02-26-2007 06:58 AM - edited 03-14-2019 08:13 PM
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
02-26-2007 07:11 AM
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
02-26-2007 07:28 AM
Thanks, but my provider send me a 10 digits number and I want to transalte it in a number like 2..
02-26-2007 07:40 AM
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.
02-26-2007 07:46 AM
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
02-26-2007 08:09 AM
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.
02-26-2007 08:06 AM
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
02-26-2007 08:14 AM
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.
02-26-2007 08:22 AM
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 ...
02-26-2007 12:30 PM
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.
02-27-2007 12:30 AM
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
02-27-2007 01:24 PM
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.
02-27-2007 04:45 PM
Woops, I see that your local phones are SCCP. Then no "voice register dn" is necessary, just "registrar" under "sip-ua".
02-28-2007 01:07 AM
Thanks for help !
Everything is working now.
02-28-2007 03:16 AM
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 ?
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