10-22-2012 02:13 PM - edited 03-16-2019 01:49 PM
Hello friends,
I have the following scenario:
MXONE (Ericsson)<=> CUBE <=> Asterisk
The configuration of CUBE is:
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
h323
emptycapability
h225 id-passthru
h245 passthru all
sip
bind control source-interface FastEthernet0/0
bind media source-interface FastEthernet0/0
header-passing
asserted-id pai
midcall-signaling passthru
!
!
voice class codec 1
codec preference 2 g711ulaw
!
!
!
!
voice class h323 1
h225 timeout tcp establish 3
call preserve
dial-peer voice 1 voip
destination-pattern 1...
session protocol sipv2
session target ipv4:10.145.3.36
codec g711ulaw
no vad
!
dial-peer voice 2 voip
destination-pattern 4...
voice-class codec 1
voice-class h323 1
session target ipv4:10.145.5.100
dtmf-relay h245-alphanumeric
no vad
!
dial-peer voice 3 voip
answer-address 1...
voice-class codec 1
no vad
!
dial-peer voice 4 voip
answer-address 4...
voice-class codec 1
voice-class h323 1
dtmf-relay h245-alphanumeric
no vad
!
!
sip-ua
!
The call MXONE to Asterisk works fine, signalling and rtp.
The call Asterisk to MXONE rings but doesnt exist audio and after 5 seconds that called party answer the call is disconnect with normal clearing.
Can you help me?
Thanks in advance
10-22-2012 10:40 PM
I guess that dial peer 1 & 3 is used for the call leg to the side that uses SIP. Please add session protocol sipv2 to DP 3, otherwise it will default to use H.323 as the transport protocol.
You should also set the appropriate dtmf relay type to be used on these dial peers.
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10-23-2012 06:54 AM
Hello Roger,
I miss says this:
MXONE (Ericsson)<=H323 Trunk=> CUBE <=SIP Trunk=> Asterisk
Thanks
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