Hi all,
I'm having trouble with H.323 outbound calls, * connects but there is no
sound in both ways.
I'm using X-Lite as SIP client with GSM codec and dialing to ITSP (which
using cisco, I think) over H.323 with G.729 codec. I have 4 digium G.729
licenses installed and this is onli one call.
I tested my * with another ITSP over SIP and G.729 codec and there was
all ok
Here is my configs and consile output:
extensions.conf:
[test]
exten => _XXXXXXXXXXXX,1,Dial,H323/${EXTEN}@111.222.111.222
h323.conf:
[general]
port = 1720
bindaddr = 0.0.0.0
tos=lowdelay
disallow=all
allow=g729
context = test
console output:
*CLI> h.323 debug
H323 debug enabled
*CLI> -- Executing Dial("SIP/234-d01b",
"H323/XXXXXXXXXXXX@111.222.111.222") in new stack
Allowed Codecs:
Table:
G.729A{sw} <1>
G.729{sw} <2>
Set:
0:
0:
G.729A{sw} <1>
G.729{sw} <2>
-- Making call to XXXXXXXXXXXX@111.222.111.222.
== New H.323 Connection created.
-- 234 is calling host XXXXXXXXXXXX@111.222.111.1222
-- Call token is ip$localhost/17498
-- Call reference is 17498
-- Called XXXXXXXXXXXX@111.222.111.222
-- Sending SETUP message
-- Ringing phone for "111.222.111.222"
-- H323/111.222.111.222 is ringing
=*= In CreateRealTimeLogicalChannel for call 17498
-- externalIpAddress: 111.111.111.111
-- externalPort: 19822
-- SessionID: 1
-- Direction: IsReceiver
-- Started logical channel: receiving G.729{sw}
-- channelsOpen = 1
=*= In CreateRealTimeLogicalChannel for call 17498
-- externalIpAddress: 111.111.111.111
-- externalPort: 19822
-- SessionID: 1
-- Direction: IsTransmitter
-- Started logical channel: sending G.729A{sw}
-- channelsOpen = 2
-- Connection Established with "Unknown [111.222.111.222]"
-- H323/111.222.111.222 answered SIP/234-d01b
channelsOpen = 1
*CLI> -- Received RELEASE COMPLETE message...
-- Sending RELEASE COMPLETE
channelsOpen = 0
-- Unknown [111.222.111.222] has cleared the call
== Spawn extension (test, XXXXXXXXXXXX, 1) exited non-zero on
'SIP/234-d01b'
== H.323 Connection deleted.
Could anybody point me what I'm doing wrong
Thanks.
--
Sincerely,
Elman Efendiyev
elman@earlinvest.com
asterisk-users-bounces@lists.digium.com wrote:> Hi all, > > I'm having trouble with H.323 outbound calls, * connects but there is > no sound in both ways. > I'm using X-Lite as SIP client with GSM codec and dialing to ITSP > (which using cisco, I think) over H.323 with G.729 codec. I have 4 > digium G.729 licenses installed and this is onli one call. > I tested my * with another ITSP over SIP and G.729 codec and there was > all ok > Here is my configs and consile output: >we use older version of h323 driver on latest CVS.. this way it works fine. i think you need to look for h323 version before may/april '04 Jeremy, would that be correct? SJ
I am having the same problem and not using the NuFone h323 but the Asterisk-OH323... Inbound to sip from h323 seems to be a problem with audio... Robert A. Huddleston, KF4BYY IT Support Analyst Cavalier Telephone LLC. (Cell) 804.400.3686 rhuddleston@cavtel.com -----Original Message----- From: Senad Jordanovic [mailto:senad@boltblue.com] Sent: Monday, September 20, 2004 10:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] H.323 call problemm (no sound) asterisk-users-bounces@lists.digium.com wrote:> Hi all, > > I'm having trouble with H.323 outbound calls, * connects but there is > no sound in both ways. > I'm using X-Lite as SIP client with GSM codec and dialing to ITSP > (which using cisco, I think) over H.323 with G.729 codec. I have 4 > digium G.729 licenses installed and this is onli one call. > I tested my * with another ITSP over SIP and G.729 codec and there was > all ok Here is my configs and consile output: >we use older version of h323 driver on latest CVS.. this way it works fine. i think you need to look for h323 version before may/april '04 Jeremy, would that be correct? SJ _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users