How to integrate Asterisk 1.6 with Avaya Voice Portal

atul_sharma
Joined: Thu 19 of Feb, 2009

Re: How to integrate Asterisk 1.6 with Avaya Voice Portal

Posted:Wed 21 of Oct, 2009 (06:38 UTC)
With 1.6.1 TCP is default. I'm not sure why its switching to UDP. Can you provide a asterik trace.
cschebel
Joined: Wed 21 of Oct, 2009

Re: How to integrate Asterisk 1.6 with Avaya Voice Portal

Posted:Wed 21 of Oct, 2009 (01:59 UTC)
Hi Atul,

Do you know if this works with the later versions of Asterisk (1.6.1.6)?

We have tried it, an cannot get Asterisk to talk TCP to AVP.. it always uses UDP even though the transport for the AVP trunk is set to transport=tcp. And tcpenable=yes is set in sip.conf

In your post you mention 'Compile Asterisk with SIP-TCP'.. does this compilation option still existing in the newer versions of asterisk? I can't find it.

Regards,
Cal
nicheplrpackage
Joined: Wed 07 of Oct, 2009

Re: How to integrate Asterisk 1.6 with Avaya Voice Portal

Posted:Wed 07 of Oct, 2009 (10:13 UTC)
<a href="http://www.nicheplrpackages.com">Niche PLR</a>
atul_sharma
Joined: Thu 19 of Feb, 2009

How to integrate Asterisk 1.6 with Avaya Voice Portal

Posted:Sat 03 of Oct, 2009 (06:33 UTC)
Avaya voice portal 4.1 and above support SIP, but only over TCP. Atsterisk 1.6.0.9 now have a support for SIP over TCP, this feature can be used to integrate AVP with Asterisk.


Configuring the Asterisk

1. Compile Asterisk with SIP-TCP 

2. Add a sip trunk in  sip.conf as follows 

[avp]
type=peer
secret=1234	
host=172.16.15.117
port=5060
canrenvite=yes
insecure=very
qualify=yes
nat=yes
transport=tcp
context=incoming ; this section will be defined in extensions.conf

3. Create the Following entries in extension.conf

;AVP Call
[incoming]
exten => 6291,1,Verbose(ANI=${CALLERID(ANI)})
exten => 6291,n,Dial(sip/202011@avp,50,r)
exten => 6291,n,Hangup()

exten => 6292,1,Verbose(ANI=${CALLERID(ANI)})
exten => 6292,n,Dial(sip/202012@avp,50,r)
exten => 6292,n,Hangup()

exten => 6293,1,Verbose(ANI=${CALLERID(ANI)})
exten => 6293,n,Dial(sip/202013@avp,50,r)
exten => 6293,n,Hangup()

exten => 6294,1,Verbose(ANI=${CALLERID(ANI)})
exten => 6294,n,Dial(sip/202014@avp,50,r)
exten => 6294,n,Hangup()


Configuring AVAYA Voice Portal

1.	After Logging into Voice Portal click on VOIP Connections and Client SIP Tab

2.	Click Add button to create New connection

Name : Asterisk-sip
Proxy TransPort TCP
Proxy ServerAddress : 172.16.15.149 <Asterisk server IP)
Proxy Server Port : 5060

Click Continue button to proceed next

Listen Port : 5060
SIP Domain : *
Call capacity : 
Maximum Simultaneous Calls : 5 ( depends on your license )

All Calls can be either inbound or outbound

Leave rest of the configuration 

3.	Restart the MPP manager 


4. Click on the Port Distribution

It should show the status of the trunk Asterisk-sip as in_service.

Assign a VXML application to extension 202011-202014. Make call to asterisk extension 6291-6294, it should playback the application assigned to mapped AVP port

Thats it !!!