Re: Cisco IVR

worldcalltel · ‎01-08-2007

hi, i am using a AS5300 and currently exploring the IVR scripts. Here is my current config

aaa new-model

!

aaa authentication login h323 group radius

aaa authorization exec h323 group radius

aaa accounting connection h323 start-stop group radius

aaa session-id common

gw-accounting aaa

method voip

attribute acct-session-id overloaded

attribute h323-remote-id resolved

interface Ethernet0

description WAN-connection Duplex-not-support on E0

ip address 192.168.10.240 255.255.255.0 h323-gateway voip interface

h323-gateway voip h323-id ivr-testing

radius-server host 172.16.16.25 auth-port 1812 acct-port 1813

radius-server key xxx

radius-server vsa send accounting

radius-server vsa send authentication

call application voice ivr-testing tftp://172.16.16.30/TCLware/clid_col_npw_3_cli.1.1.0.tcl

call application voice ivr-testing language 0 en

call application voice ivr-testing set-location en 0 tftp://172.16.16.30/TCLware/prompts/en

dial-peer voice 1083 voip

description IVR Testing

application ivr-testing

destination-pattern 02070788133

session protocol sipv2

session target ipv4:172.16.16.30 dtmf-relay rtp-nte

codec g711ulaw

!

dial-peer voice 1082 pots

description IVR Testing - Do not Remove application ivr-testing

destination-pattern 44070788133

port 2:D

My queries about this is that, when i dial the DID using Cisco 7960 ip phone my attempts wont able to reach the router. Hence using an analog phone got an error no route to destination. Using my analog phone, it is connected to another gateway equipment. and my gateway is pointed to the cisco as 5300. I need some expert advise. Thanks

goranselthofer · ‎01-09-2007

Hi,

1. Please explain what is your idea? give us topology view.

2. Are you trying to place IVR for PSTN callers when they dial in on AS5350 via POTS?

where is your 7960 in topology?

3. also, when you try with analog phone, you are coming from voip side to as5350 and by this configuration as5350 should route that call to voip peer (ivr)?

4. POTS is configured with dest-patt... if you are waitig for inbound calls on that port, you should configure incoming called-number and it should be the same as IVR number

please let us know if this helps.

thx,

selt

worldcalltel · ‎01-09-2007

1. Please explain what is your idea? give us topology view.

1. Cisco AS5300 is in London.

2. My radius server (Voice Master) is located in NY

3. I have another Quintum A800 gateway in Philippines.

4. TFTP server is also located in London.

2. Are you trying to place IVR for PSTN callers when they dial in on AS5350 via POTS? = Yes

Obective:

1. Implement IVR on AS5300. Using an analog phone which is connected to Quintum. If i want to dial the DID 442070788133, i should be able to hear the IVR prompt. Then goin to Radius for AAA. Just like that for now...

goranselthofer · ‎01-10-2007

wait, wait...

Your objective than gives negative answer to second question because if you have analog phone connected to Quintum that call still goes over IP and comes on VOIP dial-peer to AS5300, not via PSTN and POTS dial-peer?

I am not sure if you can accept voip call and forward it to IVR because both calls are on voip side...

As I understand, IVR is not a problem in your case, call routing is?

i would suggest that first you try with simple call scenario without IVR just to be sure that calls are coming to AS5350.

Still, I am sending you a document where yoo can find more information about configuring IVR.

Cheers!

worldcalltel · ‎01-13-2007

another question. If i understand it correctly based on the document that you have sent. Is that on the dial peer there is a command parameters "application , when the call matches the dial peer it will trigger the application and that application will call from TFTP server then to the memory. After that the AAA from radius server. My problem was although my dial peer was now working, but it didn't reach the IVR at all. so what's my next step on this?Is there any problem on my radius or in my configurations on radius. Take note that i am using a voice master as my Radius and AAA. An immediate response would be much appreciated. Thanks in advance.

goranselthofer · ‎01-14-2007

well, I am sorry but i am no expert in raduis... so, i suppose some other members can evaluate this configuration regarding it...

still, i would suggest you to upload tcl script on the router (flash) and also in memory (see those docs)... just to be on the safe side...

you should try first with some basic script and no radius configuration... just to see if your call would go trough...

then, if everything is ok, you can go to the next steps... tftp, radius, aaa ...

there is no use of troubleshooting those levels if we are not sure that those basics are working well...

teodorgeorgiev · ‎01-16-2007

The TCL script that is loaded via the application command on the dial-peer is responsible for handling authentication/authorization.

If you open the .TCL script with a simple text editor you will see the "aaa" commands -->

* "aaa authenticate"

* "aaa authorize" and etc

So, the problem is that for some reason it can not interact with the Radius server. The script uses the configured Radius server settings at the Cisco gateway.

Btw, leave this Sysmaster/VoiceMaster. You will have thousands of problems with it.