similar to: asterisk 1.8.10.1 meetme

Hello, I'm having issues with my phone Polycom sp450 not subscribing to Asterisk server. Asterisk server is fine, firewall is not the issue because a secondary phone is working fine, my connection to the server is fine too, any ideas or suggestions are welcome. -Motty -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, is it possible to detect 4 length pattern busy cadence detection on FXO lines in 1.8?? Here the tones are: 425Hz Pattern(0.2ms on, 0.2ms off, 0.2ms on, 0.6ms off) in asterisk 1.4 busy detect worked in asterisk 1.6 didn?t work and i was told that 1.6 can?t handle 4 length patterns, but what about 1.8?? for now I can only hangup by asking the provider polarity switch. Thanks best

Hello All, I upgraded Asterisk 1.8.10 to Asterisk 1.8.22 since upgrading I can't get meetme feature to work when dial meetme extension, can you please help? It always worked before, also I do not have dahdi installed on this machine, never did. -- Executing [104 at sipphones:1] MeetMe("SIP/101-00000813", "104") in new stack == Parsing

Hello, is there anywhere an overview of SIP error codes and under which condition they are reported by Asterisk? There are general definitions for SIP error codes, but they are quite general and it's Asterisk that actually checks what's wrong and then reports an error. Now, currently I could check the source code to get more informations what could have caused the error, but that's

Hello, I noticed Asterisk 1.8.4.1 execute number dial twice Log == Using SIP RTP CoS mark 5 -- Executing [912066604 at sipphones:1] Set("SIP/4773-0003e920", "CALLERID(num)=2066604") in new stack == Extension Changed 4773[sipphones] new state InUse for Notify User 4701 -- Executing [912066604 at sipphones:2] Dial("SIP/4773-0003e920",

I'm trying to use the password entered with Authenticate to create dynamic meetme conferences with the following dial plan: exten => _XXXXXXXXXX18467,1,Authenticate(/etc/asterisk/meetme.pw|a) exten => _XXXXXXXXXX18467,n,MeetMe(CDR(accountcode)) ; 281-8467 However CDR(accountcode) is always being set to 1022 no matter what password is used. The passwords are stored in a file so they can

hi guys: recently I want to buinding a meeting confence on asterisk and use the meetme application. I have a ztdummy kernel afteri the lsmod commond: ztdummy 5768 0 zaptel 182660 28 zttranscode,ztdummy crc_ccitt 3008 1 zaptel I also configure the meetme.conf conf => 1000; my extensions.conf [default] exten =>

Hey Guy! I have following dialplan for meetme and i want if someone type wrong meetme extension it should say invalid extension. But look like following doesn't work. its just hangup if i type wrong number. how to fix this code.. ;Conference rooms/lines: exten => 7580,1,Goto(ivr-meetme,s,1) [ivr-meetme] include => meetme exten => s,1,Answer() exten => s,n,Wait(1) exten =>

One of the features that I thought would be popular with the Web-MeetMe suite is the ability to start all non-admin callers in a muted state and selectively unmute them. For example any large conference that is of an announcment nature with a Q&A session. It's probably a feature I should have tested better, but I just discovered that a caller that is joined to a MeetMe with the |m flag

Hi, If there are 5 asterisk servers on the local net and each server runs meetme, eg. 3311,3321,3331,3341,3351 respectively. Can I connect these 5 meetme conferences to one meetme using IAX2? Regards, Zen

Hi, Is there and trick to getting a conference room up and running.. I have 'conf => 7500' in the meetme.conf file and 'exten => 7500,1,MeetMe(7500)' in the extensions.conf file (in the same context as my phone extensions).. When I dial extension 7500 I get the voice saying "That is not a valid conference number, Please try again.." <beep> <beep>

Hi, This is pretty much a repost but I still havent been able to get MeetMe to work.. I am using a Dev Kit lite.. so that should satisfy the Zaptel requirement for MeetMe.. meetme.conf looks like this.. [rooms] conf => 7500 In extensions.conf I have an [extensions] context and within that same context I have the line.. exten => 7500,1,MeetMe(7500) When I dial 7500 I get the message

Hello list, I wonder if someone might be able to clear up something for me. I recently set up asterisk and have now managed to get the MeetMe application up and running. When I dial the extension to access the conference/MeetMe application, the only prompt I hear is:"You are currently the only person in this conference." When I use a friend's newly installed asterisk, I hear:

I do not have any zaptel hardware on the Asterisk box, I could not have meetme functioning. I did modify the Makefile in zaptel directory on line 168 by including ztdummy as one of the modules to compile in. The error message from the concole: -- Executing MeetMe("SIP/1002-e9ca", "4700") in new stack == Parsing '/etc/asterisk/meetme.conf': Found

Hi! i have an error using the meetme aplication, and just dont work.. my meetme.conf is: [rooms] conf = 700 i calling from a sip phone, the extension number is 600. there is the error: Executing [700@numberplan-custom-1:1] MeetMe("SIP/600-09111e58", "700|MI") in new stack WARNING[20055]: channel.c:3024 ast_request: No channel type registered for 'zap' WARNING[20055]:

Is there any way to have MeetMe require a password normally, except when dialed through a certain context or with a certain flag? As far as I can tell, there is no flag to MeetMe() to disable PIN checks the way you can do with VoiceMailMain. This would be a very useful feature, as through context usage you could require a password from "external" users but not require them for

I'm looking into deploying * for an internal conference call server (using MeetMe) and had a couple of quick questions for those of you who have used it. I checked the Wiki but there weren't a lot of details for MeetMe. - Can you limit the size of a conference "room", ie max 8 people, etc. - Is there a list somewhere (besides the source ;) that has all the commands availible to

While researching Areski's new Web-MeetMe management gui, I found some odd (from what I expected) behaviour). Using the CLI to set un/mute status works but does not update the flags, or so it appears. Starting with a fresh conference (1 user) *CLI> meetme list 3456 User #: 1 Channel: OH323/R61 Using the CLI to mute the caller (no change in the user status0 *CLI> meetme mute 3456 1

We are developing an on-demand teleconferencing solution. We will be billing per-minute/per-user. I've successfully gotten Asterisk to write CDR data to a postgres database, but with the way I've got things setup right now the CDR does not have the dialed conference number. We need this information in order to be able to bill. As teleconferencing is the only application of the

Hi, Is it possible to create a meet me room on the go through dial plan? I am looking to use AMI Originate to drop a call into meetme room and once it's proved that party is joined, play him an announcement, grab few numbers from them, and then dial a second number and drop into the same meetme room. The reason to use this is to be able to know when the channels connected because both