similar to: Additional commands for MeetMeAdmin

We need to conference together a call center agent, a customer and a third party IVR and send DTMF tones from the agent to the IVR. MeetMe has been eating our DTMFs so we'd like to try app_conference. Has anybody setup such a configuration in app_conference and how did you configure it? -HJC

Hello users, I have been working with a conference setup. My setup includes: 1)There will be an interface number provided to the user which might be a DID number or A Toll free number When user calls the number it asks for the conference room number and the user pin . on successfull authentication he will be participated in the conference 2)by didaling the same DID number the

I've just moved into 3.3v PCI servers and found that my clone X100P cards were lying about the 3.3v supported notch. Can I use a Wildcard TDM400P without any modules as a timer for MeetMe in a 64 bit 3.3v server? Will I still need to plug the hard disk power cable into it? Is there a better cheaper 3.3v MeetMe timer? (Boss doesn't trust the kernel timer.) -HJC

Has anybody created a crontab script to check the health of an Asterisk server? The part I'm struggling with is some sort of IAX "ping" to test the connection to each provider without making a call. -HJC

Hi Chaps, Is there a way to detect/highlight poor quality voice calls going through an asterisk server? Was thinking of picking up a cdr record or some other variable showing poor quality on calls when the internet is having issues. Is there any qos or poor audio quality variables available? Cheers, Taff. ___________________________________________________________ Yahoo! Answers - Got

Hi All I need to set my Asterisk conference such way that , during confernce Admin Can kick 1 or all user , Same for mute fuction.As well as Admin can increase or decrease conf & user volume. for that i used MeetMeAdmin like this exten => 600,1,MeetMeAdmin(1111,ekKLmMNS,7010) where 1111 is conf number & 7010 is Admin user

Is there any way that I can have an admin user hit * and then Mute all other users in a meetme conference? Sort of a moderator function? I know it can be done with MeetMeAdmin, but as I see it that requires a separate extension to dial, unless I've got the logic wrong? If it can be done in a single extension please show examples. Thanks. ________________________________ This e-mail,

Hello, I installed Asterisk 1.6.2.17.3 ( latest as of yesterday ) and had multiple problems with DTMF. I have two machines, we'll call them asterisk and asterisk-pri. Asterisk does IVR and asterisk-pri has a PRI card in it and connects to the PSTN. The two servers communicate via SIP with RFC2833. I setup logger.conf on both machines to display DTMF to the console. Both are built from

Has anyone here done any work on clustering Meetme conferences over multiple Asterisk boxes? The scenario I am thinking of is where there are two or more boxes connected to a set of PRIs that all answer to the same PSTN number, and where it's not possible to know in advance on which box a call would arrive. So it would be possible to have some calls on one box and some on another, that should

Hello A user seems unable to type DTMF in our Asterisk IVR menu. Can this be due to their phone or PBX that disables DTMFs when a user is off-hook? Thank you.

Thanks very much for the reply Joshua! So I guess that setting dtmfmode=auto would be the safest choice in order to strip out the DTMFs from the recording, right? Cheers! Patrick Wakano On Tue, 1 May 2018, 19:36 Joshua Colp, <jcolp at digium.com> wrote: > On Mon, Apr 30, 2018, at 11:23 PM, Patrick Wakano wrote: > > Hello list, > > Hope you are all doing fine! > > >

Hi ! Did anyone had issues/managed to solve issues with DISA over Zap channels on * 1.0.X (STABLE) ? I have a situatuion where DTMFs that should be recognized in DISA work over SIP channels and do not work over ZAP channels (Zap channels are on TE110P) I have in default context: exten=> 299,1,DISA(no-password|default) and I have SIP extension 200 in [default] and I have Zap trunk which

Hi, I'm having two problems. First - I'm using the xten x-lite program to communicate with asterisk, and everything works fine except that DTMFs are not transferred. I've set DTMFMODE to inband on both the sip.conf file and the x-lite configuration, and still it doesn't work. Anyone had this problem before>? Second thing: I get a WARNING:[1209214400]: File dsp.c,

Hi, At one location we have a user whose Polycom IP430 suffers from erratic reboots. We swapped his phone for a brand new one, but the problem remains. Has anyone seen that?

Hi, I've just installed DAHDI at two PBXs as follows: *PBX-1 PBX-2* FXO ------------- FXS When I try to send calls from PBX-1 to PBX-2 I just receive the message: "Starting simple switch on 'DAHDI/1-1" It seems like if PBX-1 couldn't send DTMFs, but when I set immediate=yes at chan_dahdi.conf at PBX-2 dialplan is executed at PBX-2 but nothing is heard at

Hello list, Hope you are all doing fine! I have stumbled over some piece of dialplan code in which apparently they were trying to avoid recording the DTMF tones in the wav file. It is really messy and I am not sure if this really works. So after a bit of research I found this comment ( https://community.asterisk.org/t/asterisk-dtmf-record/65040) in which it is said: *"Asterisk strips the

OK, I got my Cisco 7960's to auto-answer on the second line but I can't get the Asterisk to call all the lines at one time. I have 4 phones I would like all of then to answer when I dial x300. Any help would be great Thanks Tuska extensions.conf [conference] exten => 300,1,AGI(callall) exten => 300,2,MeetMe(300,dtqp) ; press # to exit the conference exten =>

Hello Folks, I am testing Asterisk 1.4.2 running on Fedora Core 5 (version 2.6-17). I have loaded the app_meet.so module in order to activate the MeetMe, MeetMeCount and MeetMeAdmin applications. While I have been successful in loading the app_meet.so module, I am experiencing an immediate kernel panic every time I try to make a call to a room conference. Is this story unique to me? How can

The meetmeadmin() dialplan function lets you specify a user to mute, un-mute or kick. But how do you get a list of users in your dialplan? When a user joins a conference, the user number assigned is "the last user number +1." If you have a long running conference with callers joining and leaving all the time, this can grow to be a large number. I want to be able to

Hi people, I want to make a meetme between 2 numbers. First I enter the number1 into the meetme. It is waiting for the other number, but the other number never entered, so, how can I finish the meetme from the dialplan?. Is it posible by using MeetmeAdmin and kick all the users? Thanks, Anahi Ludue?a _________________________________________________________________ Descubre