similar to: Conference Leader

hi, Can anybody pls tell me, how to increase the time gap between 2 digits when you transfer a call. ie, the operator answers the call, and presses hash key to transfer, and then enters the extension number, some times, it timeouts too quickly before the operator enters the whole extension number (may be bcos the operator is slow). I tried the following, but it doesn't seems to be helping

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

Hello, I have a E100P card from digium and I try to implement a conference bridge in asterisk. I wonder since I got the E100P card do I still need to load ztdummy for caller from h323 endpoints to work with Meetme? I load the E100P driver but i did not load the ztdummy driver. My h323 caller does not hear any voice play by Meetme. Looks like ztdummy is required as long as h323 is concern and

http://bugzilla.mindrot.org/show_bug.cgi?id=980 Summary: sshd does not write the session leader pid to utmp when priv-separation is enabled Product: Portable OpenSSH Version: 3.9p1 Platform: All OS/Version: All Status: NEW Severity: normal Priority: P2 Component: sshd

Hello all, Can I execute linux command like(ls, mkdir) through the Manager interface? I can't seem to access the manual at digium.com. I keep getting 'Forbidden' error. Looks like they are upgrading or something. CF

Funny - I downloaded the latest Asterisk CVS, and it's pretty much working. Will report when I have some more success. PaulH -----Original Message----- From: Philipp von Klitzing [mailto:klitzing@pool.informatik.rwth-aachen.de] Sent: Tuesday, 28 September 2004 9:46 PM To: Paul Hales Subject: Re: [Asterisk-Users] Leader IP10S Hi! > I have been lent a Leader IP10S phone (SIP) for

Hello, Is there a asterisk developer guide/source code doc or something like that? I want to see if I can implement the admin menu function for meetme. Foong -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030922/3ff8a388/attachment.htm

My ctdb version is 4.17.7 Hello, everyone. My ctdb cluster configuration is correct and the cluster is healthy before operation. My cluster has three nodes, namely host-192-168-34-164, host-192-168-34-165, and host-192-168-34-166. And the node host-192-168-34-164 is the leader before operation. I conducted network oscillation testing on node host-192-168-34-164?I down the interface of private

Hi All, I observed the following in my try towards Multiparty Conferencing. I am establishing the Multiparty Conferencing through Asterisk Manager API. I have two users SIP/111 and SIP/101 of which SIP/101 is treated as leader. Following commands are used - Action: Originate Channel: SIP/111 Application: MeetMe Data: |edwx ActionID: ffe4563 When I use the above, Incoming call will

I wonder if anyone can suggest ways to diagnose an infuriating problem being experienced by customers of a company I did a large Asterisk project for. First some background: The system is a conferencing system using a modified MeetMe. There are seven Asterisk boxes (we call them bridges) each with four T1 PRIs into a TE405P. No VoIP is involved. A conference is always local to a single bridge.

Hello everyone Agile (XP), informal (no-tie) software house where PHP, Ruby (and Ruby on Rails) are actively used are looking to hire a development team leader that oozes strong leadership and management ability. We''re looking for someone that is able to talk openly but firmly as the challenge will be to provide cohesion between development and commercial and unify the development team

Jesus, I hope it isn't true. But if it is, we will miss him dearly, both here and very much so on the Samba list. However I can under stand and respect the decision. Good luck and God speed Jeremy. Jerry, can you pick up the slack?? An ill wind blows for us all as a result of the MS deal.... -- David C. Rankin, J.D., P.E. 510 Ochiltree Street Nacogdoches, Texas 75961 (936) 715-9333

Hi?Danny, Thank you for your prompt response. The way you are suggesting is great .?Infect?asterisk have its own functionality that if user presses *1 during meetme conferencing asterisk automatically unmute that user and user comes in talking mode.But it is not?fulfill my need. There is and issue that if 3-4 user presses *1 at the same time than how can i decide that who is asking the question

I have a model “Conversation” and a model “Leader”. A Conversation is always led by exactly 1 Leader. I’ve overridden Conversation’s “validate” method to validate the presence of an associated Leader model. If I add a similar validation to the Leader model, however, Conversations can no longer be saved on creation because the Leader model is invalid. On creation of both a new Conversation and

Hi, I was just wondering how would the application be where the Asterisk calls a number and that number joins the conference as soon as the call connects. There would be only one conference already defined in meetme.conf and there is one person already joined the conference. Currently MeetMe requires a person dialing into it and the joining the conference. How could this be done using MeetMe or

hello, my questns are about few * functionality. 1) how can I make call tranfer. Not call parking. If I'm talking with some one a I want to tramnfer call to the another extension, to the other person. 2) how can I make call confernece. Not Meetme If I'm talking with some one and I want to join another person to our talk . I haven't found this in any manual :( hudecof -- mail:

Hello, I have a Digium TE410P card. I get the "knight rider" lights before the module (wct4xxp) loads, but after the modules are loaded I don't get any lights. I have found the following 2 posts but still could not solve the problem http://lists.digium.com/pipermail/asterisk-users/2004-November/075277.html http://www.voip-info.org/tiki-index.php?page=Asterisk+TE410p+No+Interrupts

Hi, I'm struggling with a feature in my home phone setup. I have several phones using both SIP and SCCP. What I try to do is to create a dynamic feature that works similar to the blindxfer feature built into Asterisk. What I want is the possibility for the called part to push a number sequence (for example *#) to redirect the callee to a fixed extension or (for example *123#) to redirect the

Hello, Does asterisk supports VXML? Couldn't find much resource on that on google and wiki. Thanks Foong

Hello, I have 5 digium's g.729 codecs and succesfully register with asterisk, I have incomming call from my cisco AS5300 to Asterisk through IP. But Asterisk always use g711 ulaw instead of g.729. When I disable all other codecs other than g.729 in both cisco and asterisk, calls get dropped once connected. The codec list show on my cisco AS5300 for g.729 are: g729r8 g729br8 I suspect that