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: lists.digium.com/pipermail/asterisk-users/attachments/20030429/fffec1d1/attachment.htm
The example config is with the source and will get installed in /etc/asterisk/ when you do "cd /usr/src/asterisk/channels/h323; make install samples" if you installed asterisk in /usr/src/asterisk Martin On Tue, 29 Apr 2003, Darran Wilson wrote:> Is H323 built into the current CVS? If so, could someone give me an idea of a simple config? > > > > > thanks, > > darran >
I want to use Asterisk with H323. What must I do exactly? I need help because I'm not sure of having understood everything about the H323 channel, the OpenH323 , the libraries and all those things I have read. Thanks a lot. Maria deory -------------- next part -------------- An HTML attachment was scrubbed... URL: lists.digium.com/pipermail/asterisk-users/attachments/20030520/596cbae7/attachment.htm
----- Original Message ----- From: "Pawe? Go?aszewski" <blues@ds.pg.gda.pl> To: <asterisk-users@lists.digium.com> Sent: Tuesday, May 20, 2003 6:20 PM Subject: Re: [Asterisk-Users] H323> On Tue, 20 May 2003, Mar?a de Ory wrote: > > what does RTFM mean? > > Read The F****** Manual :) > > F**** has many meanings :) > > -- OK! THANKS!Mdory> pozdr. Pawe? Go?aszewski > --------------------------------- > worth to see: againsttcpa.com > CPU not found - software emulation... > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > lists.digium.com/mailman/listinfo/asterisk-users >
RTFM = read the f____ manual They get asked questions that revolve around this problem repeatedly and are a little annoyed by it The H323 docs/site refer to where to download the libs from (and are the correct version) Mar?a de Ory (mdeory@avproduc.com) wrote:> >what does RTFM mean? Now I'm a little lost, but I'll read more, and then I'll tryto do what you say.>Thanks everybody! > ----- Original Message ----- > From: Jeremy McNamara > To: asterisk-users@lists.digium.com > Sent: Tuesday, May 20, 2003 6:06 PM > Subject: Re: [Asterisk-Users] H323 > > > You CANNOT use distro specific installs of Open H.323 and PWLib!!!!!!!!!!!!Distro specific installs change the names of the library files and have even been known to change code (debian).> > Do yourself a favor and RTFM, especially the Common Errors section: > > openh323.org/build.html#unix > > > Jeremy McNamara > > > > > Pawe? Go?aszewski wrote: > >On Tue, 20 May 2003, Jeremy McNamara wrote: > First cd asterisk/channels/h323 and eyeball the README. Then you need >to spend some time reading: openh323.org. > >It all boils down to properly compiling PWLib, Open H.323 and compiling >and installing chan_h323 > >...and if you want to use openh323 and pwlib from your favourite distro - >take this patch (made against latest cvs of asterisk) :) >I'm using it succesfully with: ># rpm -q openh323 pwlib >openh323-1.10.4-1 >pwlib-1.4.4-1 >from PLD Linux. > >openh323-1.11.* does not work for me. asterisk build with that openh323 >makes sig11 on start (yes, I've tried to do everything like in README of >chan_h323). > > ---------------------------------------------------------------------------- >--- ./channels/h323/Makefile.org Tue May 13 09:53:16 2003 >+++ ./channels/h323/Makefile Tue May 13 09:55:26 2003 >@@ -61,13 +62,13 @@ > g++ -g -c -o $@ $(CFLAGS) $< > > chan_h323.so: chan_h323.o ast_h323.o >- g++ -shared -Xlinker -x -o chan_h323.so chan_h323.o ast_h323.o -L$(PWLIBDIR)/lib-lpt_linux_x86_r -L$(OPENH323DIR)/lib -lh323_linux_x86_r -L/usr/lib -lpthread -ldl -lcrypto -lssl -lexpat>+ g++ -shared -Xlinker -x -o chan_h323.so chan_h323.o ast_h323.o -L$(PWLIBDIR)/lib-lpt -L$(OPENH323DIR)/lib -lopenh323 -L/usr/lib -lpthread -ldl -lcrypto -lssl -lexpat> > chan_h323_d.so: chan_h323.o ast_h323.o >- g++ -shared -Xlinker -x -o chan_h323.so chan_h323.o ast_h323.o -L$(PWLIBDIR)/lib-lpt_linux_x86_d -L$(OPENH323DIR)/lib -lh323_linux_x86_d -L/usr/lib -lpthread -ldl -lcrypto -lssl -lexpat>+ g++ -shared -Xlinker -x -o chan_h323.so chan_h323.o ast_h323.o -L$(PWLIBDIR)/lib-lpt -L$(OPENH323DIR)/lib -lopenh323 -L/usr/lib -lpthread -ldl -lcrypto -lssl -lexpat> > chan_h323_s.so: chan_h323.o ast_h323.o >- g++ -shared -Xlinker -x -o chan_h323.so chan_h323.o ast_h323.o -L$(PWLIBDIR)/lib-lpt_linux_x86_r_s -L$(OPENH323DIR)/lib -lh323_linux_x86_r_s -L/usr/lib -lpthread -ldl -lcrypto -lssl -lexpat>+ g++ -shared -Xlinker -x -o chan_h323.so chan_h323.o ast_h323.o -L$(PWLIBDIR)/lib-lpt -L$(OPENH323DIR)/lib -lopenh323 -L/usr/lib -lpthread -ldl -lcrypto -lssl -lexpat> clean: > rm -f *.o *.so core.* > > >-- Brian Johnson This is where my witty signature line would be if I bothered to edit this line :)
