similar to: OH323 Rocks :) --- H323 guys, use it to solve no answer at this time problem!!!

hi all, i need an asterisk board that can support up to 30 FXS ports. i found the following on digium: TDM400P. It only has 4 port, does it mean that i have to buy 8 of those to have my 30 FXS line? Or is there any channel bank solution that digium or other providers can provide us with in order to do that. Experienced asterisk user, how do you manage to make PBXs for 30 to 100 regular phones?

Hello all, I found a LSI161SCREV2 Dialogic board in one of my drawers, and i was wondering if by any luck, i could make some magic happen with asterisk ... If asterisk does not support it, is there any PSTN to H323 or PSTN to SIP gateway that support this dialogic card and that can be connected to an Asterisk Box? Digium, I PROMISE that I will buy my cardf rom you once my tests are conclusive

Hello, I just downloaded and installed the latest version of asterisk under Fedora. (had it under FreeBSD but was having TOOO many problems) After my installation i noticed that the channel H323 was not included ( I remember that i did not have to install it under freeBSD) but I have seen that SIP and IAX are supported though. So i am wondering: Does asterisk consider H323 so achaic that it does

I can't seem to get music on hold working, it tries to work, but I just hear strange noises on the extension.. Here is some debug info. Looks like mpg123 starts fine, but I hear nothing. I'm on todays CVS build. -- Executing Answer("SIP/2203-062c", "") in new stack -- Executing MusicOnHold("SIP/2203-062c", "default") in new stack --

Hello, Is there an utility for asterisk for codec conversion? I tried google but i haven' got anything. I am trying to initiate a call with G711 codec to asterisk and i would like asterisk to call a gateway with an g729 codec, therefore making a codec conversion from g711 to g729. I know chan_oh323 does it by specifying the OUT_CODEC variable, but chan_h323 does not. And i was wondering is

hi when i try to run the make command for the oh323 wrapper below linux:/usr/local/src/asterisk/asterisk-oh323 # i get an error please help regards Barbra asteriskaudio.cxx:369: syntax error before `::' token asteriskaudio.cxx:374: syntax error before `::' token asteriskaudio.cxx:379: syntax error before `::' token asteriskaudio.cxx:384: syntax error before `::' token

Hello, Digium support kindly proposed to ship a TE120P card to help resolve the issue. I plugged in the card, and introduced the loopback plug. I cleared the red alarm for a while and then i started seeing alarm switching from Yel/Recovering to Blue/Rec with a lot of IRQ Misses. I call Digium that assisted us, and we noticed IRQ sharing with the VGA adapter and the Ethernet port. I changed

hi just after talking to Jeremy, I downloaded the source he's using and compiled. core dump! This was without gatekeeper: (gdb) bt #0 0x40141118 in mallopt () from /lib/libc.so.6 #1 0x401400fc in malloc () from /lib/libc.so.6 #2 0x405e981b in PAbstractArray::SetSize(int) () from /usr/src/pwlib/lib/libpt_linux_x86_r.so.1 #3 0x405ed006 in PString::SetSize(int) () from

hola soy nuevo usuario de R y necesito crear un objeto p tal que, p=lambda*integral[f´´(x)^2 dx ] donde "lambda" es uno de los parametros que resultan de la funcion "smooth.spline()" y la integral es sobre la derivada 2 de esa misma funcion... dos cuestiones: 1) como extraigo lambda de los resultados de smooth.spline() para usarlo como objeto cuando lo requiera y 2) como

Hi, anybody knows what's missing or problem why i cant compile asterisk-oh323 in my machine? i got this compiled successfully ...Openh323 - v1.12.2 ...pwlib - v1.5.2 except ...asterisk-oh323 - v0.6.5 here's the output as i run make... mkoy@sambag:~/voip/asterisk-oh323-0.6.5$ make for x in wrapper asterisk-driver; do make -C $x build || exit 1 ; done make[1]: Entering directory

