similar to: HoldMusic Beep

Displaying 20 results from an estimated 1000 matches similar to: "HoldMusic Beep"

2007 Feb 27
1
Billing Telephone Number (BTN)
I have Asterisk setup with two PRI's one going to my telco and the other going to a Norstar Meridian system. The Norstar Meridian is sending a BTN number that needs to be passed to the Telco. Is there a way to pass the BTN as a variable in the dial plan? Like CallerID(num)? What is the variable for BTN if so? Many Thanks. -- *** Forrest Beck IAXTEL: 17002871718 jonforrest.beck@gmail.com
2007 Jul 10
0
G722 and Polycom 550
Has anyone found a way to enable the g722 codec as a prefered codec in the Polycom provisioning files for the 550's? I couldn't find a pref for voice.codecPref.IP_550. What needs to be put into the allow field (sip.conf) for asterisk to allow the codec? -- *** Forrest Beck IAXTEL: 17002871718 jonforrest.beck at gmail.com http://www.shift8.biz
2007 Jul 15
1
TimeStamp a Recording
Has anyone come up with to timestamp a Recording? I am using a pretty simple dialplan to record a audio file for a hotline. I'd like to store the date and time it was recorded somewhere, Ast DB or MySQL DB. Then when the audio file is played back to a caller, the system will say something like. This message was recorded January 14th at 10 42 pm Thanks for any ideas you may have. -- ***
2007 May 03
3
SIP RealTime Friends
I setup sip realtime. Is it possible to use a type of friend? User and Peer seem to work fine. -- *** Forrest Beck IAXTEL: 17002871718 jonforrest.beck@gmail.com
2007 Feb 08
3
Automatic Dial, Play message
Does anyone have some method, or AGI scripts that will automatically call a list of numbers from a database and play a pre-recorded message? For example, you have a database of FirstName LastName PhoneNumber Jon -- *** Forrest Beck IAXTEL: 17002871718 jonforrest.beck@gmail.com
2007 Feb 25
1
Marks SNMP HowTo
I followed Marks SNMP howto on Voip Magazine and ran into a small problem... (http://www.voip-magazine.com/content/view/2877/0/1/3/) When asterisk is running as a non-root user (asterisk) SNMP request for for the Asterisk MIB tree return nothing. If I quit asterisk and run it as root, all is fine. Does anyone have a idea what is going on? I have never used agentX, so I am unsure of what it is
2007 Mar 30
1
Paging
First off, A lot of thanks to this list. I have learned ton from reading through the posts this past year. I need some advise. I have two group of phones connected to a single server. Group1= SIP/2503&SIP/2504 Group2=SIP/3501&SIP/3502 I'd like to be able to dial an extension and page a certain group of phones only if ChanIsAvail returns 1. I am not sure how to go about
2007 Apr 04
1
Polycom
I know this doesn't belong on this list but... I am looking to see if anyone is using Polycom and knows of a web based software for creating/managing the cfg files for polycom phones. I see that the AsteriskNow will add provisioning support for Polycom phones. Since it is still in beta, I was just looking to see if there was anything else out there. Thanks! -- *** Forrest Beck IAXTEL:
2007 Sep 20
1
Paging MEETME_RECORDINGFILE Variable
I am having a weird issue with setting the recording file for the Page app. Here is some quick background info I have a macro that pages all my phones: [macro-pageall] ; Context for paging all devices. ; This will search the sip table in the realtime database ; for all phones that start with a number. That number is ; passed to this macro as ${ARG1}. ; ; ARG1 = The
2007 Apr 19
3
Outgoing CallerID
I am not sure of the best way to do this, so I thought I would query the list. I have about 100 internal extensions ranging from 2000 - 2100. Each internal extension has a external DID number. For example: 2001 = 5552871620. As you can see the internal externsion and DID don't match in any way. What would be the best way to set the DID for when a extension dials out on the PRI? In
2007 May 03
2
zttranscode crashes server
I was just looking to see if anyone else has seen this problem as well. When asterisk starts up it loads the zttranscode module. The problem exist when I use the init scripts to stop asterisk and then use the zaptel init script to unload modules. Since the zaptel init script didn't load the zttranscode module it will error out when trying to unload the modules. I built
2007 Mar 01
2
Asterisk 1.4.1
Any idea when 1.4.1 will be available. There is a bug fix in the cvs head that I need, and I don't want to run the cvs build on a production machine. Thanks... -- *** Forrest Beck IAXTEL: 17002871718 jonforrest.beck@gmail.com
2007 Mar 26
1
Server Recomendation
I am looking to install a system with 200 phones (polycom). There will be about 30-40 simultaneous calls. I am looking at the Dell 1950 with Quad 2.66, 2Gig RAM, Two 160 Gig SATA Drives (Mirrored with a Perc5 card), Dual Gig NIC, and RHEL 4.0. I will use two "gateways" for my PRI's and FXS Cards so PCI won't be used. I will probably use a small 14" 2U server to handle
2007 Mar 27
0
Macro Dial - External DID
I am using the sample (slightly modified) macro for dialing phones. My extensions are in the 2000 range. Each extension has it's own external DID. Is there any way to have the macro look up the DID to be used for the extension and set the DID as the callerid? Maybe from a flat file somewhere? Or is there a better way to do this??? I know I can use callerid in sip.conf, but I only want the
2007 Apr 24
2
Voicemail on Different Server
I have two seperate systems at two different locations. Each hosts there own voicemail for their phones. I have thought about just having all voicemail on one server. Is the best way to do this just through a dial app? For example, if someone dials 1000 to check voicemail at site A. The dialplan will be something like this on Site A: [context-for-phones-at-one-location] exten =>
2007 May 05
1
ODBC
I am trying to compile asterisk with ODBC support on CentOS 4.4. I am running into the same issue as documented in this bug. http://bugs.digium.com/view.php?id=8214 The server is a Dell 2950 with Dual Core /64bit processors (2Gig RAM). I tried creating a symbolic link link mentioned in the bug report, but didn't have any luck. Any one else had this issue? What did you do to get around
2007 May 08
3
MYSQL Query --> PAGE
I have all my SIP users in a realtime database. I would like to use MySQL command to query the database and use the results from the query to page all the phones found in the query. The results from the MySQL query will be multiple rows of extension: Something like: mysql> Select extension from sip where extension like '6%' 6001 6002 6003 ex.... I need to put all the results into a
2007 Apr 11
5
What is your Backup Strategy?
I was just curious to what your redundancy solution is. I have considered many options, so I thought I would share and get an idea for what others are doing. My setup is two different locations with a 10MB WLAN fiber link between the two. Each location has it's own PRI as well. I have considered and tested many options this last year or so. 1) Using hearbeat and drbd to monitor the
2006 Oct 31
1
Strange Characters in CLI on TTY9
When I look at TTY9 (using init.d and safe_asterisk to start the asterisk process), I am getting some strange characters. When a application is run the and the CLI shows the application executing the languange almost looks russian...?? Anyone seen this before? http://picasaweb.google.com/jonforrest.beck/AsteriskCLI
2004 Feb 02
1
Norstar Integration with Asterisk via FXO or BRI ISDN
Hi, I have a legacy Norstar system that I'm looking into integrating with my Asterisk setup. My first attempts have worked, which involves a Wildcard X100P FXO card in the * box connected to the Internal ATA (FXS port) on the Norstar system. Calling from SIP -> Norstar works fine, since the SIP caller initiated the call and generally will be sane enough to hangup the phone when