Hello, Could somebody point me to some example of h323.conf file where I can set up a gateway with the cisco router 2600 using h323 protocol. The phones are sip phone so Asterisk has to make translation from sip to h323. Can somebody point me somewhere ? Thanks very much. Bartosz Jozwiak -------------- next part -------------- An HTML attachment was scrubbed... URL: lists.digium.com/pipermail/asterisk-users/attachments/20030929/f151bc65/attachment.htm
look into /usr/src/asterisk/channels/h323 and FOLLOW instructions EXACTLY as in README. Senad -------------- next part -------------- An HTML attachment was scrubbed... URL: lists.digium.com/pipermail/asterisk-users/attachments/20030929/b3c53cce/attachment.htm
Hi, I tried to add h323 support for asterisk, but I got some troubles compiling the module. Is the latest version of Open H.323 and PWLib (the ones avalaible for download) compatible with a pretty recent asterisk cvs source ( Asterisk CVS-04/08/04-08:33:18) ? Tnx for any help ! -- Best regards, Alessio mailto:afoc@interconnessioni.it
Alessio- Whoops, after writing the reply to your message (see my other message), I just read Michael Manousos message about OH323. Looks like he has fixed the parameter call mismatches in the code, so you could try OH323 instead. Cheers Scott Scott M. Stingel President, Emerging Voice Technology, Inc. Palo Alto California & London England evtmedia.com -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Alessio Focardi Sent: Monday, April 26, 2004 3:52 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] H323 Hi, I tried to add h323 support for asterisk, but I got some troubles compiling the module. Is the latest version of Open H.323 and PWLib (the ones avalaible for download) compatible with a pretty recent asterisk cvs source ( Asterisk CVS-04/08/04-08:33:18) ? Tnx for any help ! -- Best regards, Alessio mailto:afoc@interconnessioni.it _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: lists.digium.com/mailman/listinfo/asterisk-users
I am having a similar problem with one-way audio from an Avaya hardphone calling a SIP soft phone. Audio from the hardphone is heard on the receiving end (SIP), but audio is not heard on the hardphone. I know audio is being injected into the sound card and being processed by the SIP client (I've tried both X-Lite and Windows Messenger 4.7.2009) because the audio meters show signal coming into the client however nothing is heard on the other end. I am using the following: CVS-HEAD 5/21/04 Pwlib-1.5.2 Openh323-1.12.2 Regards, Andy.> -----Original Message----- > From: asterisk-users-admin@lists.digium.com > [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of T. Chan > Sent: Tuesday, June 01, 2004 1:25 PM > To: Dmitry Mishchenko; asterisk-users@lists.digium.com > Subject: RE: [Asterisk-Users] RE: H323 > > Dear All, > > Thanks, but I was already using a pre May 25 CVS version. > Does anyone else > have any idea please? Thanks > > TC > > -----Original Message----- > From: Dmitry Mishchenko [mailto:arkadia@odessa.net] > Sent: Tuesday, June 01, 2004 6:22 AM > To: asterisk-users@lists.digium.com; T. Chan > Subject: Re: [Asterisk-Users] RE: H323 > > > On Tuesday 01 June 2004 00:56, T. Chan wrote: > > Dear All, > > > > I have used Asterisk for a few months and I have been using > a January CVS > > version, it has been working but has been regularly crashing. I use > > Asterisk mostly as a softswitch application receiving H323 > calls from > > customers and send to H323 carriers. Since