similar to: exit from conference

Hi, Can anyone suggest a good motherboard for the T/E410P cards ? Coz it doesn't get inserted in the the regular P4 motherboards due to PCI slot (32 bit) .... Any suggestions. Regards Azher --------------------------------- Do you Yahoo!? Yahoo! SiteBuilder - Free, easy-to-use web site design software -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I have received a new card TDM400P revision E, from digium. When I tried to modprobe wcfxs it gave me the following errors: Freshmaker version: 63 Freshmaker passed register test ProSLIC on module 0 insane (1) 255 should be 2 Module 0: Not installed ProSLIC on module 1 insane (1) 255 should be 2 Module 1: Not installed ProSLIC on module 2 insane (1) 255 should be 2 Module 2: Not installed

Hi, I presently have 6 PRIs of IVR traffic that I am planning to migrate from Dialogic on SCO-Unix to Digium-Asterisk on Debian. Here is the general description of the traffic in question : - IVR system, 138 PRI channels (6 PRIs, multiple D-channel) - Some traffic from TV ads, so all traffic typically arrives within a few seconds - Up to 12-15 simul. outbound (external) conferencing calls for

Hi, I am using dovecot deliver as lda for sendmail. My problem is that when a user account is over quota, deliver program sends message in the logs, but the message is not bounced back to sender informing that their is disk quota problem at the recipient side. Further message remains in the mail queue. Details are below. Plz suggest how to figure out the problems: 1. Message should be

Hi all, I'm trying to get the MeetMe system to take a caller and announce to them they've joined the conference in addition to the other members of the conference assuming previous members of the conference >= 1. I can see where the meetme.c app actually processes it using the ast_pthread_create_background(&conf->announcethread, NULL, announce_thread, conf); function. The

Good day all I've installed asterisk with sip on my LAN,no special cards,if done sip.conf and extensions.conf and all work 100,I'm using x-lite as a client. I'm trying to do conferencing.What I did was to has out the meetme.conf looks like [rooms] conf => 9876 conf => 2345,9938 and extension.conf exten => 9876,1,MeetMe,9876 When I go onto x-lite and type 9876 it gives me

Hi All, I would like to mark a user so that all users other than marked user hear music-on-hold till the marked user joins the conference. I took a look at http://www.voip-info.org/wiki-Asterisk+cmd+MeetMe, but could not get sufficient info. I'm using meetme for conferencing. Could anyone point me to a url which has the configuration details using meetme. Thanks, Jagan

Hi, I am using dovecot 1.0 with squirrelmail. But when i use $imap_auth_mech = 'digest-md5' instead of plain, dovecot throws this line: dovecot: Apr 23 09:16:07 Info: imap-login: Aborted login: user=<azher>, method=DIGEST-MD5, rip=127.0.0.1, lip=127.0.0.1, secured and user can't login. Plz suggest. Regards -Azher Settings: ====== disable_plaintext_auth = yes auth

Hey! I am trying to configure the voice onference with MeetMe application for my internal users. I have my server and 4 clients on same LAN and following is my extensions.conf file: [globals] Ahsen=SIP/222 Tahami=SIP/444 Uzair=SIP/333 Wasif=SIP/555 [internal] exten => 1234,1,Macro(voicemail,${Ahsen}) exten => 4321,1,Macro(voicemail,${Uzair}) exten => 5678,1,Macro(voicemail,${Tahami})

Hellos, I am having issues with my meetme conferencing. When I dial the conferencing number, It hangs after a few seconds.I have read somewhere that I need to enable ztdummy, which I have done but still no changes. Here is my log ~=~=~=~=~=~=~=~=~=~= PuTTY log 2009.08.12 18:44:43 =~=~=~=~=~=~=~=~=~=~=~= -- Executing [1;36;40mMacro[0;37;40m("[1;35;40mSIP/1215-fc5b[0;37;40m",

Hello, I would like to ask, if there's presence support in Asterisk and how to make it work with Xten's Eyebeam client. I tried searching all the possible documentation, google, but I found only a note, that there's a module in SER, that supports the feature. Is there also support in asterisk? Any pointer to documentation describing this is welcome. One more question -- is there

> On Sun, 18 Mar 2007, Yehavi Bourvine +972-8-9489444 wrote: > >> Hello, >> >> >> On most SIP phones a conference call is done on the phone and is limited to 3 >> participants. Polycom phones has a configuration option to use a conference >> server instead of the internal conferencing feature. I guess I need some >> conference server; any experience

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?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

Hello, Our 500+ company is slowly moving away from our hosted conferencing solution to one I built a few weeks ago with Asterisk and MeetMe. When our Q3 conference call comes around, we will have the need to have approximately 300-400 users in this call. Obviously, all would be 'listen only' mode and only 1 or 2 two would be speaking as marked/admin users. Our conference hosting

Hellos, I am having issues with my meetme conferencing. When I dial the conferencing number, It hangs after a few seconds.I have read somewhere that I need to enable ztdummy, which I have done but still no changes. Here is my log ~=~=~=~=~=~=~=~=~=~= PuTTY log 2009.08.12 18:44:43 =~=~=~=~=~=~=~=~=~=~=~= -- Executing [1;36;40mMacro [0;37;40m(" [1;35;40mSIP/1215-fc5b [0;37;40m",

Hi, Thnx to all, actually I didn't red the installation instruction paper properly that came with the card, and it was written on it as step 4 to insert the power cable in the card. Now the drivers are loading without any error. Azher --------------------------------- Do you Yahoo!? Yahoo! SiteBuilder - Free, easy-to-use web site design software -------------- next part -------------- An

Hi, Can someone guide me, how to change the password of windows AD using a linux script. Regards -Azher -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean.

Hi, I have a query from a client that can he use the E100P card to terminate the 2Mbps bandwidth in a linux box, thus reducing the cost of cisco router ?? The other end is a cisco 2620 router with E1 VWIC-1MFC. Can anyone explain if its possible with Asterisk and further any configuration help. Applreciated. Regards Azher Amin ------------------------------- http://www.consulttech.com.pk

Use Snom phones. We have had around 6 participants, without problems. In theory you should be able to have around 12 people on a conference on a snom phone. Jon -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Yehavi Bourvine +972-8-9489444 Sent: 19. marts 2007 09:14 To: asterisk-users@lists.digium.com