similar to: Fw: H323

Hi All I have recently downloaded Asterisk and was so impressed I thought I would setup a home server and I went out and got myself a couple of cisco 7940's. (and a sipaura 3000!). thanks to various posts on this list and the voip-info site I have managed to get chan_sccp setup and working with the 7940's but the I tried to get the messages, services and softkeys working. It seems

Michael, Thanks a bunch, I downloaded from inaccessnetworks.com thinking that it is the latest :). Ok I will upgrade it. just for the record, following worked. exten => _87.,1,Dial(OH323/H323:${EXTEN:1}@16.52.153.206) Cheers Sathya Date: Fri, 28 Nov 2003 11:28:59 +0200 From: Michael Manousos <manousos@inaccessnetworks.com> Organization: inAccess Networks To:

Date: Fri, 10 Sep 2004 16:37:33 +0300 > From: Michael Manousos <manousos@inaccessnetworks.com> > Subject: Re: [Asterisk-Users] Problems with 0penh323 Channel Driver > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: <4141AE1D.3020403@inaccessnetworks.com> > Content-Type: text/plain; charset=us-ascii;

Hello all, This new version has more options (account code, AMA flags, lib tracing) in the config file and some improvements in the build process. The code is available from the usual location: http://www.inaccessnetworks.com/projects/asterisk-oh323 Regards, Michael Manousos

We are not having any luck with the E100p card here in Australia, it will work with a crossover cable to another device but will not talk to our Telco Telstra who probably have a weird implementation of an E1. Any suggestions on a replacement? Regards Mark McKibbin DCS Internet 64 Queen St Warragul Victoria 3820 Australia www.dcsi.net.au mark@team.dcsi.net.au Ph. 1300 665575 Fx. 1300 556595

Is H323 built into the current CVS? If so, could someone give me an idea of a simple config? thanks, darran -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030429/fffec1d1/attachment.htm

Hello Michael, Here is the BackTrace of the program which i forgot to attach BACKTRACE OF Asterisk -vvc #0 0x42074d60 in _int_realloc () from /lib/tls/libc.so.6 #1 0x420738c4 in realloc () from /lib/tls/libc.so.6 #2 0x47c7da89 in PAbstractArray::SetSize(int) () from /home/sip/pwlib/lib/libpt_linux_x86_r.so.1.5 #3 0x47c7cf4d in PContainer::SetMinSize(int) () from

Dunno why your phone isn't allowing you do negotiate g711u but I can tell you how to upgrade the firmware. I called them on Thursday for myself and they gave me the following tftp server address for which to program my phone. 4.3.153.50 Load this into your phone's tftp area and reboot it. It'll go out to the net and check the firmware revision and change it if required. I've done

swar sir, can u please unsubscribe me for your list b.regards jihad chalhoub --- asterisk-users-request@lists.digium.com wrote: > Send Asterisk-Users mailing list submissions to > asterisk-users@lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, > visit > > http://lists.digium.com/mailman/listinfo/asterisk-users > or, via email, send a message

Hello all, This new version of asterisk-oh323 has some bug fixes (in configuration file parsing), changes in configuration file parameters (an attempt to make it more readable!) an some minor improvements like in-band DTMF detection (based on code posted by Klaus-Peter Junghanns on this list) and better handling of RTP packets with multiple voice frames. The code is available from the usual

Hi, I notice that i don't have callerID in my Voimail when someone drop me a message from H323 Client. Is there a tip to have this CallerID ? Regards Rattana -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030804/af4203ca/attachment.htm

----- Original Message ----- From: <asterisk-users-request@lists.digium.com> To: <asterisk-users@lists.digium.com> Sent: Wednesday, December 01, 2004 7:07 AM Subject: Asterisk-Users Digest, Vol 5, Issue 6 > Send Asterisk-Users mailing list submissions to > asterisk-users@lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit >

----- Original Message ----- From: <asterisk-users-request@lists.digium.com> To: <asterisk-users@lists.digium.com> Sent: Saturday, September 13, 2003 7:55 PM Subject: Asterisk-Users digest, Vol 1 #1279 - 16 msgs > Send Asterisk-Users mailing list submissions to > asterisk-users@lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit >

Hello all, This new version (0.6.1) of asterisk-oh323 fixes the "one-way audio" problem of the previous release. Download from the usual location: http://www.inaccessnetworks.com/projects/asterisk-oh323 Regards, Michael.

Hi all, After having tested the SIP part (successful :-)) we are now testing the H323 part of Asterisk. The H323 channel is up and running (using NuFone Network's Open H.323 Channel Driver)) However when dialing to * using ohphone the call can not be set up / established /H323 debug enabled *CLI> == New H.323 Connection created. -- Received SETUP message...

Hello, I am pretty new with Asterisk and I am using it as an H323 gateway.I would like to keep the same h323-conf-id in the outgoing leg as in the incoming leg. So far I have only tried inaccessnetworks' oh323 module, but I think this is a more generic issue. My extensions.conf is pretty simple: [oh323_context] exten => _XXXXXXXXXX,1,Dial(oh323/${EXTEN},30,tr) So my question is the

All, I have a problem with H323 the call disconnects when answered. The debug shows -- Executing Dial("SIP/sj1-4ff7", "H323/0797617729") in new stack -- Called 0797617729 -- H323/0797617729 is ringing -- H323/0797617729 answered SIP/sj1-4ff7 == Spawn extension (default, 0797617729, 1) exited non-zero on 'SIP/sj1-4ff7' -- Executing

On Monday 03 February 2003 11:37, Michael Manousos wrote: > Hemant Kumar wrote: > > inline... > > > > Rattana BIV wrote: > >> Hi, > >> > >> There\'s anybody who knows how can I use h323 with Asterisk. > >> I install pwlib 1.3.11 and openh323 1.9.10 but I have warning > >> message and i can\'t lauch asterisk -vvvc. > >

i have had some problems with the H323 channel ... Other party not anwsering SIP 2 H323 bridge. the chan_oh323 solves the problem. Use it. (Even though it is quite complicated to install but READ the README file) Nahuel that should solve it!! Kido -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi List, I am facing the reverse problem as stated here.I am using ATA 186 to make and recieve call to * through OH323 driver. When I use G711 codec in the ATA to make call then then as soon as i dial an extension the * crashes with 'segmentation fault'. But the same scenerio works fine when i use 723 codec in the ATA .I can dial the number and extension very well/(I have 723 support in