similar to: anyone with X100P & Callerid working outside US ?

I have on Chan_h323 with G729 and X100P trying to connect to a Planet VOIP400 gateway box(http://www.planet.com.tw) I uncommented g729 in the Makefile and I'm setting g729 in h323.conf I'm receving in my side: 1:20.906 H225 Caller:810f070 h323ep.cxx(1537) H323 Clearing connection ip$localhost/4112 reason=EndedByRemoteUser and the other side(Planet) says: 15- RADH 2

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

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 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

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

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

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 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 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

Hi List, I am facing the reverse problem as stated here.I am using ATA 186 to make and recieve call to * through OH323 driver. When I use G711 codec in the ATA to make call then then as soon as i dial an extension the * crashes with 'segmentation fault'. But the same scenerio works fine when i use 723 codec in the ATA .I can dial the number and extension very well/(I have 723 support in

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:

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.

Hi, Are configurations (with separate formats per namespace) - such as ... namespace { type = shared list = children inbox = no separator = / subscriptions = no prefix = shared1/%%n/ location = maildir:/var/mail1/%%n/ } namespace { type = shared list = children inbox = no separator = / subscriptions = no prefix = shared2/%%n/ location =

Hi All. I'm using RedHat 9 I configured the chan_h323 and asterisk from CVS. This is the scenario SJ_lab_phone(sip) ---------------> Asterisk -------------> H323 GK --------------> PSTN I have tried all codec's and always the same result, the called phone will ring without dropping for how ever I allow it to but as soon as it is answered it immediately gets disconnected.

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

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

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 =>

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:

Hi, I don't know how to sample such data, it can't be done by row sampling as default method on matrix in boot. Function takes matrix and returns single coefficient. #There is a macro but I want use R :) http://www.comm.ohio-state.edu/ahayes/SPSS%20programs/kalphav2_1.SPS library(concord) library(boot) # The data are rates among observers with NA's