How to Configure Routing Point to handle inbound calls through SIP trunk

[Ali]

Hello all!

Since I am not so much familiar with URS & IRD, I would like to know how to quickly configure a Routing Point ([b]RP[/b]) to handle inbound calls through a SIP trunk.

URS strategy is ready, which simply diverts all inbound/internal calls to an ACD queue -- it works! No problem here.

What I want is to associate this RP with a SIP trunk, so all inbound calls generated by my Asterisk can be routed by my strategy.

Could you please explain me how to do this step by step?

Thank you in advance.

[Tambo]

emm  just make up a route point under switch then load strategy onto it in IRD

[nonny]

Match the route point to the number asterisk passes or setup a dial plan.

Sent from my SM-N9005 using Tapatalk

[Ali]

Dial Plan for SIP Trunk?

[font=courier]Incoming call --> SIP Trunk --> (x) --> Routing Point[/font]

I created a new Trunk with the following Annex:

[TServer]
"Contact" = "<ip>:<port>"
"prefix" = "<DN of Routing Point>"
"record" = "false"
"refer-enabled" = "false"

In General tab, Register set to "true"

Well, should I create a dial-plan and add it into Annex/TServer section?

[cavagnaro]

Trunk is for Outgoing calls
Asterisk will send a number to Genesys, lets say 9000, then you create a RP 9000 and load a strategy on it
If you don't want 9000 configure your Asterisk properly

[Ali]

Cavagnaro,

Thank you for your reply.

Like I said, I created RP 8999 (tested. It works).

On Asterisk, in /etc/asterisk/sip.conf, the configuration is as follows:

1. register to 8999:8999@<genesys sip server IP>
2. created context

[size=10pt][font=courier][8999]
type = friend
callerid = genesys
fromuser = 8999
context = callcenter
nat = yes
qualify = yes
username = 8999
secred = 8999
insecure = invite,port
host = <genesys sip server IP>
dtmfmode = rfc2833
canreinvite = no
qualify = yes
disallow = all
allow = ulaw
allow = alaw
allow = gsm
usereqphone = yes
fromdomain = <genesys sip server IP>
transport = tcp[/font][/size]

So,

1. Asterisk cannot register to 8999 at <genesys sip server IP> ([i]will check if it is really necessary[/i])
2. When Asterisk places a call to 8999@<genesys sip server IP>, the call will be automatically diverted to this RP, right?

Thank you

[cavagnaro]

So you want to route all calls from 8xxx to a single Genesys RP?
On Genesys there is no way to do that, you will have to create a DID (8000-8999) range on your Asterisk and all those calls route to a single RP with a unique DID identifier.
How to do that is more an Asterisk config question.
I, personally have no clue

Whatever Asterisk sends to Genesys SIP Server, it must be a DN on Genesys, can be a RP or an extension. RP will execute the strategy, Extension will be like any agent set

[Ali]

I already created the dial-plan on Asterisk side.


What I want is to place a call to [font=courier]sip/8999@<IP of Genesys SIP Server>[/font], which I can. However, Genesys SIP Server returns me "[i]SIP 403 Forbidden[/i]" error ([size=8pt][color=blue]The Genesys SIP server understood the request, but is refusing to fulfil it[/color][/size]).

I think this is kind of codec-related issue. Will figure out.

I will ask if I have further question.

Thank you.

[Kubig]

I think, this is caused by Genesys SIP server by default, in case your GW/Asterisk is not configured as Trunk object within Genesys configuration. For Genesys it is unknown gateway, which are declined by default. So, do following:

[list]
[li]enable unknown gateway by SIP application option[/li][/list]
or
[list][li]create the trunk which will identify your GW by IP address/host  (better solution)[/li]
[/list] 

[Ali]

Thank you, Kubig.

Since I am not that familiar with IRD/URS, I have some more newbie questions regarding what you told:

1. Is "unknown gateway" in Annex section of the SIP Server?
2. Is this trunk for Asterisk, or for RP (if for RP, how to associate it?)

[Kubig]

[list type=decimal]
[li]It is an option "enable-unknown-gateway" under section "TServer" on SIP server application object[/li]
[li]It is a trunk, associated with Asterisk (IP/host, prefix, etc.)[/li]
[/list]

[Ali]

I defined enable-unknown-gateway (true) in Annex/TServer.

The message Genesys SIP Server returns is [color=red][font=courier]SIP/genesys-0123f9d0 is circuit-busy[/font][/color].

I will also configure the trunk for Asterisk

[Kubig]

Try to post logs covering the incoming call

[Ali]

Logs are

[color=green][font=courier][size=10pt]01:55:38.295: $+NET:SIP::0:0
01:55:38.296: SIPS:LOGBLOCK:BEGIN:SIPMSG:[
01:55:38.296: SIPTR: Received [648,TCP] 865 bytes from 192.168.2.34:41202 <<<<<
INVITE sip:8999@192.168.2.247 SIP/2.0
Via: SIP/2.0/TCP 192.168.2.34:5060;branch=z9hG4bK7f4b0e25;rport
Max-Forwards: 70
From: "3119" <sip:8999@192.168.2.247>;tag=as1d375566
To: <sip:8999@192.168.2.247>
Contact: <sip:8999@192.168.2.34:5060;transport=TCP>
Call-ID: 168c89514e2bfafe7a82966e0b971615@192.168.2.247
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.0.1
Date: Mon, 22 Aug 2016 22:18:51 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 310

v=0
o=root 1975435331 1975435331 IN IP4 192.168.2.34
s=Asterisk PBX 1.6.0.1
c=IN IP4 192.168.2.34
t=0 0
m=audio 18602 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

01:55:38.297: CallMatcher: no call match attributes found
01:55:38.297: call to itself is rejected
01:55:38.298: Sending  [648,TCP] 334 bytes to 192.168.2.34:41202 >>>>>
SIP/2.0 403 Forbidden
Via: SIP/2.0/TCP 192.168.2.34:5060;branch=z9hG4bK7f4b0e25;rport;received=192.168.2.34
From: "3119" <sip:8999@192.168.2.247>;tag=as1d375566
To: <sip:8999@192.168.2.247>;tag=303A7051-50FE-446C-8801-CFA78988A5A0-30
Call-ID: 168c89514e2bfafe7a82966e0b971615@192.168.2.247
CSeq: 102 INVITE
Content-Length: 0[/size][/font][/color]

Where [b]8999[/b] is Routing Point that diverts all incoming calls to the Agent Group.

[Kubig]

The From part of SIP INVITE message is incorrect:

From: "3119" <sip:8999@192.168.2.247>;tag=as1d375566

You cannot call to itself, the from address should be address of asterisk, not of Genesys.

From my point of view, the configuration on Asterisk level is not correct, the RP should not be configured as Extension/User. Follow the existing document which describes how properly deploy and integrate the Asterisk with Genesys SIP.