similar to: Using Asterisk as H323 gateway

Hi there I am pretty close on giving up on Asterisk :-/ I am (still) trying to make a call from a H323 phone to an Asterisk provider using AIX. But H323 does not route the number to AIX. All it is transmitting is an "s". *CLI> -- Executing Dial("OH323/R27865", "IAX2/demo:demo@gw1.musimi.dk/s") in new stack -- Called demo:demo@gw1.musimi.dk/s Jul 2

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 --

hi all, i have one (teles) pbx with a BRI telephone and an outgoing E1 port. The outgoing E1 is connected to an pri_net port from my *. The incoming call will dail out to a h323 soft phone like openphone or sjphone or just netmeeting. The call will be conneted, but i don't hear any sound, from no one of the both sides. Can somebody help me? Thanks, Thomas.

hello i am using asterisk-oh323-0.7.1. i want to convert sip call to h323 (h323 sjphone or h323 proxy). what could be the best way for this. i am successfull in converting h323->sip by using asterisk as gateway. help required on sip->h323. kamran __________________________________ Celebrate Yahoo!'s 10th Birthday! Yahoo! Netrospective: 100 Moments of the Web

What is the configuration of H323.conf and openphone in order to run openphone and asterisk together ?

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, I just installed OH323 Plugin and im now tryin to make simple Configuration to connect Openphone and Xlite to my Asterisk-Server. All works fine, i just wanna know if there's a better way to do it? Is there anything wrong with my Config? OH323.conf [general] listenAddress=0.0.0.0 listenPort=1720 connectPort=1720 tcpStart=10000 tcpEnd=20000 udpStart=8000 udpEnd=8005 fastStart=no

Hi, when running a R-script like this: enable_magic() compute_stuff() disable_magic() the whole script is parsed into a single expression and then evaluated, whereas when using the interactive shell after each line entered, a REPL loop happens. Is there a way to make a script evaluation behave like this, because I need a single REPL iteration for every expression in the script. It doesn't

Hi, I've been trying, without success to find a copy of the paper, by Catherine Loader, that describes the algorithn underlying the rbinom() and associated functions. The title is "Fast and Accurate Computation of Binomial Probabilities." All of the links to the paper that I've seen (including in the R docs) lead nowhere (i.e. are 404). I've sent Dr. Loader several emails,

Hi, I run asterisk with oh323 plugins.It runs correctly with sjphone H323 Gatekeeper. But When i run openphone it doesn't recognize my asterisk server like a gatekeeper !! What is the problem ? Thx

hi all i would like to know if anyone has a reference on how one would place the "bands" on the pp plot. i want to test whether or not a certain data set comes from a particular distribution (not normal). i've already plotted F(X(j)) vs j/(n+1) where F(x) is the cum dist function, X(j) is the j'th order statistic and n is the sample size. a goole search gave arb references

Hello I am a newbie to Asterisk. We have set up Asterisk on a PC with Redhat 9.0. We have installed H323 openphone on our PC's. We are wondering what a gatekeeper does. It seems we need one but what I have seen in this group is that a gatekeeper must be installed on another box on the network. As all our PC's on the network use Microsoft OS is there a free gatekeeper software for

Hello, Good news on the Polycom front for those that are interested. It looks like we may get a dedicated Engineer for Polycom/Asterisk!!! Happy Day! Here's the message I got tonight: Matt: I heard back from our VP of Engineering- she is prepared to have an individual dedicated to working on the Digium- Asterisk project. Can we discuss again Friday or mid next week? Scott Willard

Hello. I just tried (as my first attempt in xenning) to set up a lenny amd64 domU on a partitionable Mirror RAID (/dev/md_d0p1 - /dev/md_d0p4), using the Debian installer from "people.debian.org/~joeyh/d-i/images/daily/". The partitions of the md will be represented inside the domU as /dev/xvda1-4 accordingly. The dom0 seems to run fine, it''s Xen 3.2.1 on amd64 lenny. The

I am having trouble with this. I can dial 1800 numbers fine as well as FWD service numbers but not Vonage. I can be called from ipkall and fwd and can call aixtel numbers. I use aix2 with Fwd. My extensions.conf for Vonage: ; vonage numbers ; ; +2431 exten => _2431XXXXXXXXXX,1,SetCallerID,${FWDCIDNAME} exten =>

> http://www.speakfreely.org/ > > I think this would be one of the best real-world tests of the speex codec. > This software doesnt use ACM or directsound api's but uses straight C code. > I was thinking the speexenc/speexdec should be easy enough to add. The last time I looked at this it was still very much old news - mostly half duplex audio, does not adhere to any

Hello, does anyone know how to compute the following two normality tests using R: (1) the Kiefer-Salmon (1983) statistic, Economics Letters 11, p. 123-127 (2) the modified Shapiro-Wilk statistic? Thank you very much. Geoff [[alternative HTML version deleted]]

Update on this problem: I gave up on the "native" h.323 because, like others, I couldn't get audio working. (yes, I tried disabling FastStart in ast_h323.cpp - no change) So I went and got the OH323 code from www.inaccessnetworks.com. Glad to say that everything seems to work so far. Not only does audio work, but even the handshaking is now working in both OpenPhone and even

Hello list, using Asterisk 1.4.30. I want to set the SIP-header Remote-Party-ID to display the name of the calling party on my phone in stead of the number. This is the dialplan : exten => 10,1,NoOp() exten => 10,n,SIPAddHeader(Remote-Party-ID: "eric" <sip:10 at 192.168.1.150>;party=called ) exten => 10,n,Dial(SIP/test2) This is what the CLI shows : /[Jul 12

Is there any good IAX2 or SIP free web client? Im looking for something open source or free preferably IAX2 for integrating into a web site... Any leads?