Translation pattern question

[TrickTrick]

Hello,

I'm applying the following translation pattern in my voice gateway, but can't make it work

voice translation-rule 1
rule 1 /0944994433/ /5000/
rule 2 / 5000/ /1000/

the first rule goes to CUCM and phone rings, but if CUCM is not reacheable, I want the second rule to work so the first translation will be translated again and make the 1000 extension ring instead, is that possible ? trying this in SRST router)

thanks

[Ayodeji Okanlawon]

It does not work like that. What you need to do is to configure two translation rules and then apply one for normal call routing and apply the second one during SRST. For SRST configure and  apply the translation profile as follows:

voice translation-rule 10
rule 1 /^0944994433$/ /5000/

voice translation-profile abbr-dial

 translate called 10

++ SCCP phones +++

call-manager-fallback

 translation-profile incoming abbr-dial

!

++ SIP Phones +++

voice register pool  2

 translation-profile incoming abbr-dial

[TrickTrick]

---

 Thank you Ayodeji,
Just want to make sure i'm doing things right, because i'm facing some bugs
should I create a similar dial-peer for SRST too ?

Here's the dial-peer towards CUCM

dial-peer voice 2 voip
destination-pattern 5000
session target ipv4:192.168.1.100
dtmf-relay h245-alphanumeric
codec g711ulaw

should I create another one pointing to VG ip, or create a dial-peer "pots", I tried the last one (pots) but once applied it never go back to the CUCM dial-peer, even when I create 2 traslation patterns

[Ayodeji Okanlawon]

Since this is a h323 dial-peer, you will need to configure the following, so that when CUCM is down, calls will route to SRST gateway and when CUCM is up, calls will always go to CUCM first.

voice class h323 1

h225 timeout tcp establish 2

h225 timeout setup 2

++ dial-peer to CUCM ++

dial-peer voice 2 voip

preference 1

voice-class h323 1
destination-pattern 5000
session target ipv4:192.168.1.100
dtmf-relay h245-alphanumeric
codec g711ulaw

++ dial-peer to SRST gateway IP ++

dial-peer voice 2 voip

preference 2

voice-class h323 1
destination-pattern 5000
session target ipv4:X.X.X.X ( where X.X.X.X is your SRST gateway ip ( you can use loopback address if you have one on the gateway)
dtmf-relay h245-alphanumeric
codec g711ulaw

[TrickTrick]

I'll try this
just to be sure, I don't think I should use preference since in the H323 dial-peer i'm using hunt pilot number, in the SRST ial-peer i'm using a direct number (extension of a phone)
is that correct ?

[Ayodeji Okanlawon]

Is the hunt pilot number registered in SRST? I doubt it, hence why you will need the dial-peer. In SRST the gateway will automatically create dial-peers for your phones with extension on them but for any other number like a hunt pilot you have to tell the gateway where to send the calls to.

[TrickTrick]

Hi,
Unless it's a bug (or i'm doing things wrongly), it won't work at all, tried to apply your recommandations (which I really appreciate a lot) but couldn't make it going :/, sometimes SRST work but CUCM never work again, or none of them work,...

here's the full config including what you suggested to me : https://pastebin.com/K0qBUhvG
- No dial peer for SRST
- 2 translation rules, 1 for CUCM hunt pilot, 2 for a direct extension in SRST mode applied in voice register pool
- H323 voice class applied to CUCM dial-peer

I see that dial-peers for phones created during SRST but it still doesn't work

[Ayodeji Okanlawon]

In your configuration, I don't see the second dial peer to be used during SRST..

[TrickTrick]

Hi,
reading your previouos comment I understood that I will not need it since SRST dial peers are created automatically for phone extensions, right ? I thought the translation rule will do the job and redirect the call to the defined extnsion in it (second rule is translating to a phone extension not hunt pilot).
beside of this do you think the rest is correct in the config ? Should I re-add the SRST dial-peer then ?

[Ayodeji Okanlawon]

Hi Apologies if I was not clear earlier.

You dont need to create dial-peers for phones with extensions because there dial-peers will be automatically created. But you need dial-peers for any other number that is not registered automatically. Your hunt pilot number will not be registered automatically in SRST. You will need to define it for it to be available in SRST and hence you need to create dial-peers to route to it.

[TrickTrick]

Thank you again for your explanation,
did you have the time to check my config ? All I need now is working on the dial-peer ?

I already configured the following dial-peer but it gets always matched and CUCMs dial-peer neve work again

dial-peer voice 100 pots
description all incoming calls to SRST
incoming called-number .
direct-inward-dial
port 0/1/0:0
forward-digits all
now using your suggestion (H323 voice class) it will work ?
Thakn you

[TrickTrick]

Tried the first dial peer mentioned in the following ciscopress article.. and it still doesn't

work http://www.ciscopress.com/articles/article.asp?p=2492950&seqNum=5

It makes the srst translation work always first.. even when phones are registred wit CUCM and translation rules applied... Im confused