similar to: OH323 + G729 + Go2Call

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

Hello, I've been trying to send traffic to a Cisco Call Manager 3.2, but with no luck. Here's whats happening: * Call gets to CCM * Call gets to the gateway * Rings a couple times on destiny * Call gets hungup. On the CCM I get the following error: MediaManager - ERROR wait_AuConnectErrorInd On the Gw (Cisco AS5300) I get a disconnect cause of 2F (Resource not available) On asterisk:

I'm attempting to get chan_h323 working on Asterisk CVS-STABLE. I've compiled it ok using the Janus release of pwlib/openh323, by editing the makefile as per the comments. Call setup and cleardown seems to work fine, but no audio is being passed in either direction. Doing an "h.323 trace 9", I noticed the following sequence at the end of the call setup: h323.cxx(1685)

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

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

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

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, I use the oh323 driver to answer H323 calls. The connection is set up normally. In my extensions.conf file I use: exten => s,1,Answer exten => s,2,Playback(demo-instruct) exten => s,3,Hangup So that when a call is answered i get: *CLI> -- Executing Answer("H323/ip$10.0.3.23:32782/6502", "") in new stack -- Executing

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

Greetings All- I have an * box with the NuFone H.323 channel driver installed. I also have an Altigen VoIP system with a PRI to the PSTN. I can sucessfully make a call from a SIP extension (snom190) to an H.323 extension (altigen phone) The thing I can't seem to make work is a call from a H.323 phone to a SIP extension. Here's the layout:

Hello all, I just installed Asterisk with H323 support (chan_h323 from Jeremy McNamara). But experience problem while connecting OpenPhone to Asterisk Here is h.323 trace: 5:37.444 H323 Listener:9c86de0 transports.cxx(1504) H323TCP Started connection: host=10.120.160.15:3172, if=10.120.160.99:1720, handle=27 5:37.444 H225 Answer:9cc1250 transports.cxx(564) H225

Hello all, When I generate an out-going call from *, the DTMF detection is not working ? ASTERISK --> GW AS5300 --> PSTN But the DTMF is working correctly when it's an incoming call. PSTN - -> GW AS5300 - -> ASTERISK Well, I tried with the 3 dtmfmode of asterisk inband, rfc2833 and info, no way !!! Is it normal that asterisk try to setup the outgoing-call using ULAW ? if I

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

Hello, Users. I loaded module chan_h323.so, chan_vpb.so. I have met a message : "No one is available to answer at this time". I don?t know what I do.. My 'h.323 trace 5' result is : == vpb/1-8: Starting record mode (codec=0)[AST_FORMAT_SLINEAR:VPB_LINEAR] -- Executing Dial("vpb/1-8", "h323/192.168.1.107") in new stack 1:21:34.936 ThreadID=0x06f2bbb0

I've configured Asterisk 0.7.2 to work together with Cisco ATA186 (SIP,G.711. RFC2833) and OpenPhone (H.323, G.711). But there is an issue while calling from ATA186 to OpenPhone via Astrisk - when I press any key on analogue phone connected to ATA, Asterisk shows following message: -- Executing Dial("SIP/519-3781", "OH323/62.213.36.100|20|Tt") in new stack --

FC2 Asterisk 1.0 When I dial a H323 dialup to an existing OKI Voip Router (BV1250), I get an EndedByRefusal yet the OKI Gateway is setup with the corrent reply ip addresses etc etc, unfortunately its an existing multiple voip router setup with g723.1 and g729a, so changing the codec on the router maybe an issue. I have compile in the h323 as per the channels/h323 setup with the listed libraries.

Hello, I think I have the same problem as Martin Bene mentioned in http://lists.digium.com/pipermail/asterisk-users/2004-January/034521.html Since I found no further information about this I'd like to ask wether you know what the reason for this problem is and how one can get around this. * is registered to the innovaphone gatekeeper. Trunk connection is done with an AVM-B1 and chan_capi.

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

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",

I have searched and searched for over a week on this but can't seem to find the issue. Calls from CallManager to Asterisk are being disconnected immediately. I have setup CallManager and Asterisk per Shaun Ewing's pdf http://asterisk.edropbox.net/ccmasteriskvm.pdf I have installed Asterisk 1.4.0-beta3 on Fedora Core 5. I got libpri, zaptel, and asterisk compiled and installed.