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SNR on CME going to a call manager

sherif.elella
Level 1
Level 1

here is my CME Config

voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
sip
bind control source-interface GigabitEthernet0/0
bind media source-interface GigabitEthernet0/0

--------------------------------

dial-peer voice 60 voip
preference 1
destination-pattern 332..$
session protocol sipv2
session target ipv4:cucm-01
incoming called-number 332..$
dtmf-relay rtp-nte sip-kpml
no vad
!
dial-peer voice 61 voip
destination-pattern 332..$
session protocol sipv2
session target ipv4:cucm-02
incoming called-number 332..$
dtmf-relay rtp-nte sip-kpml
no vad
!
dial-peer voice 62 voip
preference 2
destination-pattern 332..$
session protocol sipv2
session target ipv4:cucm-03
incoming called-number 332..$
dtmf-relay rtp-nte sip-kpml
no vad

-----------------------------------

ephone-dn 11 octo-line
number 3270
description xxxxxxx
mobility
snr calling-number local
snr 33270 delay 0 timeout 60
snr ring-stop
corlist incoming css-international

ephone 11
device-security-mode none
mac-address xxx.xxx.xxx
type xxx
button 1:11

--------------------------------

and there is a SIP Trunk on Call manager (run on all nodes), that has significant digits to 4 which translates the 332.. to 32..

with the correct calling search space. 

--------------------------------

yet on CME, i get something like "No device handler found". i cant explain the problem.

any idea?

1 Accepted Solution

Accepted Solutions

Brandon Buffin
VIP Alumni
VIP Alumni

Can you dial 33270 directly from a phone on CME? Also, verify the codec in use on the SIP trunk. VOIP dial peers use g.729 by default. If your SIP trunk is expecting a different codec that could be the problem.

Brandon

View solution in original post

2 Replies 2

Brandon Buffin
VIP Alumni
VIP Alumni

Can you dial 33270 directly from a phone on CME? Also, verify the codec in use on the SIP trunk. VOIP dial peers use g.729 by default. If your SIP trunk is expecting a different codec that could be the problem.

Brandon

added a voice class codec and worked, thank you. have another problem now.