similar to: H323 call dropped when answered

All, I have a problem with H323 the call disconnects when answered. The debug shows -- Executing Dial("SIP/sj1-4ff7", "H323/0797617729") in new stack -- Called 0797617729 -- H323/0797617729 is ringing -- H323/0797617729 answered SIP/sj1-4ff7 == Spawn extension (default, 0797617729, 1) exited non-zero on 'SIP/sj1-4ff7' -- Executing

I have compiled chan_oh323 and when starting * I get the following. [chan_oh323.so]Aug 15 12:40:00 WARNING[1076245120]: loader.c:242 ast_load_resource: /usr/lib/asterisk/modules/chan_oh323.so: undefined symbol: __use_ast_pthread_create_instead__ Aug 15 12:40:00 WARNING[1076245120]: loader.c:423 load_modules: Loading module chan_oh323.so failed! Can anyone tell me how to fix this, or what

Colleagues- I am working on the design of a fairly large samba deployment, and I am looking for feedback on some of my design ideas. I have 10 buildings spread out in and around a city, all interconnected via 1.5Mb leased lines. There are samba servers in each building. I have some users that move from building to building. We are using primarily windows 98 desktops, with a few 2K and XPP

Hi All I'm trying with no luck to connected the Quintum D series Gateway with the new SIP release to asterisk. Have anyone done this? If yes then how should I configure the sip.conf to accept the registration? maybe a sample config? Thanks /Krystian

Hi all I have asked this question before but have not got any helping input. I'm really new to this and need some explanation about ASTCC. So here is the question again. In the ASTCC web admin there are Trunks, Routes, IAXFriends, SIPFriends, Brands, Cards. As I understand Brands is not used, Cards just makes the cards. Routed in the dialplan and pricelist, Trunks is for ASTCC to

Hi all, I have VOIP network built with MGCP endpoints.The manufacturer of endpoints is ASKEY. I downloaded latest Asterisk software and found it very useful for me. I configured it and it seems taht everything works OK when I am testing it with one or two endpoints. After that I tried to move Asterisk to working network and replace existing call manager. It starts working and calls are

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 All, I am trying to setup a SIP to H323 system using SER, Asterisk And GnuGK. Following is the configuration: CISCO ATA (NAT) -> SER -> ASTERISK -> GNUGK My Cisco ATA is registered with SER and When I dial a number, SER forwards the call to Asterisk, and Asterisk forwards the call to the GateKeper. This is ok, call reaches the gatekeeper, however the gatekeeper drops the call

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 .. i've just installed and licensed an instance of the G729 codec. I am trying to connect through asterisk to Go2Call server .. According to their info it involves dialling extension 729 on voip01.go2call.com, to get the IVR. my extensions.conf shows : exten => s,2,Dial(OH323/h323:729@216.52.153.206) which I think is correct, I have G729 enabled in the OH323.conf file and it seems to

Hi Michael and Everyone I'm trying to connect Asterisk to a CISCO AS5350 using oh323 and I'm getting this error "reason 24 (Call ended with Q.931 cause)" I've checked the Asterisk wiki and several other resources. Please can anyone give me a hint on what the problem is I reach my wits end. Thanks Tola my config and debug Configuration of OpenH323 channel driver

Hi all, i've installed the a new pwlib (1.5.0) / oh323lib (1.12.0) on my *. Then i've installed the new chan_oh323 (0.5.6). when i try to make a call with "netmeeting" through * ( * dial out with "Dial,OH323/${EXTEN}@xx.xxx.xxx.xx" ) the call will be blocked. Before, there was chan_oh323 0.5.5 and pwlib(1.4.11) and openh323(1.11.7) installed, and it worked. Is here

I'm just curious if anyone has the X100P & Callerid receiving working outside US. Replies are appreciated. Also if it's not working for you in a certain coutry you can respond too. regards Martin

Hi Friends, Hope you would help me out here, I have searched the asterisk user list for hours and also read the readme and test files that comes with the driver. I need a very simple scenario. I have SIP clients and want to use oh323 to dial out to PSTN using a h323 gateway. a)If I set the extention.conf like this: exten => _87.,1,Dial(OH323/16.52.153.206) oh323 dials out (I can ring a

i am trying to make calls between two workstations using netmeeting and asterisk. i get the popup on both when i call the extensions 665 and 667 but when accept, i get this error *CLI> 0:18.190 H225 Caller:8112978 H225 Received connect PDU. 0:18.288 H245:810b388 H245 Read error: Bad file descriptor 0:18.318 H323 Cleaner H323

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 =>

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.

Hi, I'm using a Cisco 7960 with a SIP load, and a Cisco 2600 router with an FXO port. Asterisk talks to the router via h323 and opens a call and connects with no problem. At exactly 74 secs (timer on the phone) the call drops, and Asterisks displays this message: -- H323:29764 answered SIP/6000-9794 15:20.606 H225 Caller:80eea08 H225 Received connect PDU.

Hi All I seem to have a small problem with the music on hold button on SJPhone. I have 2 asterisk installations one from the Rapid distribution and one from the latest CVS. On the rapid dist when I press the music on hold button on my SJPhone I get music on hold. When I do the same I get no music on hold just silence. I create extension like this exten => 1111,1,MusicOnHold(Default),

Trying to configure voicemail with H.323 all I get is the following errors when I call 123, 666, 665, 664 or 031. I'm a newbie at this so, I think it might be a simple fix. [chan_oh323.so] => (OpenH323 Channel Driver) == Parsing '/etc/asterisk/oh323.conf': Found 0:00.004 OpenH323 Wrapper OpenH323 Wrapper Version 0.0alpha0 by inAccess Networks