they have 1 IP address and because the OXE does not recognize the invite on incoming calls the calls come in on Internal sip gateway. Disabled sdp in 18x on internal sip gateway and speech was fine but problem was then that the moment a call gets transferred it gets cut.

So after massive fight and them not able to send me FQDN per sip trunk on their invites and only the SAME IP address I told them the pabx cannot distinguish which username and password they require per every invite from the pabx if they use the same IP address.

Now I set up 1 outbound sip trunk with all relevant info for outbound (FQDN, username, password etc) and a separate inbound SIP gateway where I have their IP address as my remote domain with username and password which they need in my invites. The call then comes in on correct SIP gateway with correct username and password and relevant settings.

They admitted today that they cannot have 5 sip accounts running from the same IP address to my pabx as they cannot supply me with FQDN in their invites as they only use the IP address.

So all problems solved...but it was a a big struggle and fight to make them see their mistake