SIP Trunk CallManger

Answered Question
May 13th, 2009

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

I have this problem too.
0 votes
Correct Answer by gogasca about 7 years 8 months ago

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?

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Overall Rating: 5 (3 ratings)
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Correct Answer
gogasca Thu, 05/14/2009 - 00:15

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?

Marwan ALshawi Thu, 05/14/2009 - 04:12

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

Afragoso2009 Thu, 05/14/2009 - 05:08

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

Afragoso2009 Sat, 05/16/2009 - 09:56

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

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