I have been > using an older CVS > > version, the OpenH323 and Pwlib libraries in use have been > 1.11.7 and > > 1.4.11 > > respectively. > > > > I was thinking of using a current asterisk version and see > if it is more > > stable comparing to the version in use. I upgraded to new > version, and I > > understand that with the new version and the H323 code, I > need to use the > > 1.12.2 and 1.5.2 versions of the OpenH323 and Pwlib libraries > respectively. > > I have, therefore, erased the whole Pwlib and Openh323 > folders, recreated > > with the new versions and did the ./configure.....make > clean.....make opt > > procedures to compile the drivers. > > > > I have then compiled all the zaptel, libpri, asterisk as > usual, but when I > > ran the asterisk, I noticed that most calls (calls mostly > were sent from > > Cisco AS5300 and Cisco AS5350) were getting one way audio, > the calling > > party was not able to hear anything even the call was > connected, I am not > > sure if the called party would hear anything, but obviously > something is > > not working properly. > > > > I have not exactly the same but rather similar issue with the latest > cvs-head. > There are recent changes in call of on_start_logical_channel() > After moving it to > MyH323_ExternalRTPChannel::OnReceivedAckPDU it stopped > being called in my configuration. As a result I don't get any > audio after > call established. And with older approach when > on_start_logical_channel was > called at MyH323Connection::OnStartLogicalChannel it was working fine. > This change was done on May 25 so you may try to use older > code from CVS > before this date. > Jeremy saying the latest version approach is fine, but its > not working for > me :(. > > Dmitry > > > Can any of your experts out there help please, thanks? > > > > TC > > --- > > Outgoing mail is certified Virus Free. > > Checked by AVG anti-virus system (grisoft.com). > > Version: 6.0.687 / Virus Database: 448 - Release Date: 5/16/2004 > > > > _______________________________________________ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > > lists.digium.com/mailman/listinfo/asterisk-users > > > --- > Incoming mail is certified Virus Free. > Checked by AVG anti-virus system (grisoft.com). > Version: 6.0.687 / Virus Database: 448 - Release Date: 5/16/2004 > > --- > Outgoing mail is certified Virus Free. > Checked by AVG anti-virus system (grisoft.com). > Version: 6.0.687 / Virus Database: 448 - Release Date: 5/16/2004 > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > lists.digium.com/mailman/listinfo/asterisk-users > >
Good day all I have a asterisk server running sip and sip phone How do I get asterisk to call another h323 server? Please Help Thanks Altus
Good day all Can asterisk connect h323 clients to each other and h323 to sip and what about h323 video? Please Help and advice
Altus, Yes, Asterisk can do the following scenarios, amongst others: Client <-- H.323 --> Asterisk <-- H.323 --> Client Client <-- H.323 --> Asterisk <-- SIP --> Client In these scenarios, it is acting as a Back To Back User Agent (BTBUA). It can also handle video calls, though I have not used this myself. Alistair Cunningham, Integrics Ltd, Telephony, Database, Unix consulting worldwide +44 (0)7870 699 479 integrics.com Altus Snyman wrote:> Good day all > Can asterisk connect h323 clients to each other and h323 to sip and what > about h323 video? > Please Help and advice > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > lists.digium.com/mailman/listinfo/asterisk-users > >
Hello all i'm build success H322 in channel/H323 of asterisk. but don't know how to use it. i run GNUGK on server and client using ohphone. when i dial to asterisk server. the connection accept and disconnect. please help me to configure in H323.conf and extensions.conf. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: lists.digium.com/pipermail/asterisk-users/attachments/20050509/90e914d4/attachment.htm
Hello! How can I check if oh323 is loaded and working? Is there a quick test for this? Thank you. Micko
