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Re: SIP Trunking OXE. Can not route incomming call

Posted: 03 Apr 2014 15:48
by pantisimus
Well, I have checked everything..., but always diverted to Extension 100...

I was talking with specialist from SIP Provider Company. Told me that in "INVITE" message they put the "main" number 381117159100. For DID, they put other called numbers in "To" field.

PBX should read it from "To" field. Can OXE read from "To" field? In Version 9.1?

Re: SIP Trunking OXE. Can not route incomming call

Posted: 12 Apr 2014 14:41
by tgn
Put a CUBE or NeoGate (BorderSessionController) between yout provider and the oxe. Some times ago, i've used an asterisk to do such job for me (but just for testing purposes).

regards...

Re: SIP Trunking OXE. Can not route incomming call

Posted: 12 Apr 2014 14:41
by tgn
sry i mean SBC (Session Border Controller)

Re: SIP Trunking OXE. Can not route incomming call

Posted: 13 Apr 2014 20:32
by joao.carlos
Try using the last four digits of called number, and check on second page of TG to use only the last four

Re: SIP Trunking OXE. Can not route incomming call

Posted: 14 Apr 2014 08:51
by pantisimus
I have tried to change in Trunk Group number of digits unused to 9 (381117159101) in order not to use DDI.
But again, in OXE trace, I have "To" field 381117159101 and call is comming to extension 100.
Contact field is again 381117159100@provider.com. INVITE field is 381117159100.

THANK YOU!

Re: SIP Trunking OXE. Can not route incomming call

Posted: 14 Apr 2014 09:05
by haroun
a ticket with alcatel support ?! lot of epxerts havenot replayed to your topics so

Re: SIP Trunking OXE. Can not route incomming call

Posted: 14 Apr 2014 09:40
by haroun
really interessting case , hope you can solve it

Re: SIP Trunking OXE. Can not route incomming call

Posted: 15 Apr 2014 03:59
by tgn
the use of the To-Header is "dirty"

RFC 3261 ยง8.2.2.1 - "To and Request-URI":

The To header field identifies the original recipient of the request
designated by the user identified in the From field. The original
recipient may or may not be the UAS processing the request, due to
call forwarding or other proxy operations.

regards...

Re: SIP Trunking OXE. Can not route incomming call

Posted: 15 Apr 2014 05:11
by pantisimus
But the "To"header is the only place where I can see the destination number.

In Invite field, there is the main number that is also used for registration.
My all calls are forwarded to the number that is in the "INVITE" field.

Re: SIP Trunking OXE. Can not route incomming call

Posted: 15 Apr 2014 08:59
by haroun
exactly but if you have remarked in invite message user=name
can you change Passerted id for ongoing as a name alice@provider.com not as phone number and try

second test also try with deleting100 FROM DID and see what hapening