similar to: Call transfering, conferencing

hello, my situation is h323gw - gatekeeper - asterisk - SIP client my problem is, that I can't make call from h323gw, when this GW is Micronet (sp5004). A ----------- CUT ----------- -- Executing Wait("H323/ip$62.152.225.18:52434/20702", "1") in new stack == Spawn extension (postel, 169, 1) exited non-zero on 'H323/ip$62.152.225.18:52434/20702' -----------

hello, can anybody help me with folloving problem I have asterisk with the public IP and two UAs (snom100, x-lite) in the same private network behind the same FW/NAT. All is working good, but whan I tried to establish call between these two UAs, first 10-15 second is nothing to hear and then is the quality terrible :( Can anyone tell how to get it work with normal quality ? best regards

hello, my situations is as follows. In our comapny we are planing to have *. I'm testing it now. If we will buy G729 codec for * ... UA(SIP) <-> FW <-> (SIP)*(H323) <-> (H323)GATEKEEPER(H323) <-> (H323)AS5300 <-> world the following equipment speeks G729: *, GK ,AS5300. All call from UA to another endpoing go through *, because of mixed SIP and H323

hi all, I don't know how to setup asterix to work as PBX. If I want just basic configuration with 2 SIP phones (snom, ata), what all I have to write in the configuration files, or respectively in the configuration of ata and snom ? If there is any good documention available, send me URL too. All (ata, snom) are behind firewall (nat) and astrix is on the public IP, but I can move for

hi, I known, that this is not * related, but a lot of members of this ML uses GS BT phones. I have patched the atftp serber to recognize the TFTP OPTION, whis these phone send during boot. Patch includes - another locations for configs, firmware and ring tones - different FW versions for phones - custom ring tones for the phones You can find patch, source/unpatched/ and DEB for

On Feb 18, 2015 7:43 PM, "Tris Hoar" <trishoar at bgfl.org> wrote: > > On 18/02/2015 07:17, aditya hilman wrote: >> >> Hi folks, >> >> I've already configured split DNS for internal-view and external-view. Also >> already configured the master - slave dns. >> But i've problem with external-view zone transfer. >> Based on the

On 18/02/2015 07:17, aditya hilman wrote: > Hi folks, > > I've already configured split DNS for internal-view and external-view. Also > already configured the master - slave dns. > But i've problem with external-view zone transfer. > Based on the logs, the master notify to slave using the public ip, which is > not accessible by master to transfering the zone over

On Wed, Feb 18, 2015 at 7:47 PM, aditya hilman <aditya.hilman at gmail.com> wrote: > > On Feb 18, 2015 7:43 PM, "Tris Hoar" <trishoar at bgfl.org> wrote: > > > > On 18/02/2015 07:17, aditya hilman wrote: > >> > >> Hi folks, > >> > >> I've already configured split DNS for internal-view and external-view. > Also >

I have difficulties in interpreting high order interaction terms in high-way ANOVA. According to Introductory Statistics with R by Peter Dalgaard (Section 12.5), "The exact definition of the interaction terms and the interpretation of their associated regression coefficients can be elusive. Some peculiar things happen if an interaction term is present but one or more of the main effects are

My wife bought an inexpensive store brand web cam, so she can do video conferencing with people using Microsoft NetMeeting. Probably it is not supported in Linux and we will need to buy a supported web cam. I installed the Ekiga package. Is there something easier to configure and use or is Ekiga the best way to go? TIA! -- Lanny --------------------------------------------------------- Over 800

Could anyone recommend a package, preferably available via yum, to host web conferencing / collaboration? I am interested in being able to share an XP desktop with up to 5 other users. Any remote control capabilities would be a plus. Am currently running a Centos 4.4 server. Thanks, GB

Any one knows how to make Meet Me video conferencing room. Regards ********************************************* No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily

UNICOM, Video Conferencing in Pakistan http://www.unicom.net.pk/ We at Unicom are pleased to inform you that we have expanded our network of video conferencing studios in all major cities of Pakistan including Karachi, Lahore, Peshawar, Islamabad, Rawalpindi, Quetta, Hayderabad, Nawabshah, Muzafarabad, Sialkot and now in Faisalabad and Gujrawala . All of our studio are equipped with professional

Hello , I am a newbie to * and have just been able to call a sip User Agent on a different machine thru *. I was trying to set up conferencing between 3 sip useragent on different macines at my worplace but was not able to figure out the procedure. I made the changes in meetme.conf and extension.conf as specified by someone in this mailing list, but * giving some error, " No ISA Tormeta

Hi this is just a heads up about an opportunity for commercial Asterisk experts. I don't know if this even possible but don't see why not and it is way beyond my capabilities so thought I would pass it out to the list. I've been looking into Microsoft Live Communications Server over the past few months for one of my clients, it's the same as ms messenger but for closed user

Ok, maybe a _wee_ bit esoteric, but... I've setup a developemnt system on a Sun Netra T1 running Aurora Linux (rh-7.3 for sparc64) -- just a few minor Makefile changes in codecs necessary. All in all, it's running nice. My only problem is lack of meetme or comprable features. I can't get zapdummy (these boxes use usb-ohci, not usb-uhci) to compile, and zaprtc sure won't work

http://www.voip-info.org/tiki-index.php?page=Asterisk%20bounty%20Meet%20 Me%20video%20conferencing I posted this bounty for $US2,000 some months ago. Basically I needed the ability for 4 or 5 of us to conference on a weekly basis which is why I was happy to offer this bounty, however I have only had 2 people make brief inquiries and no one has really offered any substantial indication they

I'm currently writing some code to support conferencing in Asterisk without using the Meetme application. The conference runs in its own thread and every new inbound or outbound channel that is created is passed to it. This thread runs the conference loop reading and writing frames to each channel. I'm writing this as if it were a bridge with more than two channels, and I'm not

Hi We are having a significant (> 1 sec) delay in a multi-asterisk conference, with IAX2 legs connecting meetme on different boxes. All the other legs are PSTN (TE410P). The example configuration Slave box 1 meetme <--- IAX2 ---> Master box meetme <--- IAX2 ---> Slave box 2 meetme The delay is between Slave box 1 and Slave box 2 The primary suspect is our iax configuration

I received this email link from Macromedia today http://www.macromedia.com/newsletters/edge/february2005/index.html?secti onIndex=3&trackingid=AWQX They are now selling Breeze video conferencing server by the minute as an asp, this service is powered by the macromedia communications server which is just a sip server like asterisk but for video. I'm still offering a $2,000 bounty