similar to: Call Recording Servers

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 am trying to set up a Conference room where users are put on hold until the host arrives. I have figured out that the A option activates marked mode and the w option is used to activate the waiting until the marked user arrives. This seems to be what I need. What I can't seem to find is how do I mark a user? Thanks _____________________ Kevin Savoy Business Unit Telecom Analyst 2218 4th

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

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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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

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

I'm trying to make H.323 trunk between AVAYA&Asterisk. But call from AVAYA is terminated inmediatelly when apps DIAL on Asterisk is started. Does any one use AVAYA and h.323 channel? Thanks Bob.

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

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

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

This worked. Great and thanks -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Carlos Chavez Sent: Wednesday, February 07, 2007 5:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] After upgrade to 1.4 transfers don't workproperly On Wed, 2007-02-07 at 14:12

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Hi, Did anyone manage to connect either Digium or Sangoma T1 card to any other PBX/gateway using T1 E&M Wink/Wink signaling? I'm trying to connect Avaya Definity to an Asterisk box with T100P and so far no luck. (I know I can do so with ISDN PRI, but need an additional ISDN processor card for Definity.) I tried to connect Definity to Cisco 3640 CCME (call manager express) to test the link

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

Running Samba 2.0.3 on a Digital Unix 4.0D system. All functions of the new Samba version appear to run properly (previously ran 1.9.15) with exception of users being able to change their own Samba password (root is only able to change password on behalf of users). Following is permissions assigned to smbpasswd: -rwxr-xr-x 1 root system 901120 Mar 31 16:06 smbpasswd Following is

I've loaded Asterisk 1.4 with the addons 1.4, libpri 1.4 and Zaptel 1.4 as well. I can place calls but I noticed the MySQL was writing out to the database. When doing an Asterisk load with asterisk -vvvv I saw the following: [Dec 26 11:02:08] WARNING[10029]: loader.c:375 load_dynamic_module: Module 'cdr_addon_mysql.so' did not register its [Dec 26 11:02:08] WARNING[10029]:

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 could only get *ANI*DNIS* working one way and that was setting my signalling type on the Asterisk side to 'featd' The Definity won't send *ANI*DNIS* information back to the asterisk as far as I can tell. Other than that, I've been running it with wink/wink E&M for a while now. TN464 circuit pack on the definity side, te110p on the asterisk side. Just a crossover cable

Well the addons from 1.4 are installed. This original Asterisk 1.2.x box was created by my predecessor and he had the cdr_addon_mysql.so and res_config_mysql.so files on a server that we copied to any new installation. I'm not sure where he got these files. As far as I can tell shouldn't the install of the addons create these files? If not where do I get them from? I've done a search