11.1.2.2 Caller ID Name not showing on outbound SIP trunk calls.

[mhoxie]

IPO 11.1.2.2

Control unit is part of server edition.

Calls come into the Control Unit from PSTN on a PRI from carrier.

Calls to one number on the PRI are being forwarded to a dispatch SIP trunk.

Incoming number is 212-555-4000. Incoming call route points to "4000". 4000 is a short code pushing 4000 to the SIP line. SIP line is line 100 and set to Auto.

The SIP Gateway gets the call just fine. Audio is good. Calls is good. The SIP side gets CID, but no CID name.

I also tried sending calls to 212-555-4000 to an extension, and then have that extension Twinned to 4000. Same issue.

The call on the PRI shows CID and Caller ID Name (CNAM), but the IPO's INVITE to the SIP shows this. (numbers changed for privacy)

SIP Tx: UDP 192.168.43.1:5060 -> 192.168.43.2:5060
INVITE sip:4000@192.168.43.2 SIP/2.0
Via: SIP/2.0/UDP 192.168.43.1:5060;rport;branch=z9hG4bK476665c7ee82bb5e0e3b33c444ec04fd
From: "92125551234" <sip:92125551234@192.168.43.2>;tag=447f8f12564b7c4b
To: <sip:4000@192.168.43.2>
Call-ID: 16bd2472e187c304541709f69357eb00
CSeq: 2124389404 INVITE
Contact: "92125551234" <sip:92125551234@192.168.43.1:5060;transport=udp>
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,REFER,NOTIFY,UPDATE
Supported: timer,uui
User-Agent: IP Office 11.1.2.2.0 build 20
P-Asserted-Identity: "92125551234" <sip:99737665691@192.168.43.1:5060>
P-Preferred-Identity: "92125551234" <sip:99737665691@192.168.43.1:5060>
Remote-Party-Id: "92125551234" <sip:92125551234@192.168.43.1:5060>;screen=yes
Content-Type: application/sdp
Content-Length: 275
User-to-User: 04;encoding=hex;purpose=isdn-uui;content=isdn-uui

v=0
o=UserA 1073272047 2041348526 IN IP4 192.168.43.1
s=Session SDP
c=IN IP4 192.168.43.1
t=0 0
m=audio 46750 RTP/AVP 9 0 18 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

What we should be seeing is:

SIP Tx: UDP 192.168.43.1:5060 -> 192.168.43.2:5060
INVITE sip:4000@192.168.43.2 SIP/2.0
Via: SIP/2.0/UDP 192.168.43.1:5060;rport;branch=z9hG4bK476665c7ee82bb5e0e3b33c444ec04fd
From: "JOHN SMITH" <sip:92125551234@192.168.43.2>;tag=447f8f12564b7c4b
To: <sip:4000@192.168.43.2>

I dont know why the IPO is not inserting the CNAM. I can clearly see "JOHN SMTITH" when I watch the call come in on the PRI in System Status.

 

[IPOTS]

What does it show when the SIP line has P-Asserted, P-Preferred checked and the forwarding/twinning set as original caller?

 

[mhoxie]

With or without all these checked, the invite to the SIP trunk always shows FROM "number" sip:number. Instead of "NAME" sip:number.

 

[derfloh]

Set local URI "Display" to "internal data"

IP Office remote service
IP Office certificate check
CLI based call blocking
SCN fallback over PSTN

 

[mhoxie]

Internal didnt work either. It's like the IPO is refusing to "see" the Caller ID Name from the PRI call.

 

[Westi]

shortcodes don't have a name so I guess your shortcode 4000 could have the name added via the TN field 2125551234s212555400z"John Smith" to put the full information in.

You could try and dial the shortcode to see if you get name information then once you have that working go the next step and dial in the DID on the PRI

Joe
FHandw, ACSS, ACIS

"Dew knot truss yore Spell Cheque

 

[mhoxie]

The problem is it needs to show the name of the caller calling on the PRI, putting a static name in wouldn't help. The client is a Police Station and when they get a non emergency call to the Avaya PRI, they want it to come to their dispatch via SIP. This works. It's just the CID Name doesnt pass from the PRI to the Invite sent to the SIP trunk.

 

[IPOTS]

The SIP provider would be responsible for sending the name. I have SIP trunk on my office system and I receive the name.
The IP Office doesn't have the ability to adjust the name on an incoming CLID.

 

[mhoxie]

IPOTS, this is not an incoming SIP call. We are taking an incoming call from a PRI, then forwarding out to SIP. The Invite to the SIP from the IPO is missing the CID NAME. If you forward a call from a SIP trunk to another SIP trunk, the CID and CID NAME are pushed out the forwarded SIP trunk. This does not appear to be the case with PRI forwarding to SIP.

 

[Westi]

I don't think that the passthrough works for names, only number.

Try sending the call in the PRI and out the PRI again to a cell phone to see if the name shows up.

If it is not showing the name then you could do some fancy voicemail pro programming and take the name and then input it again into the outgoing call. Not sure if that actually works but that is the only way I can see this working.

Joe
FHandw, ACSS, ACIS

"Dew knot truss yore Spell Cheque

 

[mhoxie]

Sending the PRI call Incoming callroute to a VMPro Module that transferred with $CLI_NAME in Source of Transfer worked.... but get this... for only the first call. I tested calling the 212-555-4000 number and they got my CID and CNAM. While on the call, I called from another cell phone and nope, only from number. How absurd is that? I look at the INVITES generated by the IPO and the first has a NAME in the From Field and the 2nd call has just the number. If I hang up both calls, and try again, my next call has NAME in the invite to the SIP... freakin Avaya man.

 

[sizbut]

Impressive, weird and interesting.

My suspicion there would be some bug with the $CLI_NAME variable already being seen as in use and so not available until the call that used it is over. That shouldn't be so but I did say bug. (Also trying 1st call, 2nd call, then drop 1st call and seeing what happens with 3rd call might give some clue as to what the voicemail server is doing.)

I would add a extra step in the callflow, using a Generic action to save $CLI_NAME as $CP1. Then use $CP1 in the transfer action.




Stuck in a never ending cycle of file copying.

 

[Westi]

mhoxie
your system is just simply jinxed

Have you put in the $CLI_NAME into Source and Description and do you put that in directly or via a CPx variable
if you did it one way try the other one.



Joe
FHandw, ACSS, ACIS

"Dew knot truss yore Spell Cheque