I have connected an Innovaphone IP3000 (with PBX functionality) as a H323 Gatekeeper to an Asterisk installation. I have a SIP Phone (Polycom SoundPoint IP 330) connected to the Asterisk and an H323 phone (innovaphone 200) connected to the IP3000. When I place a call from the innovaphone to the Polycom everything works fine. But when I place a call from the Polycom to the innovaphone, the audio from the Polycom doesn’t reach the innovaphone and the audio form the innovaphone sounds like it comes from a tin can.
Can anybody explain to me what is going on here or give tips on how to solve the problem? It happens both with Asterisk 1.2 and 1.4 . I have turned off the firewall for the asterisk machine, but that didn’t help. Changing the codecs on both ends from alaw to ulaw didn’t help either.
this is my h323.conf:
[code][general]
port = 1720
bindaddr =
disallow=all
allow=alaw
dtmfmode=rfc2833
dtmfcodec=101
gatekeeper =
AllowGKRouted = yes
context=inno
[asterisk]
type=h323
context=inno
[/code]
This is my sip.conf:
[code][general]
context=default
bindport=5060
bindaddr=217.110.87.27
srvlookup=yes
allow=alaw
[authentication]
[polycom]
type=friend
context=internal
secret=
host=dynamic
username=polly
disallow=all
allow=alaw
progressinband=no
[/code]
This is the extension from my extensions.conf:
[internal]
exten => 34,1,Dial(H323/873,20)
This is the message log from the point where I dial the number until I have picked up the receiver:
Oct 24 09:01:53 DEBUG[2489] chan_sip.c: Setting NAT on RTP to 0
Oct 24 09:01:53 DEBUG[2489] chan_sip.c: Stopping retransmission on '5ff6bd6f-8f6b45c4-17f4e81d@217.110.87.252' of Response 1: Match Found
Oct 24 09:01:53 DEBUG[2489] chan_sip.c: Setting NAT on RTP to 0
Oct 24 09:01:53 DEBUG[2489] chan_sip.c: Checking SIP call limits for device polycom
Oct 24 09:01:53 DEBUG[2489] chan_sip.c: build_route: Contact hop: <sip:polycom@217.110.87.252>
Oct 24 09:01:53 VERBOSE[2504] logger.c: -- Executing Dial("SIP/polycom-08445b10", "H323/873|20") in new stack
Oct 24 09:01:53 DEBUG[2504] chan_h323.c: type=H323, format=4, data=873.
Oct 24 09:01:53 DEBUG[2504] chan_h323.c: Extension: Host: 873
Oct 24 09:01:53 DEBUG[2504] chan_h323.c: Setting NAT on RTP to 0
Oct 24 09:01:53 DEBUG[2504] chan_h323.c: Placing outgoing call to 873, 101
Oct 24 09:01:53 VERBOSE[2504] logger.c: -- Called 873
Oct 24 09:01:59 DEBUG[2506] chan_h323.c: Received ALERT/PROGRESS message for self-generated tones
Oct 24 09:01:59 VERBOSE[2504] logger.c: -- H323/873-1 is ringing
Oct 24 09:01:59 DEBUG[2482] channel.c: Avoiding initial deadlock for 'H323/873-1'
Oct 24 09:01:59 VERBOSE[2504] logger.c: -- H323/873-1 is ringing
Oct 24 09:02:02 VERBOSE[2504] logger.c: -- H323/873-1 answered SIP/polycom-08445b10
Oct 24 09:02:02 DEBUG[2504] channel.c: Dropping duplicate answer!
Oct 24 09:02:02 DEBUG[2489] chan_sip.c: Stopping retransmission on '5ff6bd6f-8f6b45c4-17f4e81d@217.110.87.252' of Response 2: Match Found
Apart from Wireshark, is there any way to log the H.323 protocol steps?