Translation rules and profile dial-peer

[enkli]

HI all,

I have got sip number from numbers from SIP provider so far the translation rules profiles and dial peers are as below:

voice translation-rule 1
rule 1 /022799232/ /501/
!
voice translation-rule 2
rule 1 /5../ /022799232/

and translation profiles

voice translation-profile Inbound
translate called 1
!
voice translation-profile Outbound
translate calling 2

Having dial peer as below:

dial-peer voice 100 voip
translation-profile incoming Inbound
session protocol sipv2
session target sip-server
session transport udp
incoming called-number 022799232
dtmf-relay rtp-nte
!
dial-peer voice 201 voip
description National
translation-profile outgoing Outbound
destination-pattern 02.......
session protocol sipv2
session target sip-server
session transport udp
voice-class sip profiles 1
dtmf-relay rtp-nte sip-kpml
no vad

When I Call in dial peer 201 is selected and calls are not translated or rings two times and no more.

SO I am not clear how to workaround the situation.

Any idea on organizing?

So I have the same number for incoming but it goes to outgoing dial-peer.

I do not understand how.

Please any idea how to organize things.

[b.winter]

I hope, I understood it correctly. Please correct me, if I'm wrong.

Assuming, DP 100 is inbound from provider, 201 is outbound to CUCM and you are calling the number 022799232 from external.

If you are calling the number 022799232, then you would hit dial-peer 100 as incoming dial-peer.

On this dial-peer, you would translate the number 022799232 to 501.

But the dial-peer 201 cannot match, because you are trying to match 501 with the pattern "02.......".

Therefore, you don't have an outgoing dial-peer that matches.

When you are calling out from CUCM to PSTN and your Calling Number is 022799232 (e.g. 022799232 calls 9011123123), then it will hit dial-peer 201 as inbound dial-peer, because the destination-pattern in this case matches the calling number in inbound direction.

But then, there is no outbound dial-peer.

--- Please rate this post as "Helpful" or accept as a solution, if your question has been answered ---

[b.winter]

Hi,

Assuming, you call 022799232 from external into the CUBE.

Then it would hit dial-peer 100, because of "incoming called-number 022799232".

Because you have a translation-profile applied, it converts the called number 022799232 to 501.

But then, you don't have an outbound dial-peer available, since dial-peer 201 only matches for called number "02.......".

--- Please rate this post as "Helpful" or accept as a solution, if your question has been answered ---

[enkli]

so you think to have an dial peer

dial-peer voice 203 voip
description Internal
destination-pattern 5..
session protocol sipv2

session server 192.168.10.5 (my cme ip)

[Roger Kallberg]

Is the gateway receiving the call from PSTN not the CME?

[enkli]

From PSTN

[enkli]

Phones are registerd to CME

[Roger Kallberg]

The question was is not the router that receive the calls from the public phone network, aka PSTN, not the same as where you run CME?

[enkli]

CME is behind nat and is receving calls from PSTN. Phonese are registered as SIP endpoints to CME

[Roger Kallberg]

If so the suggested changes by me should accomplish what you asked about.

[Roger Kallberg]

Make these changes to your configuration for the call to hit dial peer 100 in the inbound direction.

voice class uri PSTN sip
 host ipv4:<IP address of service provider SBC>
!
dial-peer voice 100 voip
 no session target sip-server
 no incoming called-number 022799232
 incoming uri via PSTN

[enkli]

I have a range of addresses from SIP is coming from multiple ones

[Roger Kallberg]

You can have multiple addresses or even a range defined in the configuration. Please have a look at this document for additional details.

https://www.cisco.com/c/en/us/support/docs/voice/ip-telephony-voice-over-ip-voip/211306-In-Depth-Explanation-of-Cisco-IOS-and-IO.html

[Roger Kallberg]

If you for some reason cannot use a match on information in the incoming invite, which is the recommendation, you could use “incoming called-number .” to match any incoming calls on the intended incoming dial peer. However this is not recommended for SIP trunk connections. Are there anything else in the invite that you could use, like To or From? The link I provided have a few different options for this outlined. Try yourself around for the various options at hand to find one that works for you.

[enkli]

I woud like to play with my user ID ecause it will be shoretes

so

voice class uri PSTN sip
 userid 022799232@2.2.2.2

and

translation-profile incoming Inbound
session protocol sipv2
session target sip-server
session transport udp
incoming uri to PSTN
dtmf-relay rtp-nte

Is it ok?

Thank you