CLI> show module and look for chan_oh323.so If oh323 is loaded, "oh323 show conf" will provide more useful info. On 5/17/05, Micko <micko@voljatel.si> wrote:> Hello! > > How can I check if oh323 is loaded and working? Is there a quick test for this? > > Thank you. > > Micko > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > lists.digium.com/mailman/listinfo/asterisk-users >
Can anybody give me a hint, what I am doing wrong, please? Asterisk and H.323 1. download all parts --------------------- wget inaccessnetworks.com/asterisk-oh323/download/asterisk-oh323-0.7.1.tar.gz wget inaccessnetworks.com/ian/asterisk-oh323/Libraries/openh323_1.12.2.tar.gz wget inaccessnetworks.com/ian/asterisk-oh323/Libraries/pwlib_1.5.2.tar.gz wget inaccessnetworks.com/ian/asterisk-oh323/Libraries/ohphone_1.4.1.tar.gz wget inaccessnetworks.com/ian/asterisk-oh323/Libraries/openh323-Janus_patch4-src-tar.gz wget inaccessnetworks.com/ian/asterisk-oh323/Libraries/pwlib-Janus_patch4-src-tar.gz and untar it into /usr/src/ 2. Compiling ------------ cd /usr/src/pwlib pwlib$ ./configure pwlib$ make clean; make opt cd /usr/src/openh323 openh323$ patch -p1 < /usr/src/asterisk-oh323-0.7.1/openh323_1.13.5-make.patch openh323$ ./configure openh323$ make clean; make opt 3. Download / update Asterisk ----------------------------- cd /usr/src export CVSROOT=:pserver:anoncvs@cvs.digium.com:/usr/cvsroot cvs login - the password is anoncvs. cvs checkout zaptel libpri asterisk asterisk-addons asterisk-sounds go into each directory and: make clean; make update; make install Then, edit "Makefile" inside the "asterisk-oh323-x.x.x" directory and set the paths/options according to your system: DESTDIR=/ PWLIBDIR=/usr/src/pwlib OPENH323DIR=/usr/src/openh323 ASTERISKINCDIR=/usr/src/asterisk/include ASTERISKMODDIR=/usr/lib/asterisk/modules ASTERISKETCDIR=/etc/asterisk OH323WRAPLIBDIR=/usr/local/lib SSLINCDIR=/usr/include/openssl SSLLIBDIR=/usr/lib Type "make" to build the oh323wrap library and the ASTERISK OH323 channel driver. [root@GVSCathome asterisk-oh323-0.7.1]# make for x in wrapper asterisk-driver; do make -C $x build || exit 1 ; done /usr/src/openh323/openh323u.mak:296: warning: overriding commands for target `ccflags' /usr/src/openh323/openh323u.mak:293: warning: ignoring old commands for target `ccflags' /usr/src/openh323/openh323u.mak:296: warning: overriding commands for target `ccflags' /usr/src/openh323/openh323u.mak:293: warning: ignoring old commands for target `ccflags' make[1]: Entering directory `/usr/src/asterisk-oh323-0.7.1/wrapper' ./check_ver /usr/src/pwlib pwlib ./check_ver /usr/src/openh323 openh323 g++ -DP_LINUX=2.6.5-1.358smp -ffunction-sections -fdata-sections -D_REENTRANT -Wall -fPIC -DP_USE_PRAGMA -DPHAS_TEMPLATES -I/usr/src/pwlib/include/ptlib/unix -I/usr/include/pwlib -I/usr/src/pwlib/include -DPTRACING -I/usr/src/openh323/include -DHAS_IXJ -DHAS_OSS -Wall -x c++ -Os -DWRAPTRACING -DWRAPTRACING_LEVEL=5 -DPWLIBVERSION=\"1.5.2\" -DOPENH323VERSION=\"1.12.2\" -I/usr/src/pwlib/include/ptlib/unix -I/usr/src/pwlib/include -I/usr/src/openh323/include -I/usr/src/openh323/include/openh323 -I../asterisk-driver -c asteriskaudio.cxx -o asteriskaudio.o asteriskaudio.cxx: In destructor `virtual PAsteriskSoundChannel::~PAsteriskSoundChannel()': asteriskaudio.cxx:167: error: `baseChannel' undeclared (first use this function) asteriskaudio.cxx:167: error: (Each undeclared identifier is reported only once for each function it appears in.) make[1]: *** [asteriskaudio.o] Error 1 make[1]: Leaving directory `/usr/src/asterisk-oh323-0.7.1/wrapper' make: *** [subdirs_build] Error 1
Good day all Can I register asterisk as a h323 client,like in sip where you have register => -- Thanks Altus Snyman Stormcorp Network Solutions +27 11 8071141 exten 301
Yes you can. Try with oh323 module: lists.digium.com/pipermail/asterisk-users/2005-January/081881.html With this module you can register your asterisk with a gatekeeper. Regards. JSalas. -----Mensaje original----- De: altus [mailto:altus@stormcorp.co.za] Enviado el: Thursday, August 04, 2005 5:30 AM Para: asterisk Asunto: [Asterisk-Users] h323 Good day all Can I register asterisk as a h323 client,like in sip where you have register => -- Thanks Altus Snyman Stormcorp Network Solutions +27 11 8071141 exten 301 _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: lists.digium.com/mailman/listinfo/asterisk-users
