similar to: OH323 and G729

Hello, I am facing some problems with OH323 1. I am using exten => 223,1,SetVar(OH323_OUTCODEC=gsm) or g729 exten => 223,2,Dial(OH323/x.x.x.x) but the call always go out in aLaw . 2. With OH323 I find one way audio . 3. How can I use ASTCC with OH323 , I have tried to use dialstrings like exten => 223,2,Dial(OH323/192.168.0.1/1234) to send the dialed number as 1234 , but it does

Hello I had this codec negotiation with oh323 call. i used G723 codec and the provider had G729 as first priority. In this situation what ever number i dial i used get "No one there to answer the call". As soon as i changed my codec to G729 the call went through but had other problems, which i got away by dowloading the latest code for oh323. Has anyone seen this problem? or it is

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've complied the OH323 .so successfully and can easily receive calls from my H323 gatekeeper (using 711u), however it seems that all outgoing calls are refused and I'm getting "reason 23 (Temporary failure)" as an error code which I can't find documented everywhere. My H323 gatekeeper needs a 001NXXNXXXXXX to dial out to the PSTN even if I'm in north america (Montreal)

Hi, I am using asterisk CVS 2004-06-16 with oh323-0.6.3a I have a strange problem if I start asterisk with oh323 loaded /usr/sbin/asterisk -vvvvvc once I am in the console and issue "restart now" or "reload" asterisk hangs and it not stoping or restarting at all, below is the console logging when it happens, as you can see it stucks on "Destroying any remaining

Hi, I having a problem compiling the wrapper for oh323. I am running Debian, kernel version 2.4.18-bf2.4. The pwlib version I have is 1.6.6 and the openh323 version I have is 1.13.5. I execute the following commands first before attempting to compile the wrapper: pwlib_1.6.6: make both openh323 1.13.5 ./configure make opt asterisk-oh323 0.6.2 make

Why is no one suggesting any solution here for this problem, which has been lingering for a while. Are we not supporting H.323 on Asterisk? -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com]On Behalf Of ruixun wu Sent: Thursday, July 22, 2004 4:06 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] still can't

Hi All, I have set up a box that will be used as follows: SIP Phone ----> Asterisk ----> Cisco H323 VoIP Server 192.168.1.5 192.168.1.50 192.168.1.80 Asterisk is running the latest CVS and oh323 driver. The SIP phone is a Grandstream Budgetone 100. I have everything setup and running with G.711 ALAW and ULAW and i'm able to make calls through

Hi, After a whole day of work, I finally complied oh323 0.6.3a successfully. But when I started asterisk, it cann't load oh323. Following is the error: [format_jpeg.so] => (JPEG (Joint Picture Experts Group) Image Format) == Registered format 'jpg' (JPEG (Joint Picture Experts Group)) [cdr_csv.so] => (Comma Separated Values CDR Backend) [chan_oh323.so]Jul 13 09:43:45

Hello list, I have many IP hardphone like Siemens 300 basic ( old ) , cisco ata.. etc I need: G711 from old phones must be convert to G729 via asterisk and send to provider I have this problem: oh323 (last version): ------------- asterisk work with this driver ok for old phones, if I only faststart=no . But problem with codec , asterisk can speak with provider ( G729 ) only if I disable

Hello list, I have many IP hardphones like Siemens 300 basic ( old ) , cisco ata.. etc I need: G711 from old phones must be convert to G729 via asterisk and send to provider ( G729 from digium ) I have this problems: oh323 (last version): ------------- asterisk work with this driver ok for old phones, if I only faststart=no . But problem with codec , asterisk can speak with provider (

HI ALL HI MICHAEL; My name is mohammad and I am iranian.I have been trying to install oh323 channel but I come up with dead end. In fact it makes me crazy. plz help me michael. I saw mailing list and I trid serevel CVS headers such as , 2004-06-07( seven of june) 0r 2004-07-02( second of july) besides I use: 1-openh323 v1.12.2 2-pwlib v1.5.2 3- asterisk CVS (2004-06-07,

Abwesenheitsnotiz: [Asterisk-Users] oh323 geting voice problem g729 xeon 2.8 , fedora 1 , asterisk 1.0.6Hi All I am using oh323 with 6.6 virsion , and runing under xeon 2.8 dual with 2 gb ram, with g729 for i686 , (fedora 1). my problem is sip - oh323 - h323 (quintum) - pstn , sip party can listen otherparty realtime voice , but other party geting sip party's voice 1 sec later (not

Hello, I managed to terminate calls from cisco: as5300 and 7206 to asterisk over h323. I tested both oh323 from inaccessnetwork and JerJers chan_h323. I used 1.12.2 version of oh323 and 1.5.2 version of pwlib. After latest changes from JerJer chan_h323.c works ok when receiving traffic from ciscos. I havnt found any audio problems although I didnt send much traffic. Latest oh323 has some

Hi. im compiling the wrapper for oh323(under Suse 9.0) -pwlib 1.6.6 -openh323 1.13.5. (with oh323 Patch) i execute: ./samples/simple/obj_linux_x86_r/simph323 and it works fine. When i Run asterisk-oh323 0.6.2: make I get the following errors: chan_oh323.c:660: error: `__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__' undeclared here (not in a function)

Hi everyone, Could somebody enlighten me on this one? I have configured my asterisk to run on oh323 using codec g729. Incoming calls are working okay. But the thing I want to work is say pressing some options, say dial 1 to go to voicemail or dial a certain number to dial a specific extension. I have a config for this and tried calling from a normal PSTN and is working. But i just can't seem

Hi list! I am able to successfully receive faxes with RxFax using OH323 (0.6.5) channels and G.711. On the other side I have a Quintum CMS. However, I don't know how to force the codec before I answer the calls, i.e., I would like to answer the voice calls with G.729 and the fax calls with G.711. Is there any way to do this on Asterisk's side ? Is there something similar to

Hello, I have something like this: SIPUSER <-sip-> ASTERISK <-oh323-> AUDIOCODEC <-e1-> PSTN After calling from SIP to PSTN (and from PSTN to SIP too) I can't hear anything only in my SIPUSER. At the PSTN side everything is OK. I have another network with another h323/sip (in the place of asterisk) and there everything is OK. In AUDIOCODES logs I see that everything goes

Hi there I am pretty close on giving up on Asterisk :-/ I am (still) trying to make a call from a H323 phone to an Asterisk provider using AIX. But H323 does not route the number to AIX. All it is transmitting is an "s". *CLI> -- Executing Dial("OH323/R27865", "IAX2/demo:demo@gw1.musimi.dk/s") in new stack -- Called demo:demo@gw1.musimi.dk/s Jul 2

Hi, erverybody The Asterisk is running well in the linux system. Now I would like to add oh323 in Asterisk. I have download pwlib(version is 1.6.6) and openh323(version is 1.13.5). And I sucessfully maked and installed these two packages. But I got the following errors when compling the asterisk-oh323 0.6.3a: for x in wrapper asterisk-driver; do make -C $x all || exit 1 ; done make[1]: