similar to: delays with IAX2 and Meetme

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 intermittent break up in the audio when they are receiving. I have had a look at "iax2 show channels" and

Hello, Situation: I've got two asterisk 1.2.4 servers, connected to each other over the internet with IAX2 with about 20msec delay. One of the servers is hosting MeetMe. It's working fine as long as only SIP phones connected to the meetme server participate in the conference. As soon as a participant using IAX2 is connecting, lots and lots of buffer overruns and underruns are

Hi there I have been using Asterisk Meetme with Ztdummy for timing. It seems to work fine and I haven't 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 around 30 concurrent users in different conferences. Many thanks Steven -------------- next part

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 hear their voice half a second later (because meetme mixes the voice and sends to everyone in the conference). Now because of requirements

Hi Peoples I am having problems with meetme, in that it responds with "conf-invalid" when I dial a conference number. I notice that there is a note with regards to ztdummy, and the need for that to be loaded. Is this still the case? Is meetme dependent on this module? I do NOT use zaptel cards in my system, and there for zaptel is not loading. Can anyone shed some light

Is this patch correct? The "&&" doesn't make logical sense to me. I think it should be "||" and making this change fixes the problem I have with SIP phones in MeetMe conferences. If it's correct, is there someplace more formal that I should submit it to? *** app_meetme.c.old 2009-10-11 17:56:44.000000000 -0400 --- app_meetme.c 2009-10-17

Hi all, I am trying to setup a three party conference using the Asterisk Manager API. I am using the Redirect action over an established two party call. The procedure I am using is to try to redirect the two existing channels to a third party. I would expect this to connect both channels to the third party. However, one of the two parties gets disconnected. Is this the expected behavior? Is there

hi all, is there a kind of application can let asterisk call out fellows, and invite them to come to join the meetme. these fellows do not need to call in asterisk , just wait for a call. 3x welemon

What does it take to get ztdummy to work correctly? I have a new laptop HP HDX9200. I am running asterisk 1.4.19 and zaptel 1.4.9.2 Zaptel compiles fine. asterisk compiles fine. ztdummy loads asterisk runs. Problem is playback() does not work. So then I stop zaptel, asterisk runs and playback() now works. However, meetme()'s dont work. I need ztdummy I'm pretty sure for that. I am

When I call meetme: exten => 1000,1,Answer exten => 1000,n,Meetme(|||d) Asterisk is complaing with: -- Executing Answer("IAX2/xxx.yyy.142.204:4569-2", "") in new stack -- Executing MeetMe("IAX2/xxx.yyy.142.204:4569-2", "|||d") in new stack -- Playing 'conf-getconfno' (language 'en') Warning, flexible rate not

Just curious if anyone knows of any hacks to enable announce entry/exit in MeetMe conferences with SIP (as opposed to ZAP) channels since the |i option will not work with SIP. Thanks, Mike

I'm running 2.4.5 and app_meetme never plays conf-hasleft or conf-hasjoined with user names. I looked at app_meetme.c, but couldn't determine the cause. Any suggestions are greatly appreciated. exten => 600,1,MeetMe(600|i) I get the following: -- Executing MeetMe("SIP/jon-21f8", "600|aciMps") in new stack == Parsing '/etc/asterisk/meetme.conf': Found

Hello List, I have seen from the following link that, for SIP channels there is no audio communication possible in MeetMe with AGI_BACKGROUND. http://www.voip-info.org/wiki/view/Asterisk+cmd+MeetMe Currently we are using asterisk-1.6.2 and the problem still persists. Is there any solution available to overcome this problem? According to our requirement, we have to run an AGI script in MeetMe.

Hi, In a VoIP only environment, Asterisk has to use ztdummy to have any chance of playing back understandable audio files (without drops, hickups etc). I have been using ztdummy to some degree of success, but I also have a "Wildcard TDM400P REV E/F Board 1" in the Asterisk machine I'm using. I'm not using this card for anything at all, but I'm wondering how to set it

has anyone done caller announce in MeetMe's before? Dave P >>> brian@bkw.org 5/18/2004 5:50:49 PM >>> With multiple parking lots you can give each person their own lot thus exten 800 for everyone will connect them with just their call passing the lot name which you know for X customer. bkw ----- Original Message ----- From: "Andrew Kohlsmith"

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

Anybody having issues with ztdummy under the current 2.6 RC7? I get the following errors when trying to modprobe ztdummy: "Unable to register zaptel rtc driver" Doing a Google on the error shows reference to a message from 2004 that said you might not have RTC compiled into the kernel. Checking via: cd /usr/src/linux-2.6.13-rc7 grep -i rtc .config shows: CONFIG_APM_RTC_IS_GMT=y

Would anybody mind if the the following command options where added to MeetMeAdmin? 0 - 9, * and # I'm considering hacking the code to add these commands to play the DTMFs to the specified user as tones and hope that the SIP or IAX channels then work with these correctly. -HJC

Hey folks, Using FreeBSD 5.2.1 and I've got the current zaptel driver installed from ports (0.8_1) and current ports asterisk (1.0.1). I've set options HZ=1000 in my kernel config, recompiled and rebooted and as far as I can tell, I've done everything right but when I try to use the conference, the audio is very delayed, choppy and segmented -- totally unusable. At the

I've been scratching my head trying to think of a way to do this, but without success yet. I'm using the Manager API. If I have two channels linked to each other (i.e. direct connection), or even if they are independent channels, I can transfer them both to the same extension by using Action: Redirect and using Channel: for one and ExtraChannel: for the other. This is most useful for