Genesys Integration


#1

Hi All,

I hope some-one can help. I’m trying to integrate my Asterisk box with a Genesys suite. I have a SIP trunk provided by sipgate.co.uk which works perfectly with Asterisk on it’s own. If i point the dialplan to just ring the extension, my 3CX softphone rings and two way voice is as clear as a bell. However, I cannot for the life of me get it to integrate with Genesys. Here is my set-up:

Genesys
Trunk DN - ‘asterisk’ | TServer > contact = sip:192.168.1.67:5060
> subscribe-presence-from = <sip gsip@genesys1:5060 (genesys1 being my Sip Server host)

Extension DN - 6001 | TServer > contact = 192.168.1.67:5060
> subscribe-presence = asterisk

Asterisk
sip.conf
[gsip]
type = friend
username = smithl
host = 192.168.1.86
port=5065
context = default
canreinvite = no

[6001]
type = friend
username = smithr
host = dynamic
regexten = 6001
context = default
notifyringing = yes
canreinvite = yes
nat = yes
qualify = yes

extensions.conf
[DID_trunk_1]
exten=>123456,1,Dial(SIP/${6001}@gsip,30,r)
exten=>123456,2,Hangup()

I can see activity in the Genesys SIP Server log, but the call disconnects straight away and i get no activity on GAD at all.

I’m not sure SIP Server even registers with Asterisk on startup.

Any help would be much appreciated. :confused:


#2

Hello??

Is this the right forum? Maybe there is a newbie forum i should be using?

:blush:


#3

You haven’t reached the minimum time to bump yet.

This is the right forum, but there are only about a dozen regulars and your configuration is sufficiently unusual that there is a good chance no-one has encountered it.


#4

Ah ok, thanks for the reply.

I’m pretty sure this will be a simple fix for some-one experienced with SIP integration.

I’v e spent a couple of days now reading docs and whitepapers. Hopefully, some-one will be ale to help.

Cheers


#5

Hi

[quote]
I can see activity in the Genesys SIP Server log, but the call disconnects straight away and i get no activity on GAD at all.[/quote]

But we cant, perhaps some debug information etc!!

Ian


#6

Yes, this is the final SIP messaging in SIP Server:

01:02:36.924 SIPS:LOGBLOCK:BEGIN:SIPDATA:[
01:02:36.924 Received [404,UDP] 382 bytes from 192.168.1.67:5060 <<<<<
ACK sip:192.168.1.86:5065 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.67:5060;branch=z9hG4bK76272541;rport
Max-Forwards: 70
From: “12345678” sip:12345678@192.168.1.67;tag=as66d3a3b4
To: sip:192.168.1.86:5065
Contact: sip:12345678@192.168.1.67
Call-ID: 4bf3960b18439c75646ebfa70262d529@192.168.1.67
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.2.11
Content-Length: 0

01:02:36.924 SIPS:LOGBLOCK:END:SIPDATA:]

But this was really just to prove the network, as I assume the above would never be routed, as GAD is not a SIP endpoint.

When i fire the call to the endpoint attached to the Asterisk (which should be monitored by SIP Server) i get a successful call, but no messaging in SIP Server.

i.e. it appears, that SIP Server is not subscribing to and is not receiving notifications from Asterisk. I’m sure this is something simple in my config.

I can attach the genesys output from start-up, but it just looks ok, except the SUBSCRIBE messages are not present.

grrr!


#7

Youare getting a 404
01:02:36.924 Received [404,UDP]

so the extensions doesn’t exist in the context that the call lands.

Ian