similar to: Problems with OH323/codecs

Hi Every time I hangup on my AGI script Asterisk crashes if it is not running in console mode. (happens when using python and perl AGI scripts) I'm desparatly trying to get my employer to let me use Asterisk. So I must get this to work. I've posted about this before, I'm sorry, but I'm desperate. I'm running RedHat 9.0 (kernel 2.4.20-8 everything else updated) I'm

hello, i have problem with oh323 channel driver (tryied differnet versions). when i try to make call between oh323 - sip, oh323-isdn, oh323-capi asterisk crash with segmentation fault. Channel driver was compiled with pwlib 1.5.0 and openh323 1.12.0 libs. Does anybody know solution ? WrapH323Connection::WrapH323Connection: WrapH323Connection created. -- Executing Dial("H323:31119",

Hello Guys, first of all, I'm very new with asterisk. I'm trying to set it up. I've already compiled and installed Asterisk-1.0.7 Now I'm trying with asterisk-oh323 I've already installed pwlib, oh323 and I've already set the variables. Now, when I try to "make" asterisk-oh323 I receive this error messagge: for x in wrapper asterisk-driver; do make -C $x all ||

Have any body succesfully compiled the files in "asterisk-oh323-0.2.tar.gz" ? I have the following errors: +for x in wrapper asterisk-driver; do make -C $x all || exit 1 ; done make[1]: Entering directory `/root/asterisk-oh323/wrapper' g++ -Wall -mcpu=i586 -DP_LINUX -D_REENTRANT -DP_HAS_SEMAPHORES -DP_SSL -DP_PTHREADS -DPBYTE_ORDER=PLITTLE_ENDIAN -DPHAS_TEMPLATES -O3 -DNDEBUG

1st. - I'm from Poland, we don't have (yet, and hopefully forever) software patents. Is there any free g.729.1 implementation for asterisk? I want to use it for my private use (dialing into inet->PSTN gateway), and I don't want (now) to buy codec, as I don't know if I will be using this service in future (now I just want to test it). Any solutions? Maybe even

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 When connecting IAX (gnophone) to SIP (kphone) or other way Asterisk acts as a proxy, but when connecting SIP to SIP it works only as 'SIP registrar' forwarding SIP requests to client. Is it possible to make Asterisk work as a 'proxy' so that any incoming calls would be ade to Asterisk and then internally forwarded to receiver?

I got 2 'InterStar' boxes - by dsg. Is there any way to use it with Asterisk or any other OS software? What protocol is it using? -- Witold Kr?cicki (adasi) adasi [at] culm.net GPG key: 7AE20871 http://www.culm.net

Dear Mailinglist-User currently we`re working with an IP-PBX, based on Asterisk, with SIP, H.323 and ISDN-Capabilities. SIP and ISDN works fine, but H.323 not. In our first test, we started to connect Asterisk to an Cisco IOS-Gatekeeper with the "chan_oh323" (version 0.6.5). We successfully tested in/egress calls without any problems. But when we started to connect our Asterisk

When I try to make a call, I have this error: dial 06302@gatekeeper -- Executing Dial("OSS/dsp", "OH323/06302|20|tT") in new stack *CLI> WrapH323Connection::WrapH323Connection: WrapH323Connection created. -- Called 06302 WARNING[1272234688]: File chan_oss.c, Line 624 (oss_read): Error reading from sound device (If you're running 'artsd' then kill it):

Hi, I start to be a little confused so I am asking to the list. I want to make with * a gateway from PSTN to H323, and to send all incomings call to a predefined IP, which will treat the h323 calls. let's assume that all my incoming numbers starts with 00 here is my extensions [incoming] exten => s,1,Answer exten => _00.,1,Answer exten =>

Hi, I'm trying to interconnect sip and h323 endpoints using asterisk and asterisk crashes with segmentation fault whenever h323 connection needs to be established. It registers with gatekeeper ok though. Here are the symptoms. If the call initiated by SIP device, asterisk replies to it "Trying" and then silently crashes (it launched as asterisk -vvvvcd). In debug log I can see the

Hi! When I try to make a call I have these messages: dial 06302@gatekeeper -- Executing Dial("OSS/dsp", "OH323/06302|20|tT") in new stack *CLI> WrapH323Connection::WrapH323Connection: WrapH323Connection created. -- Called 06302 WARNING[1272234688]: File chan_oss.c, Line 624 (oss_read): Error reading from sound device (If you're running 'artsd' then

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

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

Hi, does anybody manage to get music-on-hold with inaccess oh323 driver? Statement like : exten => 10,1,Dial(OH323/xx,mt) works (dials the xx number) but no music is heared. Also, if I put 'r' (ringback) it doesn't work either. With chan_h323 I got this functionality but this driver had some other problems (call transfer don't work).... Thanx in advance, Victor...

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]:

Sorry forgot the subjsect line Newbie question, please excuse me for this one. If an admin adds and extension in the voicemail.conf file will asterisks read from .conf files dynamically? Or does the asterisks daemon need to be restarted? I guess this question pertains to all .conf files. Also is there support for MySQL? Thanks Mike Michael Kane To-Talk Communications LLC. 37 Sandusky Dr.

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! Anyone with some insight into the oh323 channel driver please shed some light on the code block below from wrapendpoint.cxx. When enabling trace on the channel driver i get this, for me, strange debug info: WrapH323EndPoint::OpenAudioChannel: Direction => PLAYER, Buffer => 320 WrapH323EndPoint::OpenAudioChannel: FrameSize 8, FrameTime 8, TimeUnits 8

Hello Guys, first of all, I'm very new with asterisk. I'm trying to set it up. I've already compiled and installed Asterisk-1.0.7 Now I'm trying with asterisk-oh323 I've already installed pwlib, oh323 and I've already set the variables. Now, when I try to "make" asterisk-oh323 I receive this error messagge: for x in wrapper asterisk-driver; do make -C $x all ||