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

Ali · August 22, 2016, 9:33am

Hello all!

Since I am not so much familiar with URS & IRD, I would like to know how to quickly configure a Routing Point (RP) 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 · August 22, 2016, 9:57am

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

vaunmac · August 22, 2016, 10:10am

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

Sent from my SM-N9005 using Tapatalk

Ali · August 22, 2016, 10:47am

Dial Plan for SIP Trunk?

Incoming call → SIP Trunk → (x) → Routing Point

I created a new Trunk with the following Annex:

[TServer]
“Contact” = “:”
“prefix” = “”
“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 · August 22, 2016, 11:32am

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 · August 22, 2016, 1:17pm

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:

register to 8999:8999@
created context

[8999]
type = friend
callerid = genesys
fromuser = 8999
context = callcenter
nat = yes
qualify = yes
username = 8999
secred = 8999
insecure = invite,port
host =
dtmfmode = rfc2833
canreinvite = no
qualify = yes
disallow = all
allow = ulaw
allow = alaw
allow = gsm
usereqphone = yes
fromdomain =
transport = tcp

So,

Asterisk cannot register to 8999 at (will check if it is really necessary)
When Asterisk places a call to 8999@, the call will be automatically diverted to this RP, right?

Thank you

cavagnaro · August 22, 2016, 2:09pm

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 · August 23, 2016, 6:45am

I already created the dial-plan on Asterisk side.

What I want is to place a call to sip/8999@, which I can. However, Genesys SIP Server returns me “SIP 403 Forbidden” error (The Genesys SIP server understood the request, but is refusing to fulfil it).

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

I will ask if I have further question.

Thank you.

Kubig · August 23, 2016, 7:47am

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:

enable unknown gateway by SIP application option

or

create the trunk which will identify your GW by IP address/host (better solution)

Ali · August 23, 2016, 8:28am

Thank you, Kubig.

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

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

Kubig · August 23, 2016, 8:51am

[ol]- It is an option “enable-unknown-gateway” under section “TServer” on SIP server application object

It is a trunk, associated with Asterisk (IP/host, prefix, etc.)[/ol]

Ali · August 23, 2016, 9:00am

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

The message Genesys SIP Server returns is SIP/genesys-0123f9d0 is circuit-busy.

I will also configure the trunk for Asterisk

Kubig · August 23, 2016, 9:38am

Try to post logs covering the incoming call

Ali · August 23, 2016, 9:57am

Logs are

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

Where 8999 is Routing Point that diverts all incoming calls to the Agent Group.

Kubig · August 23, 2016, 11:12am

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.

Ali · August 23, 2016, 12:13pm

Can you please give the URL of this document?

Kubig · August 23, 2016, 12:19pm

You can follow for example this http://www.voip-info.org/wiki/view/Asterisk+Genesys+SIP+Server+Integration

Ali · August 23, 2016, 2:27pm

Solved!

Thank you all for your support.

Configuration is as follows:

On Genesys side;

SIP Server Application

Annex/TServer/enable-unknown-gateway=true

Trunk

Created trunk for Asterisk, where Annex/TServer is

“contact” = “:5060”
“prefix” = “”
“record” = “false”
“refer-enabled” = “false”

Created trunk for Routing Point (RP) with following AnnexTServer:

“contact” = “:5060”
“prefix” = “”
“record” = “false”
“refer-enabled” = “false”

On Asterisk side;

sip.conf

[genesys]
type = friend
callerid = genesys
context = callcenter
nat = yes
qualify = yes
insecure = invite,port
host =
dtmfmode = rfc2833
canreinvite = no
qualify = yes
disallow = all
allow = ulaw
allow = alaw
allow = gsm
usereqphone = yes
transport = tcp

extensions.conf

[incoming]
include = default
[callcenter]
include = incoming
include = default
exten = _00.,1,NoOP(“Outbound Call”)
exten = _00.,2,Dial(SIP/${EXTEN:2}@genesys)

Using your soft phone application, which already registered to the Asterisk server, when you dial 00, your call goes to RP over Genesys SIP Server.

P.S. You need to reload SIP and Extension settings in Asterisk after having sip.conf and extensions.conf modified.

Kubig · August 23, 2016, 4:28pm

There is no need for having trunk for RP