[Asterisk-Users] router UDP timeout

if there is no data sent from the client to the server for this amount of time. Would a solution be to send a packet of silence every minute or so? Many thanks Steven Langley ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users

[Asterisk-Users] echo cancellation

Hi there I am using asterisk version 1.2.4. I have clients based on the iax client library dialling into meetme sessions. I am experiencing echo in the case where one or more users has speakers instead of headphones. So the audio from me is fed from the other participant's speakers into their mic

[Asterisk-Users] Meetme echo cancellation

Title: Meetme echo cancellation Hi there I am using IAX2 softphones dialing into a meetme conference. In my softphone I was forcing uses to click on a button when they wanted to speak, enabling their microphone and disabling their speakers. This way when a user was speaking they did not

[Asterisk-Users] Transfer to meetme on different server

Title: Transfer to meetme on different server Hi there I am using IAX2 based phones and am wondering if the following is possible: 1. User registers with Server 1 2. User dials an extension on Server 1 3. Extension transfers call to an extension on Server 2, which transfers the call to a

[Asterisk-Users] jittering with Iax2 and Meetme on Asterisk 1.2.0

Title: jittering with Iax2 and Meetme on Asterisk 1.2.0 Hi I have been using Asterisk 1.0.9 fairly successfully. I have Iax2 softphones based on the IaxClient library that are dialing into Meetme conferences. I am using a Zaptel card as a timing source. I am now trying to migrate to

[Asterisk-Users] errors with chan_zap.c when installing asterisk-1.2.0-rc2

Title: errors with chan_zap.c when installing asterisk-1.2.0-rc2 Hi I am trying to install Asterisk-1.2.0-rc2 on Ubuntu Linux and am getting a lot of errors with chan_zap.c. Most of the errors look like this: Chan_zap.c:10927: error: dereferencing pointer to incomplete type I have already

[Asterisk-Users] Re: IAX2 calls being droppped

Steven Langley wrote: Hi there I am using an IAX2 softphone built from the IaxClient library dialing into Meetme conferences. It works fine most of the time, but sometimes calls are being dropped and this error is given: Chan_iax2.c:1480 attempt_transmit: Max retries exceeded

[Asterisk-Users] IAX2 calls being droppped

PROTECTED]:4569/3 (type = 2, subclass = 1024, ts=655380, seqno=177) This error is pretty erratic. It mostly happens the first time you try to dial, but also seems to sometimes be happening in the middle of a conversation. Any ideas what the problem could be? Many thanks Steven Langley attachment

[Asterisk-Users] Re: delays with IAX2 and Meetme

: [Asterisk-Users] Re: delays with IAX2 and Meetme To: asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] In article [EMAIL PROTECTED], Steven Langley [EMAIL PROTECTED] wrote: I am using IAX2 softphones dialing into meetme conferences. I also have jitterbuffer=yes

[Asterisk-Users] Re: delays with IAX2 and Meetme

To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; charset=ISO-8859-1; format=flowed Steven Langley wrote: Hi Tony Thanks for the reply and for posting the code. I added the code

[Asterisk-Users] delays with IAX2 and Meetme

Hi there I am using IAX2 softphones dialing into meetme conferences. I also have jitterbuffer=yes, with typical jitterbuffer settings. The problem I am having is that as soon as there is a delay from a participant, then the delay continues until the participant hangs up and dials in

[Asterisk-Users] will a firewall slow down asterisk?

Hi there I am in the process of setting up a production Asterisk server, which will mainly be used for meetme conferencing. I am considering running a firewall, but wondering whether this will slow Asterisk down if all packets are being scanned. Any ideas? Many thanks Steven

[Asterisk-Users] can one specify talking only for a participant in app_conference

compared to when they are only listening. In app_conference is it possible to stop sending voice packets to a participant when they are talking? Many thanks Steven Langley ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http

[Asterisk-Users] Ztdummy or Zaptel card on production server

to go for a Zaptel card for timing, would pretty much any FXO card be suitable? Any advice on what to do would be great. Steven Langley ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo

[Asterisk-Users] choice of processors

Hi there I am moving into a production environment. I will mostly be using Meetme, with Ztdummy for timing. I have a question on which of 2 processor setups is favourable. I have the choice between Dual 2.8GHz Xeon Processors and a single Pentium 4 3.06GHz Processor. These will cost me exactly

[Asterisk-Users] Can Ztdummy be used in production environment

Hi there I have been using Asterisk Meetme with Ztdummy for timing. It seems to work fine and I havent had any major problems. I am now moving into a production environment and am wondering if it is better to use a Zaptel card? Are there any problems with Ztdummy? I will probably have

[Asterisk-Users] Meetme - any way to stop a participant receiving audio?

Hi there I am using Meetme. Now, I know it is possible to mute a user in a conference, but is it possible to stop a user receiving audio at a specific time (basically when they speak) and to do this through the Manager API. Looking at the Asterisk wiki it seems there might be some

[Asterisk-Users] IAX Jitter

Hi there I have a question regarding IAX jitter. I have 3 users on a LAN dialing into a Meetme conference on an Asterisk box which is also hosted on the LAN. I have set jitterbuffer = no and tos = lowdelay. Now, for 2 of the users the audio is fine, but for the 3rd user there is

[Asterisk-Users] jitterbuffer stability and use with meetme

Hi there I have users that are using IAX clients, dialling into meetme conferences. They will be on varying connection speeds. Firstly, should jitterbuffer be used with meetme? Secondly, I have read some posts which indicate that jitterbuffer is not that stable. Is it stable enough to

[Asterisk-Users] IAX jitter

Hi there I have a question regarding IAX jitter. I have 3 users on a LAN dialing into a Meetme conference on an Asterisk box which is also hosted on the LAN. I have set jitterbuffer = no and tos = lowdelay. Now, for 2 of the users the audio is fine, but for the 3rd user there is

[Asterisk-Users] need resources to include iax softphone functionality in vb6 app

Hi there I am looking for an open-source softphone / control for windows that I can use in a VB 6 application that will be for commercial use. I also need support for GSM, ulaw / alaw and possibly ilbc / speex. I have found a couple of possibilities, but none of them quite suit my

[Asterisk-Users] NAT issues

Hi there I have got a really strange issue and my problem is not that it is not working, but why it is working. I have Asterisk set up on a public IP, but the clients are behind a Port Restricted NAT with no support for UPnP. My clients dial into a meetme conference. When I don't

[Asterisk-Users] codec quality

Hi there I am using Meetme and have been testing with 2 different codecs GSM and g.711 these seem to be the only 2 free codecs which are supported by my soft phones (built using the RTC Client API). All users will be using this same softphone when communicating. The quality of g.711

[Asterisk-Users] Meetme disconnecting clients that use VAD

Hi there I am using Meetme and am connecting with clients that use VAD. The clients have been built with RTC Client API. What Meetme seems to do is cut users off from the conference if it does not receive any audio packets from the user for 1 minute 45 seconds. The solution I have found