similar to: sip to oh323 converter converts sip uri to h.323 number and not h.323 url

Displaying 20 results from an estimated 2000 matches similar to: "sip to oh323 converter converts sip uri to h.323 number and not h.323 url"

2012 Apr 26
8
understanding the FUNCTION function
Hello, I am trying to understand why the FUNCTION used in several codes, won't create the object after it finishes running the code. For instance, look at the following: Number<- function(x) {MyNumberIs<-x} When I run Number(5) Everything goes well, except that if I try to call the object MyNumberIs, I won't find it. I understand that this function can assume many parameters,
2006 Feb 20
3
calling from SIP to a h.323 device with oh323
Hi, I've asterisk-oh323 0.7.3 and after 2 days of test finally I can make calls from one h.323 device to the world using sip trunks :) I can call to sip devices from the h.323 one. Now I want to make calls from sip to h.323 but it does not work. Maybe one of us have a configuration example to do this? I'm using the latest svn version (compiled yesterday).
2004 Jul 23
4
still can't load oh323 - Are we not supporting H.323 any more?
Why is no one suggesting any solution here for this problem, which has been lingering for a while. Are we not supporting H.323 on Asterisk? -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com]On Behalf Of ruixun wu Sent: Thursday, July 22, 2004 4:06 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] still can't
2004 Jul 14
1
oh323 dial structure and oh323 debug?
According to the wiki at voip-info.org, the dial structure for using oh323 without a gatekeeper is: OH323/<exten>@<host>:<port> or OH323/<exten> The second option is valid only in the case where a gatekeeper is used. NOTE: OpenH323 library v1.12.0 has a bug in the parsing of the destination host. When this version is used then the above syntax should be:
2002 May 09
4
Rsquared in summary(lm)
Hello, I'm doing some linear regression: >lm<-lm(osas~alp,data) >summary(lm) However, the Rsquared in the output of summary() is not the same as the "standard" Rsquared calculated by spreadsheets, and outlined in statistical guidebooks, being SSR/SSTO. The output says "multiple Rsquared", but it is no multiple regression... What's the difference? Thanks,
2004 Oct 04
1
some problems with OH323
Hello, I am facing some problems with OH323 1. I am using exten => 223,1,SetVar(OH323_OUTCODEC=gsm) or g729 exten => 223,2,Dial(OH323/x.x.x.x) but the call always go out in aLaw . 2. With OH323 I find one way audio . 3. How can I use ASTCC with OH323 , I have tried to use dialstrings like exten => 223,2,Dial(OH323/192.168.0.1/1234) to send the dialed number as 1234 , but it does
2004 Nov 25
2
oh323 compile issue
Hi all, I want to give a try to oh323 (currently nufone h323 channel is setup and compiling fine) on a yesterday CVS update of asterisk. I have _pwlib 1.8.1_ and _openh323 1.15.1_ What I made: openh323 dir: make clean apply the oh323 patch configure make opt asterisk-oh323-0.7 dir: make [...] wrapendpoint.cxx: In method `BOOL WrapH323EndPoint::OpenAudioChannel (H323Connection &, int,
2006 Apr 18
2
correct version of asterisk for oh323
Hi, i have been using asterisk CVS 19/07/2005 and asterisk-oh323-0.7.2. I now want to use oh323 with Asterisk 1.2.4+. Can anyone tell me what versions of oh323(and pwlib and oh323) they got to work with Asterisk 1.2.4+. -- thanks, yusuf
2004 Jul 29
1
OH323 and codec selection
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
2006 Mar 21
2
need to make my oh323 work with quintum no gatekeeper
Hi all, Can someone share with me his experience in making asterisk-oh323 talk to quintum gateway without gatekeeper. My set up is QUINTUM GATEWAY ------IP----M ASTERISK (OH323) Both are gateways.. but I don't know what authentication I will set up in oh323.conf and how to set it up I will be glad if anyone can help Goksie
