Re: [asterisk-users] ooh323 doesn't know what to do when bridging calls

2009-07-13 Thread bilal ghayyad

Dears;

I am having same problem, that when I place a call from the H323 end point 
(even if it is not added in the ooh323.conf), then asterisk handle the call and 
play the wave file in the default context. Also I added endpoint to the 
ooh323.conf and same thing, it keep goes for default context whatever the 
context placed.

My Asterisk vesion is 1.4.25
My Asterisk add-on version is: 1.4.8

What I have to do to capture the call from the IP Phone and route it using the 
correct context that I configured it in the [ ] of the ooh323.conf? Any 
specific thing need to be done?

Regards
Bilal


--
  Hi guys,
  
  I'm trying out ooh323 and couldn't bridge ooh323 and
 sip/zap. 
  I'm using netmeeting and set gateway to my asterisk. 
  
  Here's my CLI dump:
  
    == Spawn extension (h323, , 1)
 exited non-zero on
  'OOH323/(null)-8c76'
      -- Executing [9...@h323:1]
 Dial(OOH323/(null)-3074,
  Zap/8/604xxx) in new stack
      -- Called 8/604xxx
      -- Zap/8-1 is ringing
  [2008-07-02 15:48:55] WARNING[21544]: channel.c:2390
 ast_indicate_data:
  Unable to handle indication 3 for
 'OOH323/(null)-3074'
      -- Zap/8-1 is ringing
      -- Zap/8-1 answered
 OOH323/(null)-3074
  [2008-07-02 15:49:08] WARNING[21544]:
 chan_ooh323.c:1053
  ooh323_indicate: Don't know how to indicate condition
 20 on ooh323c_5
  
  My ooh323.conf:
  
  [general]
  bindaddr=192.168.1.9
  h323id=ObjSysAsterisk
  e164=100
  callerid=asterisk
  gatekeeper = DISABLE
  gateway = yes
  context = h323
  disallow = all
  allow = ulaw
  dtmfmode = rfc2833
  
  
  extensions.conf
  [h323]
  Exten = ,1,Dial(Zap/8/604xxx)
  Exten = ,n,Hangup
  
  604xxx goes to my cell. it rings fine but no
 audio. After I picked
  up from cell, netmeeting still shows watiting for
  to answer
  message.
  
  Any ideas?
 
 I don't like the look of the (null) in the channel names.
 
 If what you quoted was the whole of your ooh323.conf file,
 you don't have
 any peer, user or friend sections. Try adding something
 like:
 
 [h323gw]
 type=friend
 context=h323
 ip=192.168.1.200          (or
 whatever the IP of your remote H323 endpoint is)
 port=1720
 
 If that still doesn't help, please mention what versions of
 asterisk and
 asterisk-addons you are using.
 
 Cheers
 Tony
 -- 
 Tony Mountifield
 Work: t...@softins.co.uk
 - http://www.softins.co.uk
 Play: t...@mountifield.org
 - http://tony.mountifield.org



  

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Re: [asterisk-users] ooh323 doesn't know what to do when bridging calls?

2008-07-03 Thread Tony Mountifield
In article [EMAIL PROTECTED],
Kelvin Chan [EMAIL PROTECTED] wrote:
 Hi guys,
 
 I'm trying out ooh323 and couldn't bridge ooh323 and sip/zap. 
 I'm using netmeeting and set gateway to my asterisk. 
 
 Here's my CLI dump:
 
   == Spawn extension (h323, , 1) exited non-zero on
 'OOH323/(null)-8c76'
 -- Executing [EMAIL PROTECTED]:1] Dial(OOH323/(null)-3074,
 Zap/8/604xxx) in new stack
 -- Called 8/604xxx
 -- Zap/8-1 is ringing
 [2008-07-02 15:48:55] WARNING[21544]: channel.c:2390 ast_indicate_data:
 Unable to handle indication 3 for 'OOH323/(null)-3074'
 -- Zap/8-1 is ringing
 -- Zap/8-1 answered OOH323/(null)-3074
 [2008-07-02 15:49:08] WARNING[21544]: chan_ooh323.c:1053
 ooh323_indicate: Don't know how to indicate condition 20 on ooh323c_5
 
 My ooh323.conf:
 
 [general]
 bindaddr=192.168.1.9
 h323id=ObjSysAsterisk
 e164=100
 callerid=asterisk
 gatekeeper = DISABLE
 gateway = yes
 context = h323
 disallow = all
 allow = ulaw
 dtmfmode = rfc2833
 
 
 extensions.conf
 [h323]
 Exten = ,1,Dial(Zap/8/604xxx)
 Exten = ,n,Hangup
 
 604xxx goes to my cell. it rings fine but no audio. After I picked
 up from cell, netmeeting still shows watiting for  to answer
 message.
 
 Any ideas?

I don't like the look of the (null) in the channel names.

If what you quoted was the whole of your ooh323.conf file, you don't have
any peer, user or friend sections. Try adding something like:

[h323gw]
type=friend
context=h323
ip=192.168.1.200  (or whatever the IP of your remote H323 endpoint is)
port=1720

If that still doesn't help, please mention what versions of asterisk and
asterisk-addons you are using.

Cheers
Tony
-- 
Tony Mountifield
Work: [EMAIL PROTECTED] - http://www.softins.co.uk
Play: [EMAIL PROTECTED] - http://tony.mountifield.org

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