similar to: MeetMe Conferencing and Marked Mode

Not sure if this can be done or not, but I can't seem to find it anywhere on the Wiki. When dialing interoffice with Asterisk 1.4.2, I would like to have the caller id of the person I am dialing displayed and not the number I just dialed. Is this possible? So, if extension 4023 is John Doe, and I dial 4023, my display should read John Doe and not 4023. I am using a Polycom 501 by the way in

I was able to get it to work with 2 extensions. One for the "host" and one for the "participants" Not the best way to set it up but it works. Thanks -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Savoy, Kevin - Williston, ND Sent: Wednesday, December 13, 2006 8:06 AM To: Asterisk Users

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I have provisioned a bunch of Polycom 301 phones to get the config files from my ftp server. Out of the 4 phones 2 get the config file however the other 2 cannot contact the boot server. I have reboot the phones a number of times remotely (the client is 400 km away) through vnc and logging onto the web config internally. No matter what I change on the web config page it is not saved. I feel I

Hi guys, I have recently installed a Asterisk Server with CDR Call Detail Records. I have installed it over a Asterisk 1.2 , but now It do not run . I have installed it with the following procedure: # yum install ncurses #yum install openh323-devel # yum install mysql-server # yum install mysql # yum install php-gd # yum install php-mysql # yum install mysqlclient10 # yum install zlib # yum

I am having an issue with 1.4 where we can't successfully transfer a call directly to a voicemail box. We hit "Transfer" on the phone and dial the mailbox number we want to send it to, My dial plan for this is: exten=>_*40XX,n,Voicemail(${EXTEN:1},u) The voicemail system picks up and starts to play its message and at this point. We should then hit "Transfer"

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hi, I am facing some issues while using MixMonitor. My extensions logic is attached below: exten => s,1,MixMonitor(${CALLERID(number)}-${TIMESTAMP}-${UNIQUEID}.gsm,b) in this extensions TIMESTAMP is not working in Asterisk 1.4. can any help me why TIMESTAMP is not working in Asterisk 1.4. regards, Asif

I have discovered an issue on my system after upgrading from 1.2.13 to 1.4. A call comes in on a T1 line and goes to a Polycom 501 SIP phone. I have confirmed this on multiple phones. When the called person answers and tries to transfer the call to another extension, the call successfully transfers, however the person answering the transfer cannot hear the person that called in, the caller. My

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

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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

So no one else is having issues with MySQL and 1.4? I'm the only one? -----Original Message----- From: Savoy, Kevin - Williston, ND Sent: Wednesday, December 27, 2006 2:09 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] cdr_addon_mysql.so did not register itself duringload Well the addons from 1.4 are installed. This original Asterisk

We are looking at using Asterisk as a call recording server for an Avaya VoIP S8700 system in a multi-site VoIP Call Center. All calls will be coming in to one location and sent out via VoIP to other call centers. What kind of specs should we be looking at purchasing for our Asterisk server to be record up 200-300 calls simultaneously? Linux runs in 64 bit architecture, but does Asterisk

I'm currently writing some code to support conferencing in Asterisk without using the Meetme application. The conference runs in its own thread and every new inbound or outbound channel that is created is passed to it. This thread runs the conference loop reading and writing frames to each channel. I'm writing this as if it were a bridge with more than two channels, and I'm not

Can anyone point me to a sample or information on using MeetMe like this? Conference room is set up with 2 PINs, one for the moderator and one for the participants. Participants get music until the moderator joins (to avoid wild, un-moderated tangents). Call is ended and all participants are kicked out when the moderator leaves (or the moderator can kick everyone out via phone keypad).

Ok so something is missing. I get the below for those two lines. checking for mysql_config... /usr/bin/mysql_config checking for mysql_init in -lmysqlclient... no I even installed the mysql-devel as Bradley Watkins suggested and still it says no. What do I need to make that say yes? Thanks -----Original Message----- From: asterisk-users-bounces@lists.digium.com

Sorry if this is a repeat but I didn't receive a copy of it so I'm not sure it actually posted. The below worked for normal transfers. Now here is another situation. When we try to transfer a call directly to voicemail it plays the voicemail message but we can't transfer the call. The only way I could get it to work was to do a conference and then drop out of that conference. My

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