Update on this problem: I gave up on the "native" h.323 because, like others, I couldn't get audio working. (yes, I tried disabling FastStart in ast_h323.cpp - no change) So I went and got the OH323 code from www.inaccessnetworks.com. Glad to say that everything seems to work so far. Not only does audio work, but even the handshaking is now working in both OpenPhone and even

i use asterisk cvs head ( two days ago) more or less openh323 1.12.2 (oh323 home page) and pwlib 1.5.2 (oh323 home page) asterisk-oh323-0.7.2-pre1 library versions? where download? versions from oh323 readme are not in sourceforge site. but i obtain this error compiling: root@backup:/usr/src/asterisk/cvs/last/asterisk-oh323-0.7.2-pre1# make for x in wrapper asterisk-driver; do make -C $x

Hello list I don't get to compile h323. I have the mistake: asteriskaudio.cxx: In destructor `virtual PAsteriskSoundChannel::~PAsteriskSoundChannel()': asteriskaudio.cxx:167: `baseChannel' undeclared (first use this function) asteriskaudio.cxx:167: (Each undeclared identifier is reported only once for each function it appears in.) make[1]: ** [asteriskaudio.o] Erro 1 make[1]:

Hi everyone, Could someone help me on make my Asterisk registers to a Gatekeeper. I have compiled the chan_h323.so and it seems to be working. What I want to know is how can I "route" my SIP clients to a single account on a remote Gatekeeper. I have tried a lot of conbinations but nothing happend. For example: my account number: 123456789 ;extension.conf exten =>

Hello, I am new to this list and to asterisk and going through the archive file I did not find an answer to my problem. I have a VoIP network working fine with multiple gateways registered to a Cisco H.323 Gatekeeper. I have successfully registered Asterisk as a GW in that network and also successfully registered two X-Lite SIP Client to asterisk that call to each other. I want to connect to

Hi all Greetings from Patagonia Argentina - this is my first message to this list. I''m having a weird problem: when I reboot some domU, some OTHER domU gets killed. Scenario: > cat /etc/SuSE-release SUSE Linux Enterprise Server 11 (x86_64) VERSION = 11 PATCHLEVEL = 1 > uname -sri Linux 2.6.32.43-0.4-xen x86_64 > rpm -qa | grep xen- kernel-xen-base-2.6.32.43-0.4.1

Hello all, Has anyone use the Supermicro PDSME+ in combination with the TE110P successfully? My experience so far is not very good. I am running trixbox 2.0 but: 1) with zttool I am getting IRQ Misses. Don't seem to have IRQ conflict, but I am now running my SATA HD in DMA. And I am not able to set it in DMA(HIO... operation not permitted) 2) With zttool the Alarm is RED 3) When I do the

Hi, Has anybody had any problems getting digium hardware lately? Regards Greg Cirino ___________________________________ Cirelle Enterprises Inc. 603-425-2221 www.cirelle.com Website Design www.cirelle.net ProSpeed High Speed Dial-up - 5 Times Faster www.cedata.com Web, FTP, Email Hosting Services www.mlsbot.com MLS IDX Services When You Want It Done Well, Just Call Cirelle It's not just a

I'm trying to use a Howlink CL-100 ip phone with * It's h323 phone with very limited protocol support. But it's enough that I can use it to dial netmeeting client and artisoft pbx just fine. When I try to dial my * with it using either chan_h323 or oh323, it seems to fail on negotiating H245. Maybe this phone doesn't support it? I've used all different versions of

Hello I am trying to compile asterisk-oh323-0.7.0 with pwlib-Janus_patch4 and openh323-Janus_patch4 downloaded from inaccessnetworks so I did this: tar -zxvf openh323-Janus_patch4-src-tar.gz cd openh323 patch -p1 < /root/asterisk-oh323-0.7.0/openh323_1.13.5-make.patch ./configure make opt cd asterisk-oh323-0.7.0 vi Makefile (to set the paths and options according to my system...) NOW I