similar to: Meetme doesn't react to DTMF keys

I have looked thru the users lists, google and voip-info and did not find an answer. I am using asterisk (latest SVN as of three weeks ago). I am using real time. I have a problem that when a user enters an invalid meetme extension the meetme says invalid room and dumps the call. Here is what I have in mysql:

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

Hi, I am using Debian OS kernel 2.6.22-3-amd64 and zaptel driver 1.4 with ztdummy module for meetme application. I use meetme with SIP channels. I have such problem that when one connects to the conference voice is "cut". Each voice sequence is disturbed. Does any one have similar issue and could give me some advice?? my extension.conf for meetme: ;switch =>

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

This is weird. If I have 2 members call into meetme using zap PRI channels on the box, they can here each other's keypresses. If I have 2 members call into a separate box using the same PRI's and then forward (dial(iax/...)) them to the previous box into the same meetme, they only hear a minor "squelch" for each other's keypresses. How can I completely mute a

First, the scenarios: Call placed from Boston to locally configured Asterisk Meetme extension: Cisco 7941 <-SCCP-> Cisco 2821(CME,Boston) <-SIP-> Asterisk(Boston) Call placed from Boston to European Asterisk Meetme extension: Cisco 7941 <-SCCP-> Cisco 2821(CME,Boston) <-SIP-> Cisco 2821(CME,Europe) <-SIP-> Asterisk(Boston) In the 1st scenario, everything works

Hello List, I am facing some issue while passing DTMF (RFC2833 set globally in sip.conf) in meetme (asterisk 1.8). The issue I have observed that - if two users tries to pass DTMF simultaneously at the same time from their phones only one DTMF is detected in asterisk and broadcasted to other users. Other DTMF lost somewhere. We have tested only with sip phones. Can someone help me with this, or

Hello List, I am facing some issue while passing DTMF (RFC2833 set globally in sip.conf) in meetme (asterisk 1.8). The issue I have observed that - if two users tries to pass DTMF simultaneously at the same time from their phones only one DTMF is detected in asterisk and broadcasted to other users. Other DTMF lost somewhere. We have tested only with sip phones. Can someone help me with this, or

SYNOPSIS Erratic red alarm T1 PRI on asterisk, but zttool running concurrently during alarm shows no errors, irq misses, or alarms, on any span. Using asterisk and quad Digium T405P, configured as follows: Span 1 connects to ISDN PRI (fractional 8 B channels, D channel 24). Span 2 connects to T1 Mux and analog stations. Span 3 connects to ISDN PRI Nortel BCM hybrid key system digital trunk.

hello i want to kick participant in a meeting by pressing the digit on sip phone.when i entry the meeting ,no matter how i press the button,the dtmf does not work. here is my dialplan and my agi script,and sip.conf [from-internal] exten =>121,1,MeetMeCount(900,CONFCOUNT) exten =>121,2,GotoIF($[${CONFCOUNT}<10]?3:100) exten =>121,3,Authenticate(123456) exten

First off, let me first say that this is not a one-way audio problem. Sometimes I can get 'her' to speak to me, other times I can't for a long time. I'm just using a very simple system to dump people into MeetMe(). Nothing fancy. I do have the following in my modules.conf: preload => format_mp3.so preload => codec_ulaw.so preload => format_pcm.so My extensions.conf

Thought a different succinct subject line must drum up an answer or two... Also, this has been tested from two different carriers: We're getting an average of 2/10 call success rate. ---------- Forwarded message ---------- From: Joe Wood <schmoe at gmail.com> Date: Thu, Aug 26, 2010 at 6:58 PM Subject: Asterisk 1.6 Displaying in Debug that it is playing a ulaw file using BackGround()

Hello! I am trying to set up multi-link PPP using two T100P cards in one machine, and 1 T405P card (the 4-port one) in another machine. I have previously been able to get PPP working between the two T100P cards in separate machines.... The 4-port card seems to be my problem currently. I am trying to use the tor2 driver (from a fresh CVS download this morning). When I load the driver (or run

Hi: When should we use "unnumbered priorities"(n) in extensions.What is the different between these 2 forms of extensions.conf? and ,Are both true? extensions.conf: form1: [Conferencerooms] exten => 333,1,Answer exten => 333,n,meetme(8000|cim) exten => 333,n,playback(vm-goodbye) exten => 333,n,hangup form2: [Conferencerooms] exten => 333,1,Answer exten =>

Hi List, We have a major issue while broadcasting DTMF using meetme application. We are sending DTMF to asterisk using SIP INFO message with duration 160. INFO sip:xxx at xxx SIP/2.0 Via: SIP/2.0/UDP xxx:5060 From: <sip:xxx at xxx>;tag=43 To: <sip:xxx at xxx>;tag=9753.0207 Call-ID: xxx at xxx CSeq: 25634 INFO Content-Length: 26 Content-Type: application/dtmf-relay Signal= 2\r\n

Hi all, I'm experiencing a problem with meetme i can't resolve. This is my scenario: A iax client, say IaxComm, make a call through a zap channel. When it answers it is tranfered to a conference room. Then the iax client make a second call though a second zap channel, at the other side there is an IVR. Iax client send some dtmf to the IVR then it transfers the IVR to the previos

I have an X100P clone hocked up to an analog line of my PRI. I can use it to dial out. but when I call the extension it answers and says "GOODBY" I have a Livevoip DID which successfuly rings to ext 202 I am using asterisk@home and through the AMP inface the line should ring to ext 202 Below are Asterisk Messages, Extensions.conf and Extensions_additional.conf Extensions.conf

Hi , Is it possible to have two meetme room in asterisk 1.6 which each one have a different language? I mean, one room the annoucement is in Portuguese an another in english? Today I can change over the sip.conf and it is valid for all room. regards! Att, Flavio Roberto Miranda MSN:flaviormiranda at hotmail.com Skype: flaviormiranda -------------- next part -------------- An

Hi all, I've downloaded the sources of Wine-20010510 and installed them successfully (I think). But when running a Windows-app, the window is in screen, but doesn't react to my input, like mouse-clicking or typing on the keyboard. Is this a known prob? Thanks in advance, Bram Schoenmakers :-( ;-) :-)

If there are any other users who use AHRQ's SAS code comoanaly2010 and comformat2010 to create comorbidity variables, I thought you might be interested in the following PRELIM code we wrote to mimic its functionality in R. It seems to yield similar results, but may contain errors. Please feel free to comment (kindly) or enhance. I'm sure there are better ways to skin this cat, but we at