similar to: Time Limit on Call or Conference Room?

> On Fri, 3 Aug 2007, JR Richardson wrote: > > > Can anyone point me int he right direction? > > At the risk of coming off in a gratuitiously self-aggrandising manner > quoting myself: > > http://lists.digium.com/pipermail/asterisk-users/2007-May/188438.html > > -- > Alex Balashov Thank you, Alex. As I've said many times, this community has the

Hello Conference again. Meetme can now limit number of users in a room. Can it also limit how long a conference session? Someone ask the same question (from achive) but doesn't have a solid answer. Foong -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030821/c1ca1383/attachment.htm

Here is what I am trying to do. I have a SIP soft phone running on a PC that is streaming a local radio station. I assigned "mono out" in XP Equalizer as the mic so now I have the softphone streaming audio. I then create a conference room and dial that room from the softphone. Now anyone who joins the conference room hears the streaming audio. How can I configure Asterisk so that

Albert Graef wrote: > The broken mingw support (as pointed out by Stuart) [...] s/Stuart/Kenneth/ Sorry. -- Dr. Albert Gr"af Dept. of Music-Informatics, University of Mainz, Germany Email: Dr.Graef at t-online.de, ag at muwiinfa.geschichte.uni-mainz.de WWW: http://www.musikinformatik.uni-mainz.de/ag

Hello, I have MeetMe rooms generated dynamically and it always have two people inside that are entered by dialplan. I wish to make in some way a timeout mechanism that after X amount of time, it will disconnect the users and kick them out of the conference. How can I do such thing ? Thanks, Ido -------------- next part -------------- An HTML attachment was scrubbed... URL:

Óscar Fuentes wrote: > Some points you mention on your web page are solved. Which ones? (Just curious.) > Others are not applicable to LLVM. That might be the case now, but the lack of even basic functionality in some areas (in particular, no advanced feature checks, no make dist/distcheck, no make uninstall, lack of useful trace options when something goes wrong during a build, arcane

Hello, We are trying to send faxes by T.38 protocol to a remote SIP proxy from a local extension. The local extension sends the INVITE, Asterisk sends the call to the Proxy the call is connected with a regular audio codec. After a few seconds the remote proxy sends an INVITE with UDPTL and the Asterisk sends it to the local extension and it's accepted, but (here the problem starts) just

In some PBSx you can limit outgoing call that you cannot speak longer the 15 minutes. Is it possible to do with Asterisk ? Bart -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031113/6eb392cc/attachment.htm

Hi everyone, I'm new to Asterisk, but I like it ;o) Have a question to you; How can I limit the incoming call duration? -- Suich

I've googled and not quite found what I need, so... I have a conference room phone that I would like to make behave as follows: - when a call comes to that extension: answer the call put the call in a static MeetMe room with option 'w' ring the phone by SIP and when the phone picks up, put it in the same MeetMe room as the marked call. if subsequent calls come in, they are put

Hi All, I'm very interested in real world experience of double digit number of users sustaining good quality audio in a single meetme conference. Personally, I have seen 23 users in one conf room, all coming in SIP, ULAW. Server is 3.2GHz proc, 1Gig RAM, 1-2 % proc utilization under 23 user load, perfect audio. I'm working on a conf bridge for 150+ users, could use some advice, if

Hi All, I have 1.4.9.1 setup, with the compiler flags enabled for T38, and have a Mediatrix 2102 and a Linksys SPA 8000-G1. I can pass faxes between devices but can't seem to invoke T38 pt UDPTL. It's enabled in sip.conf [general] and well as the [peer]. I get an error at the CLI: WARNING[3096]: chan_sip.c:14149 handle_request_invite: RTP re-invite after T38 session not handled yet !

Hi, All Is there a provision for "AbsoluteTimeout" application to notify called and calling party of the reason why the call suddenly ended? This way, the parties will be much better informed, hence they will/should not think that their VOIP/telco provider(s) are providing bad service. Ta SJ

Hi All, I have a Cisco 2600 PRI gateway being hosted on an Asterisk server. The PRI on the cisco is pointing to a customer legacy PBX, the SIP VoIP side of the cisco is pointing to an Asterisk server (1.2.X). In Asterisk, the SIP peer is setup with callerid="some name"<5551212> In a SIP call from the cisco to asterisk, there is no CID name info in SIP debug, so Asterisk

Greetings, I have created simple conferencing solution before using meetme application, but this times its a little tricky. My client needs a functionality to call multiple extensions to join a conference room. Extensions will ring like in a ring group, and on pick up, user will be either automatically added to the conference room, or maybe I'll program them to enter 9 to accept and 8 to

hi; i setup a Meetme conference room and i notice the following behavior: if A calls confroom over PSTN channel 1 B call confroom over PSTN channel 2 C calls confroom over SIP/Ethernet then i have all of them talking and the media stream mixed by asterisk. However, if i hang up A, channel 1 is still ocuppied (i try dialing inbound again on channel and it continues to give a busy siganl) any

?? Hello.? We are very successfully using asterisk (in the form of trixbox 2.2/asterisk 1.2).? We run a few conference lines for customer teleconferences which mostly work well but they seem to operate at half duplex.? If a person starts talking they will cut off others on the call.? Is this normal behavior?? Are there any options I can change to change this? ?? Thanks! James -------------- next

Hi All, I have a pretty standard Asterisk PBX setup with 60 SIP Peers, mostly Polycom 501's and a receptionist phone, Polycom IP 601 with 3 attached sidecars and Buddy Watch enabled monitoring all other SIP phones. The problem occurs when a group (all SIP peers) Page is called. Not always but sometimes when the Page is executed, the IP 601 will become unreachable from Asterisk. So when the

I have been trying to figure a way to SendDTMF into a MeetMe room using the Manager API. I can't redirect everyone into another context and then bring them back because that would mess up my logic. I am trying to use local channels and the originate Action to accomplish this. Exten: 3441115 Priority: 1 ActionID: actid-00000001 Context: senddtmftones Action: Originate Channel:

Hi all, I didn't find yet any info about this. Is there any way for a Conference Room Owner to change his own password? A kind of Menu like calling his conference room: example:8200 And an IVR option to change password. That seems to me interesting, because i may not want the same users entering two diferent days on my conference room... Also I don't think it is a good choice to