Dial-peer Translation-rules when in SRST

[Marcel Tempelman]

Hi,

I was wondering how I can solve the follwing issue in a decent way:

This customer has a CUBE with a SIP-trunk. The CUBE is connected via an SIP-trunk to a CUCM 8.5. When calls come in form the SIP-trunk, the called number should not be translated e.g.:

SIP-trunk	CUBE	CUCM
012-345678	012-345678	012-345678
		(Add zero as prefix) in translation patern)

When in SRST (CME-mode) the called number should be translated to a number consisting of the last 4 digits.

Should I use 2 dial-peers with different prefences ? something like this ? (172.16.1.1 is fictional CUCM ip address)

dial-peer voice 260 voip

preference 1

destination-pattern 012345678

session protocol sipv2

session target ipv4:172.16.1.1

incoming called-number .

voice-class codec 1 

dtmf-relay rtp-nte

no vad

!

dial-peer voice 262 voip

preference 2

translation-profile incoming SRST-mode

destination-pattern 012345678

session protocol sipv2

session target ipv4:172.16.1.1

incoming called-number .

voice-class codec 1 

dtmf-relay rtp-nte

no vad

!

voice translation-rule 12

rule 1 /^01234\(....\)/ /\1/

!

translation-profile SRST-mode

translate called 12

!

My guess is that using the second dial-peer when matching numbers it will match with a ephone-dn of a registered phone. Any ideas ?

Regards,

Marcel.

In Call-manager-fallback it is possible to specify a translation-profile. In CME-mode it falls back on the dial-peer method.

[Ayodeji Okanlawon]

Marcel, ignore my previous posts..Just configure your transltion rule under tekepohony-sevice..so the xaltion rule will be appied only in SRST

telephony-service

translate calling SRST-mode

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"opportunity is a haughty goddess who waste no time with those who are unprepared"

[Marcel Tempelman]

:-)

What I understand form reading the documentation that the translate option is only available when using call-manager-fallback and not in SRST/CME-mode (using telephony-service).

Just checked it on a CUBE-router with a SRST-config and Telephiny-service does not have any translate command available.

Main problem with the current documentation is that there is no real example of SRST with a SIP-trunk and SCCP-phones. Almost every example available uses traditional PSTN-connections with MGCP or if a SIP-trunk is used the example also uses SIP-phones registering themselves at the SIP-ISP :-(.

[Chris Deren]

Alternatively you can use dialplan-pattern under SRST.

HTH,

Chris

[Marcel Tempelman]

Hi Chris,

Dial-plan pattern is from extension to E.164 not from E.164 to extension.

[Chris Deren]

Sure it is, you can use it to convert any digits to any.

Chris

[Ayodeji Okanlawon]

Chris I already suggested this..but been a while I used it I bought into marcelo's idea that it only for expanding local digits..I am sure it can be used too

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"opportunity is a haughty goddess who waste no time with those who are unprepared"

[Marcel Tempelman]

So this should work:

dialplan-pattern 1 012345... extension length 4 5...

If this works what about the (pattern-)preference because the CME-mode will always be on so won't this interfere with normal operation ? Not sure is dailplan-pattern has a higher or lower match preference than a dial-peer.

regards,

Marcel.

[Ayodeji Okanlawon]

The dialplan-pattern as I suggested earlier should be used under telephony-service..So it doesnt come into play during normal operation..

You can use that or just this

telephony-service

dialplan-pattern 1 012345... extension-length 4

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"opportunity is a haughty goddess who waste no time with those who are unprepared"

[Marcel Tempelman]

Found some indepth info:

http://it-certification-network.blogspot.nl/2008/12/voice-translation-profiles-versus.html

Technically the dialplan-pattern creates a pots dial-peer with the expanded number. I've created a dummy dialplan-pattern and when I use dial-peer voice summary I see a lot of extra dial-peers (it creates one for every match available in the pattern). All have preference 0 and are up. I'll have to test this because I'm not sure how this is going to "behave".

BTW only usable for phones not for FXS and or VM numbers.

Nonetheless thanks in advance for the input!

regards.

Marcel.

[Ayodeji Okanlawon]

Thats correct It creates extra dial-peer and yes doesnt work with FXS..This has been used many times in the past..Its not new..Make sure you do in on telehony-service..

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"opportunity is a haughty goddess who waste no time with those who are unprepared"

[Marcel Tempelman]

Must say it feels like a workaround but for this scenario it will work. I do not have to worry about a VM or B-ACD script.

Thanks anyway :-)

[Ayodeji Okanlawon]

If it bothers you too  much..use the dial-peer option...and use the "monitor probe icmp-ping" on the cucm dial-peer.

Create a loopback address on the gateway...configure the session target to point to the loopback address and apply your voice translation rules on that dial-peer..Its that simple

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"opportunity is a haughty goddess who waste no time with those who are unprepared"

[Marcel Tempelman]

Here's an update:

I must say I want to recreate this in a lab but this happened when dialplan-pattern is configured (and not in SRST-mode):

If there are phones configured on the CUBE (in CME/SRST mode) and the called number matches a dialplan-pattern the CUBE tries to contact the local phone with the matching extension. The dialplan-pattern takes precedence over a dial-peer with the same number. I had added these for the SRST-setup but they interfered with normal operation so I have removed them.

[EngineerRivas]

I think this is the best solution and what something I plan to implement soon. without the monitor icmp-ping .  just a session target pointing to the loopback or lan facing interface.