I try to setup the following scenario:
Asterisk --> SIP --> IPIPGW --> H323 --> CUCM --> SCCP --> IPPhone
When I setup a call from asterisk, this fails due to a codec mismatch in the IPIPGW:
1267 : 134 9889630ms.1 +-1 pid:1 Answer 2002 connecting
dur 00:00:00 tx:0/0 rx:0/0
IP 172.20.0.250:12578 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g711ulaw
media inactive detected:n media contrl rcvd:n/a timestamp:n/a
1267 : 135 9889640ms.1 +-1 pid:300 Originate 7090 connecting
dur 00:00:00 tx:0/0 rx:0/0
IP 0.0.0.0:0 SRTP: off rtt:0ms pl:0/0ms lost:0/0/0 delay:0/0/0ms g729r8 pre-ietf
media inactive detected:n media contrl rcvd:n/a timestamp:n/a
Telephony call-legs: 0
However I force G.711 under the dial-peers, for some reason the outbound call-legg stil tries G.729. I also tried to force it using a voice-class codec. The same result.
During call-setup I can see that dial-peer 1 is matched as the inbound dial-peer, so I am not matching the default-dialpeer..
This is my configuration.
voice service voip
allow-connections h323 to sip
allow-connections sip to h323
fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback cisco
modem passthrough nse codec g711alaw
!
!
!
voice class codec 1
codec preference 1 g711ulaw
codec preference 2 g711alaw
!
!
!
voice class h323 1
h225 timeout tcp establish 3
h225 connect-passthru
call start fast
telephony-service ccm-compatible
ccm-compatible
!
!
dial-peer voice 200 voip
destination-pattern 20..
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
codec g711ulaw
no vad
!
dial-peer voice 300 voip
destination-pattern 7090
voice-class h323 1
session target ipv4:172.20.0.1
dtmf-relay rtp-nte
codec g711ulaw
no vad
!
dial-peer voice 1 voip
incoming called-number .
dtmf-relay rtp-nte
codec g711ulaw
!
dial-peer voice 500 pots
destination-pattern 0T
port 0/0/0:15
Any ideas?
