05-13-2009 11:42 AM - edited 03-15-2019 06:00 PM
I team I am trying to configuring a SIP account for SIP provider on a callmanager ver 6.DOes any body have any experience:
My scenary
Cisco IP-Phone (SCCP) ---> CuCM6.0(H.323) ---> Cisco Router 3825 (12.4) ---> SIP-Trunk to SIP-Provider
I configure the sip-ua on Cisco Router ( Voice Gateway )and its regist OK with the provider.
I add an Route patern on the CCM and a dial-peer on the Gateway, but I am not able to finalize the call
Solved! Go to Solution.
05-14-2009 12:15 AM
we dont support SIP account registration in cucm.
If you already did that in router itself,
make sure you are pointing to correct ip address, proper sip binding is configured in router, also incoming and outgoing dial-peers are configured.
which calls r failing? incoming or outgoing?
05-14-2009 12:15 AM
we dont support SIP account registration in cucm.
If you already did that in router itself,
make sure you are pointing to correct ip address, proper sip binding is configured in router, also incoming and outgoing dial-peers are configured.
which calls r failing? incoming or outgoing?
05-14-2009 02:10 AM
05-14-2009 03:17 AM
05-14-2009 04:12 AM
first make sure u have the follwoing commands in ur gateway
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
sip
then can you add the following commands to your dial-peer i mean the one you attached and let me know if t works or not
answer-address .T
incoming called-number .
also make sure there is no other dial-peers match the number you dial in your sip dialpeer becuase i have seen you have preference as 1
if did not work after the above changes do the following to ur dial-peer
ial-peer voice 444 voip
description SIP CALL
translation-profile outgoing SIPCALLCENTRIC
preference 0
destination-pattern 901T
answer-address .T
incoming called-number .
codec g711ulaw
voice-class sip url sip
session protocol sipv2
session target dns:callcentric.com
no vad
u may need to add dtmf for future use but this has not effect to call in or out
dtmf-relay rtp-nte
good luck
05-14-2009 05:08 AM
Hi just add ( answer-address .T
incoming called-number . ) and ist working fine from Phone on the CCME but its stil not working from ext on CCM
05-15-2009 08:14 AM
Hi Team
ANy body have any idea?
05-16-2009 09:56 AM
Hi Team
Problem solved
I create a special SIP Profile and a SIP trunk . I add a route patern using this SIP trunk
Thanks to all
05-22-2009 02:17 AM
Hello,
Please, can you post the configuration of your gateway ?
05-22-2009 02:38 AM
Hello,
Please, can you post the configuration of your gateway ?
05-22-2009 04:13 AM
Hi Driss
See in attac , my dial-peer and some rule
In my case in add a especial SIP trunk in call manager
Any other question please let me now
If you want this is my msn
05-26-2009 06:13 AM
Thanks Afragoso.
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