similar to: oh323 codec negotiation

Dear All, I have problem with new oh323 0.6.3a , I try use var OH323_OUTCODEC, but it don't work. oh323 driver don't want connect to gateway with g729, it's work if I only use in oh323.conf one codec ( g729 ). If I enable 2 or more codecs - always in use other codec: -- Executing SetVar("IAX2[4010@4010]/1", "OH323_OUTCODEC=g729a") in new stack -- Executing

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 am trying to configure my extensions.conf and oh323.conf to termination calls to a H323 service provider. Can anyone send me a sample config files? or tell me where to put the username and password which my service provider has given? also how to put the Dial command in extensions.conf Asterisk rocks. I have a SIP provider configured for all my international calls and it works absolutely

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)

Hello when i dial a toll free no using sipphone i get this error message. How do i solve this? Any help will be appreciated. console message: Starting simple switch on 'Zap/2-1' -- Executing SetCallerID("Zap/2-1", "17473863282") in new stack -- Executing SetCIDName("Zap/2-1", "Deepak JV") in new stack -- Executing

According to the wiki at voip-info.org, the dial structure for using oh323 without a gatekeeper is: OH323/<exten>@<host>:<port> or OH323/<exten> The second option is valid only in the case where a gatekeeper is used. NOTE: OpenH323 library v1.12.0 has a bug in the parsing of the destination host. When this version is used then the above syntax should be:

Hi, i have been using asterisk CVS 19/07/2005 and asterisk-oh323-0.7.2. I now want to use oh323 with Asterisk 1.2.4+. Can anyone tell me what versions of oh323(and pwlib and oh323) they got to work with Asterisk 1.2.4+. -- thanks, yusuf

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.

Joao, Thanks for sending the Installation tips as pasted below. It works. Seshu ---------- Get oh323 from http://www.inaccessnetworks.com/projects/asterisk-oh323/Libraries/openh323-Janus_patch4-src-tar.gz Get pwlib from http://www.inaccessnetworks.com/projects/asterisk-oh323/Libraries/pwlib-Janus_patch4-src-tar.gz Get asterisk-oh323 from

Hi all, Can someone share with me his experience in making asterisk-oh323 talk to quintum gateway without gatekeeper. My set up is QUINTUM GATEWAY ------IP----M ASTERISK (OH323) Both are gateways.. but I don't know what authentication I will set up in oh323.conf and how to set it up I will be glad if anyone can help Goksie

Hi all, I want to give a try to oh323 (currently nufone h323 channel is setup and compiling fine) on a yesterday CVS update of asterisk. I have _pwlib 1.8.1_ and _openh323 1.15.1_ What I made: openh323 dir: make clean apply the oh323 patch configure make opt asterisk-oh323-0.7 dir: make [...] wrapendpoint.cxx: In method `BOOL WrapH323EndPoint::OpenAudioChannel (H323Connection &, int,

I am experimenting with oh323 channels and h.323 gateways and a Cisco CallManager. I am not using a gatekeeper at this time. Is it possible to place calls coming into Asterisk from specific peers into specific contexts? In iax.conf eaxh peer has a context in which I can specify the context an inbound call will be placed in. I don't see anything like this in the oh323.conf file or the oh323

I have installed oh323 channel driver (finaly! :)). I head some problem starting * so I have put the smallest possible oh323.conf file to se what happens. When I don't put available codec's in oh323.conf (*1) Asterisk starts but he also disables h323 channel because there are no available codec's (*2). When I put codec (*3) Asterisk doesn't start (*4). What have I done wrong? I

I have just compiled & installed the latest oh323, on a fresh asterisk installation however using a previously working oh323.conf file. When I try to dial an outbound oh323 call I get the following error : -- Going to extension s|1 because of immediate=yes -- Executing Wait("Zap/1-1", "1") in new stack -- Accepting call from '21382890' to 's'

I am blocked on the most simple step when compiling Oh323: Cd /openh323 patch -p1 ../asterisk-oh323-0.7.2-pre1/openh323_1.13.5-make.patch it hangs for ever and never finishes in my machine with Red Hat Enterprise Linux 3, fully updated. I need to use Asterisk HEAD and I therefore I want to use asterisk-oh323-0.7.2-pre1. Additionally, because it is the most debugged version. In the past, if

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

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 ||

Hi, what's the difference between h323 and oh323 modules? which one should I use? Regards. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030722/3d3edb73/attachment.htm

I'm having a small issue with the oh323 implementation when it comes to codec selection. Version info: CVS Head 6/30/2004 OH323 0.6.3 OpenPhone for windows version 1.8.1 Asterisk is configured as a h323 endpoint which either terminates to the PSTN locally through a PRI or terminates the h323 call to an IAX provider remotely. Asterisk also has G729 licences installed. in oh323.conf we

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,