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Re: SIP Trunking OXE. Can not route incomming call

Posted: 15 Apr 2014 09:50
by haroun
readagain your two trace for 100 & 101
FOR 100 we have
(074009:000007) +--------------------------
----------------------------------+
(074009:000008) | Message received SIP ----> UA (neqt : 992)
(074009:000009) | INVITE : 381117159100@10.1.1.21:5060 ; user=name
(074009:000010) | From : <> 0648109051@10.0.0.2:5060 ; user=phone
(074009:000011) | To : <"TERI Jurija Gagarina"> @:0 ; user=phone
(074009:000012) +------------------------------------------------------------+
(074009:000013) |

for 101 We got
Message received SIP ----> UA (neqt : 992)
(074257:000324) | INVITE : 381117159100@10.1.1.21:5060 ; user=name
(074257:000325) | From : <> 0648109051@10.0.0.2:5060 ; user=phone
(074257:000326) | To : <"381117159101 381117159101"> 381117159101@ims.telekomsrbija.com:5060 ; user=phone

Re: SIP Trunking OXE. Can not route incomming call

Posted: 15 Apr 2014 11:14
by pantisimus
"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"

I see that for "invite" field" user=name and in "To" field user=number.
This is something I get from provider.
Can I manage something in OXE? Or you suggest to me to ask provider for modification?

PS:I tried to delete 100 from DID, but then when calling 101 call is rejected.

THANK YOU A LOT for effort!

Re: SIP Trunking OXE. Can not route incomming call

Posted: 24 Apr 2014 11:31
by haroun
hi any updates about the case ?

Re: SIP Trunking OXE. Can not route incomming call

Posted: 25 Apr 2014 08:23
by pantisimus
No, currently I had to switch to another project.

Tried few things.

I have noticed interesting thing.

When extension 100 is active and incomming call is for extension 101 (called number number 381117159101), on IP Touch screen of extension 100(that is ringing) is written 100 - 381117159101. Forwarding from 100 to 31117159101. But I can not define extension number 381117159101 in OXE in any way. Translations are not working inside OXE.

Re: SIP Trunking OXE. Can not route incomming call - Resolve

Posted: 10 Jun 2014 06:39
by pantisimus
Well..., the problem is in OXE.

For incoming calls - DID, OXE is looking at the INVITE field.
But, Telecom Provider is sending the main number in the INVITE field, and the extension number required in the TO field.

So, all calls are coming to the extension number defined in the INVITE Field that is always the same, but the called number is in the TO field.

Telecom provider did a favor to us. They have changed configuration in the way that each called number is in the INVITE filed. Not only in the TO field.

And the incoming call rings on the extension that is required.

Can we tell OXE to look at the TO field???

Re: SIP Trunking OXE. Can not route incomming call

Posted: 10 Jun 2014 07:33
by haroun
invite specified that user=name , and To is with user=phone , so oxe should use TO field because it designs the phone number stronge , may be on release 11 it's possible i'm not sure .
see with Alu support