>From wiki...(voip-info.org/tiki-index.php?page=Asterisk+oh323+channels) "The second option is valid only in the case where a gatekeeper is used. OH323 supports only one gatekeeper (or none, but not multiple gatekeepers). OH323 itself only acts as H.323 Gateway. " As I look, asterisk didn't act like gatekeeper. JS.>Yes, it worked here. > >part of oh323.conf example: > >. >. >. >;----------------------------------------- >; Configure H.323 aliases, prefixes and >; related ASTERISK's contexts >;----------------------------------------- >[register] >; >; Aliases/prefixes associated with the default context >; defined in section [general]. >; >;alias=asterisk >;alias=123 >; >; Aliases/prefixes routed in "all-aliases" context. >; >context=all-aliases >alias=asterisk >alias=99001701 >alias=99001702 >. >. >. > > This defines h.323 id and the aliases for each channel. > > So, now I would like to know if asterisk can support h.323 gateway >registration, like SIP. Can a h.323 gateway register on asterisk ? >Thanks > >-- > >[ ]'s > >Daniel Varella de Oliveira >Tecnologia IP Ltda >Tel.: +55 (21)2495-0936 / r. 108 >tecnologiaip.com.br > > >On Thursday 04 August 2005 10:54, Juan Salas wrote: >> Yes you can. >> Try with oh323 module: >> >> lists.digium.com/pipermail/asterisk-users/2005-January/081881.html >> >> With this module you can register your asterisk with a gatekeeper. >> >> Regards. >> >> JSalas. >> >> >>> -----Mensaje original----- >>> De: altus [mailto:altus@stormcorp.co.za] >>> Enviado el: Thursday, August 04, 2005 5:30 AM >>> Para: asterisk >>> Asunto: [Asterisk-Users] h323 >>> >>> >>> Good day all >>> Can I register asterisk as a h323 client,like in sip where you have >>> register =>>>>-----Mensaje original----- >>>De: Daniel Varella de Oliveira [mailto:dvarella@tecnologiaip.com.br] >>>Enviado el: Thursday, August 04, 2005 10:42 AM >>>Para: Asterisk Users Mailing List - Non-Commercial Discussion >>>Asunto: Re: [Asterisk-Users] h323_______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: lists.digium.com/mailman/listinfo/asterisk-users
Good day all Im trying to get asterisk and oh323 to work I following the instruction on lists.digium.com/pipermail/asterisk-users/2005- January/081651.html It on fedora core 1,and I downloaded the lated dev. of asterisk Installation: tar -zxvf asterisk-oh323-0.7.1.tar.gz tar -zxvf pwlib-Janus_patch4-src-tar.gz tar -zxvf openh323-Janus_patch4-src-tar.gz cd pwlib ./configure make cd openh323 patch -p1 < /root/asterisk-oh323-0.7.1/openh323_1.13.5-make.patch (pach to openh323) cd openh323 ./configure make opt but at make opt I get this error g++: Internal error: Illegal instruction (program cc1plus) Please submit a full bug report. See <URL:bugzilla.redhat.com/bugzilla> for instructions. make[1]: *** [/root/openh323/lib/obj_linux_x86_r/h323rtp.o] Error 1 make[1]: Leaving directory `/root/openh323/src' make: *** [opt] Error 2 Can someone please help Thanks Altus -- Thanks Altus Snyman Stormcorp Network Solutions +27 11 8071141 exten 301
Good day all How do I get h323 and video working? -- Thanks Altus Snyman Stormcorp Network Solutions +27 11 8071141 exten 301
With a liberal application of RFTW altus wrote:> Good day all > How do I get h323 and video working?-- Mark, G7LTT/KC2ENI Randolph, NJ g7ltt.com
RFTW or RTFM On Wed, 2005-08-10 at 09:36 -0400, Mark Phillips wrote:> With a liberal application of RFTW > > > > altus wrote: > > Good day all > > How do I get h323 and video working? >-- Thanks Altus Snyman Stormcorp Network Solutions +27 11 8071141 exten 301
Hi all, i need to install chan_h323, just a question: to use it i need the CVS ? Is there a version that i could use with asterisk 1.0.9 ? Thanks Giordano -------------- next part -------------- An HTML attachment was scrubbed... URL: lists.digium.com/pipermail/asterisk-users/attachments/20060124/236a7ef7/attachment.htm
