It' seems to be RTP problem.
sendto() in rtp.c fails to send rtp packets.
When I change channel from H323 to SIP, no problem.
Any idea?
Regards,
Jason.
---------- from /var/log/asterisk/full----------
[Nov 26 18:57:15] DEBUG[21863] rtp.c: RTP Transmission
error of packet 39407 to 192.168.1.116:8528: Invalid
argument
[Nov 26 18:57:15] DEBUG[21863] rtp.c: RTP Transmission
error of packet 39408 to 192.168.1.116:8528: Invalid
argument
[Nov 26 18:57:15] DEBUG[21863] rtp.c: RTP Transmission
error of packet 39409 to 192.168.1.116:8528: Invalid
argument
--- Jason Kim <asterjason@yahoo.com> wrote:
> Hi,
>
> My configuration is SipPhone<-->*1<--->*2.
> My asterisk version is 1.4beta3.
> I installed pwlib,openh323,chan_h323.
> When i call from
> SipPhone--(SIP)-->asterisk1---(H323)-->asterisk2,
> there is no audio.
> Using 'rtp debug', I can see that rtp packets are
> being received.
> Rtp packets are being exchanged.
> I also tested chan_ooh323, but to fail.
> Can anyone recommand best h323 channel driver?
>
> Regards,
> Jason.
>
> #------h323.conf for both------------------------
> [general]
> port = 1720
> bindaddr = 0.0.0.0
> disallow=all
> allow=ulaw
> context=default
>
> #------dial plan of asterisk1--------------------
> exten => *59,1,Wait(1)
> exten => *59,2,Dial(H323/3500@192.168.1.150)
>
> #------dial plan of asterisk2--------------------
> exten => 3500,1,Playback(hello)
> exten => 3500,2,Hangup()
>
> #------console messages with 'rtp debug'---------
> -- Executing [*59@from-internal:3]
> Dial("SIP/3503-0921cb88", "H323/3500@192.168.1.150")
> in new stack
> -- Requested transfer capability: 0x00 - SPEECH
> -- Making call to 3500@192.168.1.150:1720 without
> gatekeeper.
> == New H.323 Connection created.
> -- root is calling host
> 3500@192.168.1.150:1720
> -- Call token is ip$localhost/29426
> -- Call reference is 29426
> -- DTMF Payload is [pt=101]
> -- Called 3500@192.168.1.150
> Setting capabilities to 0x8 (alaw)
> Capabilities in preference order is (alaw)
> Allowed Codecs:
> Table:
> G.711-ALaw-64k <1>
> UserInput/hookflash <2>
> UserInput/RFC2833 <3>
> UserInput/dtmf <4>
> Set:
> 0:
> 0:
> G.711-ALaw-64k <1>
> 1:
> UserInput/hookflash <2>
> 2:
> UserInput/RFC2833 <3>
> UserInput/dtmf <4>
>
> -- Sending SETUP message
> -- Transmitting RFC2833 on payload 101
> -- Started logical channel: receiving
> G.711-ALaw-64k
> -- channelsOpen = 1
> External RTP Session Starting
> RTP channel id 1 parameters:
> -- remoteIpAddress: 127.0.0.1
> -- remotePort: 13710
> -- ExternalIpAddress: 192.168.1.116
> -- ExternalPort: 29388
> -- Started logical channel: sending
> G.711-ALaw-64k
> -- channelsOpen = 2
> External RTP Session Starting
> RTP channel id 1 parameters:
> -- remoteIpAddress: 127.0.0.1
> -- remotePort: 13710
> -- ExternalIpAddress: 192.168.1.116
> -- ExternalPort: 29388
> - Progress Indicator: 8
> -- H323/192.168.1.150-3 is making progress
> passing
> it to SIP/3503-0921cb88
> -- Inbound RFC2833 on payload [pt=101]
> Peer capability is G.711-ALaw-64k <1>
> Found peer capability G.711-ALaw-64k <1>, Asterisk
> code is 8, frame size (in ms) is 20
> Peer capability is UserInput/hookflash <2>
> Peer capability is UserInput/RFC2833 <3>
> Peer capability is UserInput/dtmf <4>
> Peer capabilities = 0x8 (alaw), ordered list is
> (alaw)
> =-= In OnConnectionEstablished for call
> 29426
> -- Connection Established with
> "3500"
> -- H323/192.168.1.150-3 answered
> SIP/3503-0921cb88
> -- Received Facility message...
> Got RTP packet from 192.168.1.204:16434 (type
> 00,
> seq 014405, ts 328224084, len 000240)
> Sent RTP packet to 127.0.0.1:13710 (type 08,
> seq
> 008392, ts 000096, len 000160)
> Got RTP packet from 192.168.1.204:16434 (type
> 00,
> seq 014406, ts 328224324, len 000240)
>
>
>
>
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