similar to: Updating Asterisk and its use with MySQL

[2009-05-27 02:06:16.294] WARNING[6971] chan_dahdi.c: No D-channels available! Using Primary channel 24 as D-channel anyway! [2009-05-27 02:06:16.295] VERBOSE[6971] logger.c: [2009-05-27 02:06:16.295] == Primary D-Channel on span 1 up [2009-05-27 02:06:16.301] ERROR[6971] chan_dahdi.c: !! Got a UA, but i'm in state 7 I noticed the above error many days after this at around 2AM. This

Hello list, I notice on the wiki that it is possible to execute a macro or a gosub within the queue-command in asterisk 1.6.x 1. Does this mean the macro/gosub is executed everytime a queued call is answered by a queue member ? 2. I'm using asterisk 1.4.30. Is there a backport or other way to make use of this 1.6-functionality ?? Kind regards, Jonas. -------------- next part

Apologies if this has been asked before. Does anyone know if I can simply recompile * 1.4.34 over 1.4.24.1? Ie. perform an upgrade from 1.4.24.1 to 1.4.34 by just rebuilding the source files for 1.4.34 over the top of the existing 1.4.24.1 files. Obviously, I will need to keep my config files (and sound files etc) - so I'll back them up first. Also, will I need to stop * to perform this

Is there a way to inject sound from a sound file into an established call using AMI? I have an established call from which I can record either or both legs. I can additionally "spy" on the call. Is there any way I can play a sound file into the call and not loose the ability for the people to continue talking while listening to the sound file? -- Jim Dickenson mailto:dickenson at

In one of my center , its not taking root password. Anyways to recover it ? In other terms , I lost the control of server. Any solution or re-installation is the only way left ? I am using CentOS. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090122/ef95ad6e/attachment.htm

Hello. There are so many sound files in /var/lib/asterisk/en. Is there an easy way to let me play back all of them one by one while I am watching CLI to see the current file name? Thanks for help. -- Jian Gao IT Technician SJ Geophysics Ltd. <http://www.sjgeophysics.com> jian.gao at sjgeophysics.com <mailto:jian.gao at sjgeophysics.com> Tel: (604)582-1100

Hi All, pl help me with this basic question. I have a users (soft clients) with usernames having Alphabetics. I want to use Asterisk as my server. How should I have the dial plans as there are no numbers involved . so How can I make the configuration to work ( with numbers I can get this done using extensions.conf) my expected result is : alice at pbx.com should be able to call bob at

Jim, and all: Thanks for the response. If I can repeat what you are saying: you don't have to define the multiple lines in sip.conf? For comparison, with my current esi setup, we have 10 outgoing lines. If one line is busy, then the service just rolls to the next number. This is set up with the phone service. That doesn't have to done with outgoing sip lines? Does the dialstatus

Dear all, Anyone of you knows how to play an audio file at the beginning of a call for both Caller and Callee? A(x) of Dial application only plays audio for callee. I don't want to use MeetMe because I want to use Monitor and MixMonitor. Thank you! ________________________________ Este mensaje se dirige exclusivamente a su destinatario. Puede consultar nuestra pol?tica de env?o y recepci?n

if i run let's say 1 pbx running on my main linux box and a another on my windows box if a person dial my main number and press lets say 1 are it possible to transfer the call over to my other pbx hope anyone understand -------------- next part -------------- An HTML attachment was scrubbed... URL:

asterisk 1.4.32 have zapata.conf soft link to chan_dahdi.conf to use flash operator panel < 2.0 (from extensions.conf) exten=> 304,1,ChanSpy(Zap/4|q) exten=> 304,2,hangup There is no entry ChanSpy(Zap/41) in extensions.conf On dialing 304 and Zap/41 is in use this happens: [Jul 1 18:24:47] VERBOSE[14447] logger.c: -- Executing [304 at flash:1] ChanSpy("Zap/31-1",

We have a client that currently has a Microsoft Lync setup. I must admit I know nothing about this setup. What we would like to be able to do is allow the phones on desks connected to this server the ability to dial something that would allow the phone to access an asterisk box to be able to do an agent login over their LAN. Is there any way to do this? Can the Lync server have a SIP trunk to

Does Asterisk, currently using version 1.4, get any more information about the result of an outbound call made over a PRI line compared to a call via a SIP trunk? As an example, in a PRI call there is this message that shows up on the console: [2011-01-05 14:59:02] -- Channel 23 detected a CED tone from the network. for a call to a fax machine. Does asterisk set anything that a dialplan can

Hi all, Now i want to check a number (channel) online, offline or unreachable on asterisk but i don`t know to do. Can anyone help me to solve this issue. Thanks and best regard! -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110109/c193b48d/attachment.html>

Hi all, I am trying to isntall the v1.6 version of Asterisk on my Intrepid system, but I get an error after I have typed make: [CC] manager.c -> manager.o manager.c: In function ?action_getvar?: manager.c:1732: error: ?SENTINEL? undeclared (first use in this function) manager.c:1732: error: (Each undeclared identifier is reported only once manager.c:1732: error: for each function it appears

Hello, On an Asterisk 1.4.33.1 in a simple scenario: [test] exten => _X.,1,Dial(SIP/12345 at peer01,,,) exten => i,1,Hangup(${HANGUPCAUSE}) exten => t,1,Hangup(${HANGUPCAUSE}) exten => h,1,Hangup(${HANGUPCAUSE}) I have noticed that no matter what value we set in the Hangup(<cause code>) commands, if the call is not answered by peer01 for any reason, the actual cause code

Hello. I'm new with asterisk. Can you help me in this: I have cisco sip phone (601) connected to asterisk server, and 1 client number (500). I want to dial from 601 to 500. But get error in cli console: [Apr 27 15:30:15] NOTICE[9650]: chan_sip.c:20059 handle_request_invite: Call from '601' to extension '500' rejected because extension not found. What's wrong?

I am trying to get updateconfig working. I found an example of updating configuration files here: http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+Action+Upd ateConfig When I tried it the conf file was updated but the new entry was not added. action:updateconfig reload:no srcfilename:manager.conf dstfilename:manager.conf Action-000000:append Cat-000000:newuser

Bad day today. Why this new JIRA system not working. I have created issue and submit and i got blank page.. Please someone help me to create BUG!!!!!!!!!!! -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110608/e99afa31/attachment.htm>

I have an i686 cpu and when compiling from source I get this error: touch /usr/src/dahdi-linux-2.2.0/drivers/dahdi/xpp/init_fxo_modes.verified Building modules, stage 2. MODPOST WARNING: could not find /usr/src/dahdi-linux-2.2.0/drivers/dahdi/vpmadt032_loader/.vpmadt032_x86_32. o.cmd for /usr/src/dahdi-linux-2.2.0/drivers/dahdi/vpmadt032_loader/vpmadt032_x86_32.o Anyone else seeing this?