CAN NOT MAKE OUTGOING CALLS OR RECEIVE CALLS ON SIP LINE

[macdonaldmauye]

hello guys please help....

I have a CUCM interfacing with a 2901voice gateway. In the gateway i have configured a SIP trunk from a service provider with IP add 77.X.X.X .

Now i have configured a SIP trunk in CUCM with destination IP 192.168.1.X of the voice gateway and also configured a voice gateway .

i have configured a route pattern poiniting to the SIP trunk i created in CUCM. But i can not seem to receive any calls or make any calls out as i am receiving a busy tone please help someone.

Below is my voice gateway config

dial-peer voice 8677 voip
description DID [263] OUTGOING local calls via VoIP Line
translation-profile incoming IncomingVoIP
translation-profile outgoing DIDCall
destination-pattern [263,0][0,1,2,3,4,5,6,7,8,9]T
session protocol sipv2
session target ipv4:77.[SIP SERVER ADD FROM SERV PROVIDER]
incoming called-number .
voice-class codec 1
voice-class sip dtmf-relay force rtp-nte
dtmf-relay rtp-nte h245-alphanumeric h245-signal cisco-rtp
no vad
!
dial-peer voice 1000 voip
description *** H323 TRUNK TO CUCM ***
destination-pattern 1...
session target ipv4:{CUCM IP ADD}
voice-class codec 1
dtmf-relay h245-alphanumeric
ip qos dscp cs3 signaling
no vad
!
dial-peer voice 1001 voip
description *** CALL AA SHORT CODE ***
destination-pattern *89
session target ipv4:{CUCM IP ADD}
voice-class codec 1
ip qos dscp cs3 signaling
no vad
!
!
sip-ua
credentials username 263[SIPNUMBER] password 7 040A5E545D701C18584F5654 realm sip.zol.co.zw
authentication username 263[SIPNUMBER] password 7 040A5E545D701C18584F5654 realm sip.zol.co.zw
authentication username 263[SIPNUMBER] password 7 040A5E545D701C18584F5654
registrar dns:sip.zol.co.zw:5060 expires 1800
sip-server dns:sip.zol.co.zw:5060

[Jaime Valencia]

OK, so, you have a SIP trunk on CUCM, but you're using H323 dial peers in the GW??

You should be using SIP all the way to avoid problems with protocol translation, fix that.

No idea about your dial plan or CUCM configuration, but looking at this:

destination-pattern [263,0][0,1,2,3,4,5,6,7,8,9]T

[263,0] -> Not sure if you think this would match calls that start with 236 (as a 3 digit number) or 0, but that's not how it works, this matches a SINGLE DIGIT that can be 2 OR 6 OR 3 OR 0

[0,1,2,3,4,5,6,7,8,9] -> You can simply replace this with a dot .

What troubleshooting have you done so far?

[macdonaldmauye]

my apologies the h.323 dial peers are for the intergration between te cisco cucm and the avaya IP500v2 unit.

Thanky you let me change on the dial plan

[Aseem Anand]

Hi,

can you do the following:

conf t

no logging console

no logging monitor

no logging rate-limit

no logging queue-limit

service timestamps debug datetime msec localtime show-timezone

service timestamps log datetime msec localtime show-timezone

logging buffered 5000000 debugging

service sequence

Enable the below debugs:

Debug ccsip messages

debug voice ccapi inout

Make a test call and share the debugs

Aseem Anand

end

[macdonaldmauye]

on debugging ccsip messages if i make an outgoing call the call is silent for a moment then gives a busy tone but i can not get any output of the debug on the voice gateway

[macdonaldmauye]

Hello guys...
So what i did i changed the session target ipv4 address and also instade of using resistrar and sip-server dns i used the IP addesss.
and my sip line registered.
Next i created an h.323 gateway from cucm to voice gateway and configured my routes and my phones could dial out and receive.....Thanks very much for your help.