"Failed to get registration info" for SIPCS

Voice & Telephony
1

Hi, all,
has anyone seen the following error in their T-Server logs?

09:02:48.708 +++ CIFace::Request +++
– new invoke
Parsed: RequestMakeCall
From: CTISIP[392]/5
Numbers: +<6000> -<2701>
Status: parsed:1 queued:0 sent:0 acked:0 preevent:0 event:0 context:0 transferred:0

– validate
– state check: ok
SIPTS_IFace::QServe()01BDDE80
CIFace: Sent CRequest@01BDDE80 RequestMakeCall-CTISIP[392]/5
sipcs: Number:2701 did not match any configured or registered internal DNs
sipcs: Number:2701 did not find any gateway to send this call to
sipcs: SIPTS_Request::MakeCall:ERROR - Failed to get registration info for 6000

We are using SipTserver and there is a VoIP gateway and SIP phones and DN 6000 and everything works. Help please :-[

Vic

2

Does the annex tab of the extension DN in cme have a TServer section ?

section TServer

subsection contact

value ip address:port example 192.168.22.88:5061

3

Oops. You were right. There was no registration there. Good catch :slight_smile:
I am wondering - do you need to register it for all DNs that are going to be SIP? If you think of it, this is actually silly, because SIP client provides DN as URI and TServer should be able to see it IP address as well, right? But, then, I am sure there is a reason why they are doing it and I am not here to question almighty Genesys about it. Not yet, at least :>

Thank you once again!
Best regards,
Vic

4 Likes
4

Ok, I am confused.

When you use SIP TServer and you want to make a call, does a normal TDial do the trick?

DesktopToolkitX.TStatus rc=this.TLine.TDial(this.txtANI.Text.Trim());

Or, do I still have to try to create a new session with SIP? SIP TServer would not create a channel for me, right?

Commenting as I go along…

5

not sure. When I used it I had a GAD desktop and most of the dialing seemed normal except for the fact that I had to create alot of routing rules in the gateway. Since then a sip add on has been added to GAD.

6

I was able to get Dialing to work.
This involved the fact that I have to:

  1. Tell Genesys I am dialing via CTI
  2. Actually DIAL via SIP

Did the trick.

But, now stuck on the new one - I cannot SIP T-Server to connect to external VoIP gateway, which requires authentication.

In the log, you see INVITE but no authentication of any sort:

18:43:14.349 — CIFace::Event —
sipcs: party [6001@01b88150:0] +dlg [4773@199cea0]
sipcs: dialog [4773:01@0199cea0] : -party [0x00000000] +party [0x01b88150]
sipcs: party [99117@01b8fcd0:0] +dlg [0@182e580]
sipcs: dialog [0:00@0182e580] : -party [0x00000000] +party [0x01b8fcd0]
sipcs: dialog [4773:01@0199cea0]: << Event 12 << TRN[4848]
sipcs: 18:43:14.349 Stored This SDP [4773]
sipcs: Number:99117 did not match any configured or registered internal DNs
sipcs: Selected Service: victor its priority 0
sipcs: GetLongestPrefixMatch: selected prefix for number 99117 is 9, gateway: victor
sipcs: Number:99117 did not match any configured or registered internal DNs
sipcs: Selected Service: victor its priority 0
sipcs: GetLongestPrefixMatch: selected prefix for number 99117 is 9, gateway: victor
gsip:DLG[4774]: INVITE TD = TRN[4849]

sipcs: 18:43:14.349 Sending [368,UDP] 1149 bytes to 172.30.0.1:5060 >>>>>
INVITE sip:99117@172.30.0.1:5060 SIP/2.0
From: "ALEGRIA-6001"sip:6001@172.30.0.222;tag=b85ce62a
To: "99117"sip:99117@172.30.0.222
Call-ID: 9AC4BA9E-98C8-405F-8D47-4D357E44CE34-2396@172.30.0.222
CSeq: 1 INVITE
Content-Length: 377
Content-Type: application/sdp
Via: SIP/2.0/UDP 172.30.0.222:5060;branch=z9hG4bK2D817229-83F5-4F07-90A4-02C69D725287-2427
Contact: sip:172.30.0.222:5060
Max-Forwards: 69
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
User-Agent: X-Lite release 1003l stamp 30942
Call-Info: http://example.com; a81d713f42590a63ZjBmY2QyYmRlNThmNGFlOGQxZGM1ZGVjODc3YWUyNDY.;gen-rt=b85ce62a;gen-lt=5F1DA793-EBBA-430B-A517-0E1DCAA67A2B-2405
Session-Expires: 1800;refresher=uac
Min-SE: 90
Supported: timer

I specifically created a gateway entry in CME by registering the “dial-out” digit as a trunk and then adding TServer option as shown in the screenshot.
Has anyone being able to get SIP Tserver to authenticate with VoIP gateways?
I tried sip:name@gateway: username=name;password=pwd for contact, but it did not work either.

Any ideas?
Vic

7

Hi Vic,

On a side note, which voice gateway are you using? and which configuration? did you need a softswitch or any other products in the mix to get the components working.

Ian