ccme + sip provider

Unanswered Question
Feb 26th, 2007

Hello,

I tried to connect my ccme 4.0 with a sip provider. I managed to configure it for outgoing calls, but I'm unable to get incoming calls...

Can sombody show me how it works ?

Thanks

I have this problem too.
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gogasca Mon, 02/26/2007 - 07:11

Configure an incoming dial peer which match codecs used by your provider.

For example:

dial-peer voice 4001 voip

voice-class codec 1

incoming called-number 5...

dtmf-relay rtp-nte

Where 5... are your extensions and in class codec you have:

voice class codec 1

codec preference 1 g729r8

codec preference 2 g711ulaw

HTH

kern.jerome Mon, 02/26/2007 - 07:28

Thanks, but my provider send me a 10 digits number and I want to transalte it in a number like 2..

Paolo Bevilacqua Mon, 02/26/2007 - 07:40

Just configure the 10 digits number as secondary in the corresponding ephone-dn. You can also configure sip-ua and it will register this number (no-reg primary) to the ITSP.

Then, "debug ccsim message" is your friend and will tell you what the ITSP is sending, and why is rejected.

gogasca Mon, 02/26/2007 - 07:46

You can create a translation rule like the following:

Lets say your provider sent you:

919 991 5454

voice translation-rule 100

rule 1 /^919....\(...\)$/ /\1/

voice translation-profile TOEXT

translate called 100

dial-peer voice 2000 voip

translation-profile incoming TOSIP

session protocol sipv2

dtmf-relay rtp-nte

codec g711ulaw

This should do it.

Number converted will be 454

or use:

dialplan-pattern command

http://www.cisco.com/en/US/products/sw/iosswrel/ps5207/products_administration_guide_chapter09186a00801f12a8.html#wp2261346

Paolo Bevilacqua Mon, 02/26/2007 - 08:09

Again, the voice translation-rule, although a very powerful tool, is not necessary in this case.

The secondary number in ephone-dn, is instead what the CME architecture gives you to easily route call coming in as "pots" numbers.

kern.jerome Mon, 02/26/2007 - 08:06

Here you can see part of my configuration. It still don't work...

The dial-peer 3 should the one which is used for incoming calls ...

dial-peer voice 1 voip

translation-profile outgoing OutgoingProfile

destination-pattern T

session protocol sipv2

session target ipv4:213.161.201.200

dtmf-relay rtp-nte

codec g711alaw

!

dial-peer voice 2 voip

destination-pattern 8..

session protocol sipv2

session target ipv4:10.10.0.33

session transport tcp

dtmf-relay rtp-nte

codec g711alaw

!

dial-peer voice 3 voip

session protocol sipv2

incoming called-number 0877192763

dtmf-relay rtp-nte

codec g711alaw

(...)

sip-ua

authentication username login password ****

sip-server ipv4:213.161.201.200

notify telephone-event max-duration 500

(...)

ephone-dn 1 dual-line

number 200 secondary 0877192763 no-reg primary

Paolo Bevilacqua Mon, 02/26/2007 - 08:14

can you post output of "debug ccsip message" ? You will need of course "term mon" to see it on a telnet connection.

Dial-peer 3, being an incoming only, does not even need "session protocol", but I'm not sure if this is the problem.

kern.jerome Mon, 02/26/2007 - 08:22

I get no messages from a debug command using a console connection.

I will phone the provider to see if he can see something wrong ...

Paolo Bevilacqua Mon, 02/26/2007 - 12:30

You must be able to see debug output, that is key to working with routers. Telnet to the router, and type "terminal monitor" before or after the debug commands.

kern.jerome Tue, 02/27/2007 - 00:30

I can show debug, but only during outgoing calls... nothings appears during an incomming call. That's why I'm trying to find a solution directly with my provider.

I'll let you know

Paolo Bevilacqua Tue, 02/27/2007 - 13:24

Possibly you need to register with ITSP ro receive incoming calls. Create a "voice register dn" with the number that ITSP wants to be registed and a register server in sip-ua.

Paolo Bevilacqua Tue, 02/27/2007 - 16:45

Woops, I see that your local phones are SCCP. Then no "voice register dn" is necessary, just "registrar" under "sip-ua".

kern.jerome Wed, 02/28/2007 - 03:16

Another question..

If I configure my cisco like you said, the ephone-dn registers the phone number with the secondary line.

Is possible to register with a dial-peer or something else to use hunt-groups ?

Paolo Bevilacqua Wed, 02/28/2007 - 06:27

Hi Kern,

You can indeed register pots dial-peers but I don't think that is what you want.

To "register" an hunt group, just create an ephone-dn with number assigned by the ITSP, either primary or secondary, and registering. Then in this ephone-dn, configure Call Forward All to the actual pilot number of the hunt group you want to use.

kern.jerome Wed, 02/28/2007 - 06:58

The problem is that I have to put my sip login instead of the e164 number to make it work (my sip provider told me that):

ephone-dn 1 dual-line

number 200 secondary login_sip no-reg primary

I don't know how it works if I had multible sda (e164 numbers) with the same login...

Paolo Bevilacqua Wed, 02/28/2007 - 07:13

Well, it should work.

The number in ephone-dn must be a number and not an username, because it's what CME will try to register, and it's not possible to register an username.

In many cases, ITSP assigns the username to be the same as number, but not always.

The thing is that under sip-ua you can configure "credentials username/password/realm", so you can register as many numbers you want using the same set of credentials.

kern.jerome Wed, 02/28/2007 - 07:57

Here you can see the sip-ua section and my ephone-dn configured with the login and the e 164number. I also pasted the sh sip-ue register status associated

Do you think that the problem is coming from my provider?

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

sip-ua

authentication username "sip_login" password ****

registrar ipv4:registrar_ip_address expires 60

sip-server ipv4:registrar_ip_address

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

ephone-dn 1 dual-line

number 200 secondary "sip_login" no-reg primary

label 200

description 200

Router#sh sip-ua register status

Line peer expires(sec) registered

============ ============= ============ ===========

"sip_login" 20008 45 yes

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

ephone-dn 1 dual-line

number 200 secondary "e164_number" no-reg primary

label 200

description 200

Router#sh sip-ua register status

Line peer expires(sec) registered

============ ============= ============ ===========

"e164_number" 20008 110 no

Paolo Bevilacqua Wed, 02/28/2007 - 08:16

Hi,

What can I say, this ITSP does things differently.

If I understand correctly, he wants you to register the username only, and I suppose that they will present calls to all your numbers to the system where you registered from.

To the CME, that does do not make much a difference, as long the SIP username contains only digits/abcd to be used as number in ephone-dn.

Configure an ephone-dn, single-line, primary number, registering as the username assigned. The only purpose of this will be triggering registration. Or, just do a you did an reause an existing ephone-dn as secondary number.

Configure all the others ephone-dn with the secondary numbers that you got, and no-reg both.

Ephone-hunt also supports secondary number and should work as well.

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