2004 Jul 07
1
OH323-COMPILE
HI ALL HI MICHAEL; My name is mohammad and I am iranian.I have been trying to install oh323 channel but I come up with dead end. In fact it makes me crazy. plz help me michael. I saw mailing list and I trid serevel CVS headers such as , 2004-06-07( seven of june) 0r 2004-07-02( second of july) besides I use: 1-openh323 v1.12.2 2-pwlib v1.5.2 3- asterisk CVS (2004-06-07,
2004 Jul 12
1
Problems Compiling asterisk-oh323 0.6.3a
Hi, erverybody The Asterisk is running well in the linux system. Now I would like to add oh323 in Asterisk. I have download pwlib(version is 1.6.6) and openh323(version is 1.13.5). And I sucessfully maked and installed these two packages. But I got the following errors when compling the asterisk-oh323 0.6.3a: for x in wrapper asterisk-driver; do make -C $x all || exit 1 ; done make[1]:
2005 Jun 11
2
Help with Oh323
I am blocked on the most simple step when compiling Oh323: Cd /openh323 patch -p1 ../asterisk-oh323-0.7.2-pre1/openh323_1.13.5-make.patch it hangs for ever and never finishes in my machine with Red Hat Enterprise Linux 3, fully updated. I need to use Asterisk HEAD and I therefore I want to use asterisk-oh323-0.7.2-pre1. Additionally, because it is the most debugged version. In the past, if
2006 Apr 08
2
oh323.conf problem
I have installed oh323 channel driver (finaly! :)). I head some problem starting * so I have put the smallest possible oh323.conf file to se what happens. When I don't put available codec's in oh323.conf (*1) Asterisk starts but he also disables h323 channel because there are no available codec's (*2). When I put codec (*3) Asterisk doesn't start (*4). What have I done wrong? I
2003 Nov 28
0
Re: Resend: Help for oh323
Michael, Thanks a bunch, I downloaded from inaccessnetworks.com thinking that it is the latest :). Ok I will upgrade it. just for the record, following worked. exten => _87.,1,Dial(OH323/H323:${EXTEN:1}@16.52.153.206) Cheers Sathya Date: Fri, 28 Nov 2003 11:28:59 +0200 From: Michael Manousos <manousos@inaccessnetworks.com> Organization: inAccess Networks To:
2004 Jun 02
1
oh323: Failed to create smoother
Hello, I tried to get the oh323 drivers running. The driver loads, but as soon as a H323 voice communication should be started, following error occurs: -- Executing Playback("OH323/R1", "invalid") in new stack Jun 3 01:26:20 ERROR[294931]: chan_oh323.c:1933 oh323_write: OH323/R1: Failed to create smoother. Jun 3 01:26:20 WARNING[294931]: file.c:539
2004 Sep 09
2
Dial Out w/ OH323
Due to the format of the message coming from the H323 channels included w/ Asterisk we were unable to use our gatekeeper. For a quick solution we tried the OH323 channel drivers and can receive inbound calls from the parent gatekeeper. We are trying to do a dial to gatekeeper... I am trying exten => 5551212,1,Wait,2 exten => 5551212,2,Dial,OH323/5551212 But I am not sure if this is the
2004 Oct 08
0
problems with asterisk-oh323-0.6.3b
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
2005 Jan 13
0
oh323 compile problem still
Followed instructions from these old post, CVS updated my asterisk too, edites makefile... but ---------------------------------------------------------- Get oh323 from http://www.inaccessnetworks.com/projects/asterisk-oh323/Libraries/ openh323-Janus_patch4-src-tar.gz Get pwlib from http://www.inaccessnetworks.com/projects/asterisk-oh323/Libraries/ pwlib-Janus_patch4-src-tar.gz Get
2005 Jan 03
3
oh323 context for peers
I am experimenting with oh323 channels and h.323 gateways and a Cisco CallManager. I am not using a gatekeeper at this time. Is it possible to place calls coming into Asterisk from specific peers into specific contexts? In iax.conf eaxh peer has a context in which I can specify the context an inbound call will be placed in. I don't see anything like this in the oh323.conf file or the oh323