Hi What is the best solution for H323 in asterisk -- h323 in source, -- oh323 or -- ooh323c? which is most robust and reliable? Which supports gatekeeper functionality? Best wishes Andrutto ---------------------------------------------------------------------- Najnowsze fakty!!! >>> link.interia.pl/f1996
Hi, i used to work ooh323 with my asterisk. it gives better performance than other oh323 or H323 comes with asterisk... i got H323 channel and oh323 with a lot of error.( like codec selection )....but ooh323 works fine with me........ thanks atik On 26 Aug 2006 12:13:52 +0200, andrutto <andrutto@poczta.fm> wrote:> > Hi > > What is the best solution for H323 in asterisk > -- h323 in source, > -- oh323 or > -- ooh323c? > > which is most robust and reliable? Which supports gatekeeper functionality? > > Best wishes > > Andrutto > > ---------------------------------------------------------------------- > Najnowsze fakty!!! >>> link.interia.pl/f1996 > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > lists.digium.com/mailman/listinfo/asterisk-users >
i am also using ooh323 - it works fine on sjphone ekiga etc but i cant seem to get it to work with ms netmeeting On 8/26/06, atik khan <atik.khan@gmail.com> wrote:> > Hi, > > i used to work ooh323 with my asterisk. it gives better performance > than other oh323 or H323 comes with asterisk... > > i got H323 channel and oh323 with a lot of error.( like codec > selection )....but ooh323 works fine with me........ > > thanks > atik > > > On 26 Aug 2006 12:13:52 +0200, andrutto <andrutto@poczta.fm> wrote: > > > > Hi > > > > What is the best solution for H323 in asterisk > > -- h323 in source, > > -- oh323 or > > -- ooh323c? > > > > which is most robust and reliable? Which supports gatekeeper > functionality? > > > > Best wishes > > > > Andrutto > > > > ---------------------------------------------------------------------- > > Najnowsze fakty!!! >>> link.interia.pl/f1996 > > > > _______________________________________________ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > lists.digium.com/mailman/listinfo/asterisk-users > > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > lists.digium.com/mailman/listinfo/asterisk-users >-------------- next part -------------- An HTML attachment was scrubbed... URL: lists.digium.com/pipermail/asterisk-users/attachments/20060826/66b8b8d6/attachment.htm
any one try that with g723 codec? thanks Salaque On 8/27/06, Rosli Sukri <roslisukri@gmail.com> wrote:> i am also using ooh323 - it works fine on sjphone ekiga etc but i cant seem > to get it to work with ms netmeeting > > > On 8/26/06, atik khan < atik.khan@gmail.com> wrote: > > Hi, > > > > i used to work ooh323 with my asterisk. it gives better performance > > than other oh323 or H323 comes with asterisk... > > > > i got H323 channel and oh323 with a lot of error.( like codec > > selection )....but ooh323 works fine with me........ > > > > thanks > > atik > > > > > > On 26 Aug 2006 12:13:52 +0200, andrutto < andrutto@poczta.fm> wrote: > > > > > > Hi > > > > > > What is the best solution for H323 in asterisk > > > -- h323 in source, > > > -- oh323 or > > > -- ooh323c? > > > > > > which is most robust and reliable? Which supports gatekeeper > functionality? > > > > > > Best wishes > > > > > > Andrutto > > > > > > > ---------------------------------------------------------------------- > > > Najnowsze fakty!!! >>> link.interia.pl/f1996 > > > > > > _______________________________________________ > > > --Bandwidth and Colocation provided by Easynews.com -- > > > > > > asterisk-users mailing list > > > To UNSUBSCRIBE or update options visit: > > > > lists.digium.com/mailman/listinfo/asterisk-users > > > > > _______________________________________________ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > > lists.digium.com/mailman/listinfo/asterisk-users > > > > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > > lists.digium.com/mailman/listinfo/asterisk-users > > >-- Got my Gmail ?