similar to: Multicall

Hi all, I have a question concerning ActionWebService XMLRPC servers: Is it possible to send multicall requests to the Web service? I tried to use multicall and get the error message: no such method ''system.multicall'' on API [MyAPI] In Changeset 2021 there is the following commit message: add ''system.multicall'' support to XML-RPC. boxcarred methods must

Hi guys, I need to realize a sort of automatic call monitoring dialplan. This is exactly what I need: - A user originate a call - When the call is bridged (or just before) I need to invite automatically a third party to the conversation that should hear the audio channel but not speak (it's a monitoring application for a callcenter) The person in charge of monitoring cannot use ChanSpy or

Hello, I am running asterisk 1.4. For argument's sake I have 4 telephones. 2 support video, 2 do not. Calls between phones work fine and codecs are properly negociated. I have videosupport=yes in sip.conf and when the two video phones communicate I have video. I call meet me with this command EXEC MEETME 1234|d SIP looks like this : -- AGI Script Executing Application: (MeetMe)

This buffer overflow was introduced with 91e0c5f3dad47838cb2ecc1865ce789a0b7182b1 (2.6.24). Cc: Konrad Rzeszutek Wilk <konrad.wilk at oracle.com> Cc: Jeremy Fitzhardinge <jeremy at goop.org> Cc: Thomas Gleixner <tglx at linutronix.de> Cc: Ingo Molnar <mingo at redhat.com> Cc: "H. Peter Anvin" <hpa at zytor.com> Cc: x86 at kernel.org Cc: xen-devel at

This buffer overflow was introduced with 91e0c5f3dad47838cb2ecc1865ce789a0b7182b1 (2.6.24). Cc: Konrad Rzeszutek Wilk <konrad.wilk at oracle.com> Cc: Jeremy Fitzhardinge <jeremy at goop.org> Cc: Thomas Gleixner <tglx at linutronix.de> Cc: Ingo Molnar <mingo at redhat.com> Cc: "H. Peter Anvin" <hpa at zytor.com> Cc: x86 at kernel.org Cc: xen-devel at

Hi, the following szenario leads to a problem: I connect an CAPI channel to an AGI-Script per Manager API. This Agi script starts the MeetMe-Application. The Person on the Capi Channel is now able to speak with the other conferess in the MeetMe-Room. But if the CAPI channel hangs up, the busy tone is streamed into the MeetMe Room for several seconds, until the CAPI HANGUP-Signal is finally

We want to test capacity of our MeetMe room. The thing that is distinct about this is that the incoming line is being delivered IAX2 to our server across the net - so Telephone -> VoIP Gateway -> MeetMe. We want to test both the VoIP Gateway and the MeetMe room performance. You can reach our MeetMe room directly at 1-301-561-9229 If you want to test with us we're thinking maybe 9pm

Hi, We're running on a * 1.4.21 system. We run about 80 SIP Extensions, mainly ATCOM phones (and a few Snoms - 300 and 360), and have an additional 80 IAX2 extensions to iaxmodem devices for fax2email. We are rapidly growing and will be adding an additional PRI trunk and grow to about 150 SIP & IAX2 extensions towards the end of the year. We have two Digium Wildcard TDM800P cards (8 x

I try to use meetme app after reading manual i compile and install zaptel with ztdummy when i make lsmod i have ztdummy 2532 0 (unused) wcusb 20064 0 (unused) zaptel 179168 4 [ztdummy wcusb] usb-uhci 26348 0 [ztdummy] usbcore 51616 0 [wcusb usb-uhci] after it i recompile asterisk and after it i have

I'm having some problems getting meetme to work now that I have upgraded to .5 I am able to conference calls but every time I try to manage the conference through meetme it just says "No users in this conference" Any ideas why it doesn't see the conference call? Thanks for any help! Jason This message along with any attachments is intended only for the use of the individual or

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

Dear Users, Thank you all for a great Forum. In the past with my analog PBX I had 6 separate telephones configured auto pickup and they where connected to the meeting room's microphone. When we had a meeting, it was then possible for 6 members simultaneously to call in with a normal landline and be transferred to one of the phones. In this way he could be part of the meeting. Is it possible

Hello-- What I'd like to do: Use IAX softphones running on computers, in Auto-answer mode, with sound going to speakers, as a sort of public announcement system. What isn't working: Well, my first experiment was to set up the MeetMe system described on the Wiki... This works fine for voice announcements. You pick up a phone, dial the right extension, and an agi is fired up to put files

Hello System: Debian etch with kernel 2.6.18-4-686 or 2.6.18 custom. Asterisk Version: SVN-branch-1.4-r55483M Zaptel Version: SVN-branch-1.4-r2302 modules all ok in compilation time. And modules loaded: ztdummy 5928 0 rtc 13364 1 ztdummy zaptel 181540 1 ztdummy crc_ccitt 3200 1 zaptel In /dev/zap directory I have:

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

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 =>

Hi, We are using Vicidial and sometime even when agent disconnects, outgoing call originated by dialer is still active. Since call was initiated by dialer and then bought into meetme conference of agent and we can't corelate this call to any agent channel. When agents are dialing, channels doesn't show calls vicidial2*CLI> show channels Channel Location

I've got an Asterisk (v1.2.11) installation running, but it doesn't seem to have the Meetme() app. At the CLI, I type Meetme , and it responds No such command 'Meetme'; meetme doesn't show up in CLI show modules . I'm running a SIP-only server at a datacenter where I can't add Digium (or any other) HW, and am running under CentOS. There is an /etc/asterisk/meetme.conf

Hi i have a similar dialplan: exten => 99,1,Gotoif(....?2:3) exten => 99,2,Meetme(100) exten => 99,3,Meetme(100|options) i'd like to do something like: exten => 99,1,Gotoif(....?2:4) exten => 99,2,Meetme(100) exten => 99,4, ... exit ... exten => 99,3,Meetme(100|options) How can i exit the dialplan? I won't use meetme twice! Thanks nik