SIP Trunk and DID numbers

[stevea2Z]

Hi guys,

Hopefully one of you can help me out. I´m setting up an IP Office R8 with a prionet SIP trunk. Outgoing calls are working fine, but incoming DID numbers are a pain. All DID numbers are coming on on 1 number (the username of the trunk) In the monitor trace i can see the DID number in the diversion header, but i´m unable to use this to route my calls to the destination.

Anyone has some pointers or some great iedeas, I´m running out

 

[Bas1234]

Add an extra SIP URI and put a * in the bellow options.

Local URI > * (instead of Use Authentication Name)
Contact > * (instead of Use Authentication Name)
Displayname > * (instead of Use Authentication Name)

incomming group id 0
outgoing group id 100 (or any free number)

Now you can add any number in the incomming call route.

___________________________________________
It works! Now if only I could remember what I did...

Dain Bramaged (Avaya Search tool

http://tinyurl.com/bas1234

)
______________________________________

 

[stevea2Z]

in the SIP URI I already have:
Local URI: *
Contact: *
Display Name: *
PAI: *
Registration: 1

Now all my DID numbers are coming in on 1 destination

For example: I dial with my phone to 00442034321044 the to header shows 0034931770511 instead of 00442034321044

monitor trace:
Warning: Logging to Screen Started **********
301101mS SIP Rx: UDP 85.119.188.3:5060 -> 192.168.100.100:5060
INVITE sip:0034931770511@192.168.100.100:5060;transport=udp SIP/2.0
Record-Route: <sip:85.119.188.3;lr=on;ftag=as5a2df96d>
Via: SIP/2.0/UDP 85.119.188.3;branch=z9hG4bKf0e1.27682fb4.0
Via: SIP/2.0/UDP 85.119.188.31:5060;received=85.119.188.31;branch=z9hG4bK26f73181;rport=5060
Max-Forwards: 69
From: "0032486798038" <sip:0032486798038@85.119.188.3>;tag=as5a2df96d
To: <sip:0034931770511@85.119.188.3:5060>
Contact: <sip:0032486798038@85.119.188.31>
Call-ID: 0775252638ba930e016fb43002262b1e@85.119.188.3
CSeq: 102 INVITE
User-Agent: Integrics Enswitch
Date: Thu, 08 Mar 2012 18:06:42 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Diversion: <sip:00442034321044@ast2>
Content-Type: application/sdp
Content-Length: 358
X-Enswitch-RURI: sip:0034931770511@85.119.188.3:5060
X-Enswitch-Source: 85.119.188.31:5060

 

[phoneguy610]

did you build the incoming call route?

 

[Bas1234]

Put PAI and registration to none

___________________________________________
It works! Now if only I could remember what I did...

Dain Bramaged (Avaya Search tool

http://tinyurl.com/bas1234

)
______________________________________

 

[amriddle01]

The provider needs to send the correct information, they need to do it proprly rather than you working around their errors. Do you have that number added to Incoming call route though?

 

[stevea2Z]

I have changed PAI to NONE and registration to NONE.
I have configured all DID numbers in the incoming call treatment.

I still have the same problem. In the monitor all calls are going To: <sip:0034931770511@85.119.188.3:5060>
instead of the DID number i´m calling.

In the trace i can find the DID in the diversion
Diversion: <sip:00442034321044@ast2>

But the TO header always is the same main number 0034931770511


New trace

256056mS SIP Rx: UDP 85.119.188.3:5060 -> 192.168.100.100:5060
INVITE sip:0034931770511@192.168.100.100:5060;transport=udp SIP/2.0
Record-Route: <sip:85.119.188.3;lr=on;ftag=as41ae689e>
Via: SIP/2.0/UDP 85.119.188.3;branch=z9hG4bKeb57.9b8e94c7.4
Via: SIP/2.0/UDP 85.119.188.31:5060;received=85.119.188.31;branch=z9hG4bK47f6f9d5;rport=5060
Max-Forwards: 69
From: "032486798038" <sip:032486798038@85.119.188.3>;tag=as41ae689e
To: <sip:0034931770511@85.119.188.3:5060>
Contact: <sip:032486798038@85.119.188.31>
Call-ID: 56ae76201c6b9b6b0ddcd39831861250@85.119.188.3
CSeq: 102 INVITE
User-Agent: Integrics Enswitch
Date: Thu, 08 Mar 2012 19:20:28 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Diversion: <sip:00442034321044@ast2>
Content-Type: application/sdp
Content-Length: 358
X-Enswitch-RURI: sip:0034931770511@85.119.188.3:5060
X-Enswitch-Source: 85.119.188.31:5060

 

[amriddle01]

You can't change what they are sending by configuring the system, they need to send the correct info, your URI needs to be * * * for when they fix it, so you may as well change it back

 

[Bas1234]

On the sip line, put the Send caller id to none.
Leave the rest default.

___________________________________________
It works! Now if only I could remember what I did...

Dain Bramaged (Avaya Search tool

http://tinyurl.com/bas1234

)
______________________________________

 

[amriddle01]

The IPO will only route incoming calls from the Request URI or the To Header

 

[stevea2Z]

I guess the only solution is to have a nice conversation with the sip provider