SRST + Unity Express Problem

Answered Question
Apr 23rd, 2008

I've got a bit of a head scratcher (for me anyway).

Setup is as follows. Centralized CM5 cluster. Remote gateway with Unity Express and SRST configured. Normal operation is MGCP.

I have the following dial-peers set up for Unity Express for SRST mode:

dial-peer voice 1 voip

description Local NM-CUE (CME) Voicemail

destination-pattern 3030

session protocol sipv2

session target ipv4:172.17.0.99

dtmf-relay sip-notify

codec g711ulaw

no vad

!

dial-peer voice 2 voip

description Local NM-CUE (CME) Auto Attendant

destination-pattern 3050

session protocol sipv2

session target ipv4:172.17.0.99

dtmf-relay sip-notify

codec g711ulaw

no vad

!

dial-peer voice 3 voip

description Local NM-CUE (CME) Greeting Management System

destination-pattern 3040

session protocol sipv2

session target ipv4:172.17.0.99

dtmf-relay sip-notify

codec g711ulaw

no vad

When SRST becomes active, I can dial into any of those 3 extensions from a phone without issue.

The main AA number is 555-555-8321. This is not the same as the AA dial-peer which is 3050. So I have translation rules and dial-peers set up as follows:

voice translation-rule 100

rule 1 /^15555558322/ /2305/

rule 2 /^8322/ /2305/

rule 3 /^15555558323/ /2318/

rule 4 /^8323/ /2318/

rule 5 /^15555558324/ /2309/

rule 6 /^8324/ /2309/

rule 7 /^15555558321/ /3050/

rule 8 /^8321/ /3050/

rule 9 /^15555558325/ /2308/

rule 10 /^8325/ /2308/

voice translation-profile SRST-1

translate called 100

dial-peer voice 83215 voip

translation-profile incoming SRST-1

destination-pattern 1555555832[1-5]

session target ipv4:172.17.0.101

incoming called-number 1555555832[1-5]

dial-peer voice 983215 voip

translation-profile incoming SRST-1

translation-profile outgoing SRST-1

destination-pattern 832[1-5]

session target ipv4:172.17.0.101

no vad

So from the translation rules we see that an incoming call to 15555558321 gets translated to 3050.

The problem is I get a busy signal.

When I dial 8321 from a phone, I get a busy signal as well.

When I dial 8322, the phone with extension 2305 rings.

I'm not sure what I am doing wrong or even if this is normal behaviour.

Any help would be great on this.

Jon Woloshyn

I have this problem too.
0 votes
Correct Answer by kelvin.blair about 8 years 7 months ago

I can't see the entire configuration, so i'm going to go out on a limb and ask the following questions:

Do you have the following information in the configuration:

voice service voip

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip

Fallback to H323

int faste 0/0.X

h323-gateway voip interface

h323-gateway viop bind src X.X.X.X

Call application alter default

Have you tried using number expressions instead of Translation profiles?

num-exp 8321 3050

num-exp 15555558321 3050

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Overall Rating: 5 (1 ratings)
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Rajesh Revuru Wed, 04/23/2008 - 08:38

Do you have translation-profile incoming on your POTS dial-peer too? I only see it is applied to VOIP dial-peer

*******************

dial-peer voice 983215 voip

translation-profile incoming SRST-1

translation-profile outgoing SRST-1

destination-pattern 832[1-5]

session target ipv4:172.17.0.101

no vad

****************

jonwoloshyn Wed, 04/23/2008 - 08:59

I don't think I need a pots dial-peer for that.

The default dial-peer is handling all the incoming calls:

dial-peer voice 10 pots

incoming called-number .

direct-inward-dial

port 0/0/0:23

After matching on this dial-peer the call will match the following dial-peer as an outbound dial-peer.

dial-peer voice 83215 voip

translation-profile incoming SRST-1

destination-pattern 1555555832[1-5]

session target ipv4:172.17.0.101

incoming called-number 1555555832[1-5]

It then hits the translation patterns and the called number gets modified.

Then it has to match a dial-peer for the modified called-number.

Works great for the dynamic dial-peers set up by SRST. Won't work for my dial-peers set up for Unity Express. Might have something to do with SIP but I'm not sure.

When I call in to 1555-555-8322, the correct extension rings so I don't think it is a problem with the inbound dial-peer.

kelvin.blair Wed, 04/23/2008 - 09:05

You may be running into a redirection problem. The debugs in my last post will tell us that. You mention that if you call 8322 from outside it rings the extension fine, but does it go to VM with no issue?

kelvin.blair Wed, 04/23/2008 - 08:53

Can you run these debugs and post the output?

debug voip dialpeer inout

debug voice ccapi inout

debug voice translation

Correct Answer
kelvin.blair Wed, 04/23/2008 - 09:28

I can't see the entire configuration, so i'm going to go out on a limb and ask the following questions:

Do you have the following information in the configuration:

voice service voip

allow-connections h323 to h323

allow-connections h323 to sip

allow-connections sip to h323

allow-connections sip to sip

Fallback to H323

int faste 0/0.X

h323-gateway voip interface

h323-gateway viop bind src X.X.X.X

Call application alter default

Have you tried using number expressions instead of Translation profiles?

num-exp 8321 3050

num-exp 15555558321 3050

kelvin.blair Wed, 04/23/2008 - 10:27

Cool!.. Glad to hear it is working now with the number expansions statements. The orginal problem seems to be with the translation pattern and how you are translating the digits.

shaftstri Mon, 05/26/2008 - 00:56

Hi All

I understand CUE works with SRST. I need to confirm that it is Cisco/TAC supported

Thanks

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