Displaying 20 results from an estimated 7000 matches similar to: "H323 on way voice"
2006 Jun 28
12
Ajax.Updater
Hi,
someone can help me, I am ot able to find the way how to user
Ajax.updaterto test if the request give some positive or negative
result.
I am able only to return the result inside a div.
An example is appreciated.
_______________________________________________
Rails-spinoffs mailing list
Rails-spinoffs-1W37MKcQCpIf0INCOvqR/iCwEArCW2h5@public.gmane.org
2006 Jun 17
6
Canreinvite
I put canreinvite=yes in my sip, for a sipura 3000 and a xlite, however, if
I call the traffic still go throw the asterisk. How come?
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060617/8f4449fa/attachment.htm
2006 Apr 01
1
channel.c:787 channel_find_locked: Avoided initial deadlock for '0x8446b50', 10 retries!
I never so this error.
I am using H323 with Asterisk 1.2.6 Any idea what can be the problem?
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060401/2e8ad498/attachment.htm
2005 Aug 16
3
TAFM
Hi,
I installed this program but I am not able to configure, it does not
want to work.
Someone can help me?
2006 Mar 29
2
H323 behind a Firewall
There is a proble to put an H323 Asterisk server behind an iptables
firewall?
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060329/e0627e85/attachment.htm
2006 Jun 27
5
WebPhone
Hi,
someone know a good webphone, possibily a free one
Thx
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060627/0e83bc29/attachment.htm
2007 Feb 09
2
Chan_Cellphone
Hi,
I download the last svn and I also look around but I cannot find the source,
I only found the patch
http://bugs.digium.com/print_bug_page.php?bug_id=8919
any one can help me out.
thx
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070209/6780fde6/attachment.htm
2007 Oct 23
2
Force codec order
There is a way to force the order of the codecs in the sip.conf since the
allow seams to let know only the accepted codec.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071022/619b8f2b/attachment.htm
2007 Feb 07
1
H323 to SIP - One way voice
Hello all,
I want to use asterisk as protocol converter, H323 to SIP. I am using
Asterisk 1.2.14 with chan_h323 and the free version of g729.
When calling from SIP to H323 everything is fine. But when calling from H323
to SIP, the phone using SIP doesn't hear the other party.
The phones and Asterisk are in the same subnet and the firewall of the
Asterisk box is off. Please advice.
Thank you,
2003 Apr 15
2
Suppport for Asterisk, asterisk-h323 package and Voice Mail
Hi.
I've recently installed Asterisk on my Linux system and added the
asterisk-h323 package. I'm working with a H323 plataform with gatekeepers
and gateways. I'm trying to implement a Voicemail for the endpoints,
that works when the endpoinst are BUSY. I'm a newbie in asterisk so i
need a little help here...
1.- I have succesfully route the BUSY calls from my endpoints to my
2006 Jun 27
1
Capture click
Hi,
I saw one site (bubbleshare) that it is able to caputer the click on the log
in link, however, I cannot understand how they can do that
Someone can explaint it to me?
Thank you
_______________________________________________
Rails-spinoffs mailing list
Rails-spinoffs-1W37MKcQCpIf0INCOvqR/iCwEArCW2h5@public.gmane.org
http://lists.rubyonrails.org/mailman/listinfo/rails-spinoffs
2005 Aug 29
1
TXFAX() status
Hi,
I'm using a script in order to send out my faxes with the application
txfax, therefore, I do not know how to see if the faxes are sent.
Any idea?
2007 Feb 14
1
Strange behaviour with Dial cmd
I have this simple context
I am register to an external provider and when I am not home I would like to
transfer the phone outside
The problem that the call goes in loop
I cannot understand why.
Can you figure out my error?
Thank you
sip.conf
register => user:pass@provider/400
[inside]
exten => _4X.,1,dial(SIP/ext_400_124/5551234444,5,tT)
exten => _4X.,2,hangup
-- Executing
2005 Jul 02
3
What to use h323 or oh323 ???
I m new to asterisk n i've got an IP phone that supports h323 protocol.... but i dont know how to configure asterisk to use it... i m comfortable in using sip & iax softphones.... but there is no h323.conf in /etc/asterisk/ .... i read that i've to compile some files but i m confused regarding h323 & oh323 ...... which one should i use.. plz tell me or atleast give some helpful
2003 Jul 23
4
h323 and oh323 modules
Hi,
what's the difference between h323 and oh323 modules? which one should I use?
Regards.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030722/3d3edb73/attachment.htm
2004 Dec 09
4
Get rid of H323 problems for 100$
Hello!
I see many of you experience troubles with H323 stack. I am focusing
on building H323-SIP Asterisk based softswitch with all codecs
supported (including G729 and G723).
I can setup Asterisk from scratch with H323 support or solve your h323
nightmare with existing asterisk system for for 100$.
Contact me pls offline.
2004 Jul 06
3
H323 channel
Hello everybody,
my * box is connected to gnugk with H323 channel. If I call from an H323
EP to SIP EP (GS HandyTone or Xlite), when callee is picking up, audio
start but noisy (scratch) , then became ok for callee (SIP EP) but still
scratching on H323 EP. Now I stop/start asterisk, call from SIP to H323
EP and it's ok. And from now, it's also ok when H323 EP call SIP one's!
No
2004 Aug 15
2
GrandStream ATA286 & RC2 (was RC2 - H323 channel broken)
Hello everybody,
when I upgraded from RC1 to RC2 I didn't had any audio between my ATA286
and H323 EP (my post from 13/08/04) I checked further and discover that
problem is with ATA286 who is unable to call. I always get an 404 error.
Coming back to RC1 everything works fine again. I tried to modify my
dtmfmode from rfc2833 to info but in change nothing. Local call to
asterisk are
2004 Jun 29
1
Registration of H323 Endpoints?
Hi,
I am using the asterisk-oh323 wrapper and I am looking to allow
registration of h323 endpoints and allow Asterisk to act as a gateway. The
idea is simple: H323 endpoints would register with Asterisk. They each would
have their own internal extension (like SIP). If a H323 endpoint dials an
outbound extension, then the h323 call gets routed to a H323 Gatekeeper which
then terminates
2005 Jan 23
2
sip - h323 translation stability & capacity limit
Hi! All
I would appreciate if someone could advice me on how stable is sip-h323 & h323-sip translation as well as how many calls can it handle when doing such translation.( assuming single 2.8Ghz intel processor & 1GB RAM)
Regards,
John
--
___________________________________________________________
Sign-up for Ads Free at Mail.com
http://promo.mail.com/adsfreejump.htm