Dinesh Nair
2006-Apr-04 21:54 UTC
[Asterisk-Users] asterisk-ooh323, asterisk 1.2.6 and netmeeting
has anyone managed to get these three beasties to work together ? we're
using ooh323 from asterisk-addons-1.2.2, asterisk 1.2.6 and microsoft
netmeeting default from windows xp.
the symptoms are that calls from a SIP client to NetMeeting rings on
NetMeeting, but upon answering the call in NetMeeting, no audio is passed
between the two. eventually, the call times out and hangs up.
on a reverse call from NetMeeting to the SIP client, the SIP client rings
and when it's answered, the same thing happens, i.e. no audio is passed.
the same happens when netmeeting calls an IVR-related app like Directory,
SayDigits et al.
my ooh323.conf file is attached. also, here's the asterisk console output
with ooh323 debug on:
NetMeeting H323 to SIP
--- onNewCallCreated ooh323c_7
+++ onNewCallCreated ooh323c_7
--- ooh323_onReceivedSetup ooh323c_7
--- find_user
+++ find_user
Adding capabilities to call(incoming, ooh323c_7)
--- configure_local_rtp
+++ configure_local_rtp
+++ ooh323_onReceivedSetup - Determined context default, extension 6384
--- onAlerting ooh323c_7
--- find_call
+++ find_call
+++ onAlerting ooh323c_7
-- Executing Dial("OOH323/mms mms-fa6a",
"SIP/6384|40|owWtT") in new stack
-- Called 6384
-- SIP/6384-d9f2 is ringing
----- ooh323_indicate 3 on call ooh323c_7
++++ ooh323_indicate 3 on ooh323c_7
-- SIP/6384-d9f2 answered OOH323/mms mms-fa6a
----- ooh323_indicate -1 on call ooh323c_7
Apr 4 18:16:34 WARNING[3021]: src/chan_h323.c:952 ooh323_indicate: Don't
know how to indicate condition -1 on ooh323c_7
++++ ooh323_indicate -1 on ooh323c_7
--- ooh323_answer
+++ ooh323_answer
-- Attempting native bridge of OOH323/mms mms-fa6a and SIP/6384-d9f2
--- onCallEstablished ooh323c_7
--- find_call
+++ find_call
+++ onCallEstablished ooh323c_7
--- onCallCleared ooh323c_7
--- find_call
+++ find_call
== Spawn extension (macro-stdexten, s-DIAL, 1) exited non-zero on
'OOH323/mms mms-fa6a' in macro 'stdexten'
== Spawn extension (macro-stdexten, s-DIAL, 1) exited non-zero on
'OOH323/mms mms-fa6a'
--- ooh323_hangup
hanging mms mms
+++ ooh323_hangup
--- ooh323_destroy
Destroying mms mms
+++ ooh323_destroy
SIP to H323 NetMeeting
-- Executing Dial("SIP/6384-b575",
"OOH323/6985|40|owWtT") in new stack
--- ooh323_request - data 6985 format 0x8 (alaw)
--- find_peer
+++ find_peer
+++ ooh323_request
--- ooh323_call- 6985
+++ ooh323_call
-- Called 6985
--- onNewCallCreated ooh323c_o_3
--- find_call
+++ find_call
setting callid number 6384
Outgoing call 6985(ooh323c_o_3) - Codec prefs - (ulaw)
Adding capabilities to call(outgoing, ooh323c_o_3)
Adding g711 ulaw capability to call(outgoing, ooh323c_o_3)
--- configure_local_rtp
+++ configure_local_rtp
+++ onNewCallCreated ooh323c_o_3
--- onAlerting ooh323c_o_3
--- find_call
+++ find_call
+++ onAlerting ooh323c_o_3
-- OOH323/6985-e521 is ringing
--- onCallEstablished ooh323c_o_3
--- find_call
+++ find_call
+++ onCallEstablished ooh323c_o_3
-- OOH323/6985-e521 answered SIP/6384-b575
-- Attempting native bridge of SIP/6384-b575 and OOH323/6985-e521
--- ooh323_hangup
hanging 6985
+++ ooh323_hangup
== Spawn extension (macro-stdexten, s-DIAL, 1) exited non-zero on
'SIP/6384-b575' in macro 'stdexten'
== Spawn extension (macro-stdexten, s-DIAL, 1) exited non-zero on
'SIP/6384-b575'
--- onCallCleared ooh323c_o_3
--- find_call
+++ find_call
+++ onCallCleared
--- ooh323_destroy
Destroying 6985
+++ ooh323_destroy
--
Regards, /\_/\ "All dogs go to heaven."
dinesh@alphaque.com (0 0) http://www.alphaque.com/
+==========================----oOO--(_)--OOo----==========================+
| for a in past present future; do |
| for b in clients employers associates relatives neighbours pets; do |
| echo "The opinions here in no way reflect the opinions of my $a
$b." |
| done; done |
+=========================================================================+
--
Regards, /\_/\ "All dogs go to heaven."
dinesh@alphaque.com (0 0) http://www.alphaque.com/
+==========================----oOO--(_)--OOo----==========================+
| for a in past present future; do |
| for b in clients employers associates relatives neighbours pets; do |
| echo "The opinions here in no way reflect the opinions of my $a
$b." |
| done; done |
+=========================================================================+
-------------- next part --------------
[general]
h323id=QubeTalkECS
callerid=QubeTalkECS
gatekeeper=DISABLE ; DISCOVER or IP addy
logfile=/var/spool/asterisk/log/h323_log
gateway=no ; or yes
faststart=yes
h245tunneling=yes
port=1720
bindaddr=0.0.0.0
context=default
[6970]
ip=192.168.1.160
type=friend
context=phones
disallow=all
allow=alaw
allow=ulaw
h323id=rosli
email=6970@qc.com
group=1
callgroup=1
pickupgroup=1
accountcode=QC_H323
amaflags=default
callerid=6970 <6970>
[6985]
ip=192.168.1.185
type=friend
context=phones
disallow=all
allow=ulaw
h323id=muzz
email=6985@qc.com
group=1
callgroup=1
pickupgroup=1
accountcode=QC_H323
amaflags=default
callerid=test 323 <6985>
[6999]
ip=dynamic
type=friend
context=phones
disallow=all
allow=alaw
allow=ulaw
h323id=hock
email=sin@qubeconnect.com
group=1
callgroup=1
pickupgroup=1
accountcode=QC_H323
amaflags=default
callerid=Sin Hock Kian <6999>
Avi Miller
2006-Apr-04 22:17 UTC
[Asterisk-Users] asterisk-ooh323, asterisk 1.2.6 and netmeeting
Dinesh Nair wrote:> the symptoms are that calls from a SIP client to NetMeeting rings on > NetMeeting, but upon answering the call in NetMeeting, no audio is passed > between the two. eventually, the call times out and hangs up.I had a similar problem connecting Asterisk to an Avaya IP403 via OOH323: In the end, I removed all the disallow=all and allow=<codec> lines in Asterisk. This seems to have allowed the two systems to overcome the codec negotiation problems they were having and proceed with actual audio transfer. :) I have no idea if this is related, but I thought I'd just throw that out there, if only for testing purposes. cYa, Avi -- National Manager - Special Projects < Sydney / Melbourne / Canberra / Hobart / London /> 2/340 Gore Street T: +61 (0) 3 9486 0411 Fitzroy, VIC F: +61 (0) 3 9486 0611 3065 W: http://www.squiz.net/ .....>> Open Source - Own it - Squiz.net ...../>