similar to: Help for oh323

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 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, My configuration is SipPhone<-->*1<--->*2. My asterisk version is 1.4beta3. I installed pwlib,openh323,chan_h323. When i call from SipPhone--(SIP)-->asterisk1---(H323)-->asterisk2, there is no audio. Using 'rtp debug', I can see that rtp packets are being received. Rtp packets are being exchanged. I also tested chan_ooh323, but to fail. Can anyone recommand best

Now, TOSHARG-DomainMember.xml translate to Japanese finished. and Some bug found. <procedure> <title>Server Manager Account Machine Account Management</title> -------Domain? <step><para> From the menu select <guimenu>Computer</guimenu>. </para></step> When the user elects to make the

Hello, I have a strange h323 issue. After executing command "Dial("SIP/333-0d1dfe00", "H323/361737052390920 at ccg|5|tT")" at Oct 18 22:32:23. Meanwile I have sniffing traffic on port 1720. The call was established just at Oct 18 22:33:03 (New H.323 Connection created.) and also packet sniffer grabs the h323 invites at this time also. So my question is what

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:

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

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

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:

Somehow autofilter doesn't allow this message to be posted, will try another time. -----Original Message----- From: Yuri Volchik <volchik2000 at list.ru> To: r-devel at r-project.org Date: Thu, 25 Jan 2007 22:27:13 +0000 Subject: Using Windows API functions in R > > Hi to all. > > In programming one application i have to "press" button to have > application

Date: Fri, 10 Sep 2004 16:37:33 +0300 > From: Michael Manousos <manousos@inaccessnetworks.com> > Subject: Re: [Asterisk-Users] Problems with 0penh323 Channel Driver > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: <4141AE1D.3020403@inaccessnetworks.com> > Content-Type: text/plain; charset=us-ascii;

Hi all, I'm using following software and equipment and I have very strange behavior: Asterisk CVS-NHEAD-05/30/05-16:42:41 H323 gatekeeper - GnuGK 2.2.2 IAX2 client - Firefly 1.9.8 build 3945 H323 client - SJPhone Build 1.50.271d H323 gateway - Welltech Wellgate 3504A When I dial from Firefly (IAX2) -> SJPhone (H323) everything works as expected. When I dial from SJPhone (H323) ->

Hi guys, I've been trying to update my chan_oh323 from 6.1 to 6.3b. I built asterisk from cvs-head on the date Micheal said he made it compatible, pwlib-1.6.6 and openh323-1.13.5 (both with nothing more than the ./configure, make, well aplied patch on openh323) When I start * with my normal config I get this: [chan_oh323.so] => (OpenH323 Channel Driver) == Parsing

Could you help me with this problem? When I call H323 gateway there is no sound in both ways. Here is h323 debug: ----- begin ------------------------ -- Executing Dial("SIP/msn-6297", "H323/73952389512@peer:1720") in new stack Allowed Codecs: Table: G.729A{sw} <1> G.729{sw} <2> G.711-uLaw-64k <3> G.711-ALaw-64k <4>

Hi all, I used oh323 support from inaccess. It work very well. I would like to test h323 integrated support. This my problem when I test it : I cannot heard any thing in both way. The test is : SIP --> Asterisk --> H323 This is th debug trace from h.323 : -- Executing Dial("SIP/someaccount", "H323/0033172897104@somehost") in new stack

Hello, I'm trying to use the GSM codec with Nufone H323 but it's not working. Does somebody has some idea? Have I missed something? Thanks!! Celso Fassoni Some additional info: (I'm using CVS-HEAD - downloaded today) monkey:~# cat /etc/asterisk/h323.conf [general] port = 1720 bindaddr = 192.168.0.100 ; this SHALL contain a single, valid IP address for this machine

Hi, all! when my h323 phone dial in Asterisk system, i can hear nothing. and the following is the log slice i picked from /var/log/asterisk/full. ps: i am using red hat AS5 kernel 2.6.18-53.el5,Asterisk-1.4.24.1, pwlib_v1_11_0, openh323_v1_19_0_1. Best Regards! 81948 [May 6 10:07:34] VERBOSE[11579] logger.c: -- Remote UNIX connection 81949 [May 6 10:07:51] VERBOSE[29627] logger.c:

Here is an example of how to do it. > library(survival) > vfit <- coxph(Surv(time, status) ~ celltype + trt, data=veteran) > userinput <- data.frame(celltype="smallcell", trt = 1) > usercurve <- survfit(vfit, newdata=userinput) #the entire predicted survival curve > user2 <- summary(usercurve, time= 2*365.25) # 2 year time point > user2$surv [1]

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

Hello all, I am a medical student and as a capstone for my summer research project I am going to create a simple online web "calculator" for users to input their relevant data, and a probability of relapse within 5 years will be computed and returned based on the Cox PH model I have developed. The issue I'm having is finding a definitive method/function to feed the user's