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Hello to all, I have a strange behavior of my asterisk box. I'm running asterisk with asterisk-oh323 channel driver and everything works very well. But after few hours, my asterisk stop running and I have to restart it by typing "asterisk -vvvc". Most of the time I connect to my asterisk with a remote host so I don't know exactly which error causes my box to stop, but I found on

...:Main() () from /home/sip/openh323/lib/libh323_linux_x86_r.so.1.12 #7 0x47c756f1 in PThread::PX_ThreadStart(void*) () from /home/sip/pwlib/lib/libpt_linux_x86_r.so.1.5 #8 0x4002e332 in start_thread () from /lib/tls/libpthread.so.0 Rgds Sip Rtp ----- Original Message ----- From: "Michael Manousos" <manousos@inaccessnetworks.com> To: <asterisk-users@lists.digium.com> Sent: Friday, August 08, 2003 3:56 PM Subject: Re: [Asterisk-Users] Problem -ATA-711-723-Oh323-Asterisk > > Sip Rtp wrote: > > Hi List, > > > > I am facing the reverse problem as state...

...and which has the most functions implemented? Its a simple home environment that I am planning but it would be good to be able to use the softkeys to transfer calls and to pickup messages. Thanks in advance, Sam Kevin Walsh <kevin@cursor.biz> wrote on 27/08/2004 13:59:09: > Michael Manousos [manousos@inaccessnetworks.com] wrote: > > Kevin Walsh wrote: > > > Michael Manousos [manousos@inaccessnetworks.com] wrote: > > > > a) The transmitter detected silence and sent nothing but the last CN > > > > packet was lost. According to the above interpret...

...ictoria 3820 Australia www.dcsi.net.au mark@team.dcsi.net.au Ph. 1300 665575 Fx. 1300 556595 -----Original Message----- From: Simon J Mudd [mailto:sjmudd@pobox.com] Sent: Friday, 13 June 2003 7:10 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Asterisk logging questions manousos@inaccessnetworks.com (Michael Manousos) writes: > > While logger.conf indicates that you can log to a file I can't see > > _where_ > > the logging is sent to (which directory). Looking at the source it seems > > that the directory used is specified in ast_config_AST_L...

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

This module wont compile can anyone give me any assistance -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040614/03ae433c/attachment.htm

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.

I've merged some changes from Michael Manousos that should improve sound quality on non-zap channels, including music on hold. I'd like to hear back on or off list about your experiences with the new code. Thanks! Mark

Hi, I've found that neither Michael Manousos patch nor ztdummy driver do not fix musiconhold sound interruption problem up to acceptable quality level. Sound is choppy here anyway. It is my understanding (please correct me if I'm wrong) that if I have a Digium card in my asterisk machine, these problems should be gone 'cause those c...

...t; asterisk-users-admin@lists.digium.com > > When replying, please edit your Subject line so it > is more specific > than "Re: Contents of Asterisk-Users digest..." > > > Today's Topics: > > 1. Re: Asterisk-oh323 0.6.1 Compiling problem > (Michael Manousos) > 2. Re: IP local loop? (Steven Critchfield) > 3. Channelized T1, SIP phones, HW Echo Canceller > (Steve Creel) > 4. Re: Help with IAX , voice Distortion or > Breakage. (Alexey Ostrovsky) > 5. Re: Where to get 48 volt Power Supplies for > Cisco > IP Pho...

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

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

...t; To: <asterisk-users@lists.digium.com> > > Sent: Tuesday, May 20, 2003 5:10 PM > > Subject: Re: [Asterisk-Users] H323 > > > > > > > Thanks a lot, I'll start with it right now! > > > ----- Original Message ----- > > > From: "Michael Manousos" <manousos@inaccessnetworks.com> > > > To: <asterisk-users@lists.digium.com> > > > Sent: Tuesday, May 20, 2003 11:42 AM > > > Subject: Re: [Asterisk-Users] H323 > > > > > > > > > > Mar?a de Ory wrote: > > > > &gt...

Hi, Is it possible to use 2 B channels simultaneously with either I4L or CAPI drivers? We use AVM A1 (Fritz) PCMCIA with I4L driver and AVM B1 PCMCIA with CAPI driver. Thanks, Michael.

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

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: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Resend: Help for oh323 Reply-To: asterisk-users@lists.digium.com Hi Sathya, I bet you use OpenH323 v1.12.0. Go to v1.12.2 and you will be OK. There isn't a...

Due to the format of the message coming from the H323 channels included w/ Asterisk we were unable to use our gatekeeper. For a quick solution we tried the OH323 channel drivers and can receive inbound calls from the parent gatekeeper. We are trying to do a dial to gatekeeper... I am trying exten => 5551212,1,Wait,2 exten => 5551212,2,Dial,OH323/5551212 But I am not sure if this is the

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-...

Hi there. I thy to compile asterisk-oh323-0.6.3a but it fail in the make command. I have the pwlib-v1_6_6-1 and openh323-v1_13_5-1 as saying in the README file of the packet asterisk-oh323-0.6.3a I do make and this is the error: # make for x in wrapper asterisk-driver; do make -C $x all || exit 1 ; done make: *** No rule to make target `ccflags'. Stop. make: *** No rule to make target

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: