similar to: Question

Hi all, In order to do loadbalancing between my two *, i wanted to stock all things concerning voicemail on a NFS partition... I see that the voicemail system put his files onto two differents directories : /var/spool/asterisk/voicemail/mycontext etc. and /var/lib/asterisk/voicemail/mycontext etc. I've two questions : Why ? and how can i do to centralize the destination of the messages AND

Hello, I checked out the asterisk version from the CVS. But I dont seem to have the addmailbox script. How can I setup a mailbox without this utility. Regards, Rajesh Golani

hello according to this user guide found at http://www.automated.it/guidetoasterisk.htm#_Toc49248768 it says the following Voicemail - Please leave a message after the tone... Ok, so you've got the basics going, and it's great - if you happen to sit by you phone all the time. What happens if you are out/away from your desk/sleeping you'll miss those vital calls. We need to set up

Hi, I am a new user to both Linux and Asterisk and would be grateful for any help and advice anyone has to offer. I have installed Linux and asterisk as per Andy Powell's excellent getting started guide. The problem I have is that the addmailbox utility does not work and I cannot find the file anywhere on the machine. I downloaded the files via CVS so assume I have the current

Hi all, I've note that the variable DIALEDPEERNUMBER is broken. Now i want to know if exist another method to retreive the called number on *, and, if it's possible, an example ;) Regards.

Hi; Do you know where the addmailbox script goes? Now in CVS, it shows revision 1.3 date: 2005/08/02 11:57:06; author: markster; state: dead; lines: +0 -0 This script is now useless... ---------------------------- So any new tool to create voicemail folder? Thanks Kun

Hi Having issues configuring voicemail box Ran command /usr/src/asterisk/addmailbox . .gets error 'No such file or directory' Asterisk was installed without any error message, Running on R.H Fedora - all packages installed Please assist -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi Guys, My name is Marcias, and I am setting up for the first time an Asterisk PBX, I am learning as I go along. I have been able to download and install Asterisk, Libpri and I have been able to get Asterisk up and running. I have several questions: 1 .I can call the Asterisk server from my Xten phone and it picks up. I have 3 computers (one of them being the asterisk server) I can seem to call

I have googled this one to death, and can't find anything. I added a number of new users to my asterisk (current CVS) system. I am using the "Voicemail2" family. I added entries in extensions.conf and voicemail.conf for my new users, and I have tested leaving and retrieving new voicemails for them. All of this works fine. But if one of the new users tries to "Administer

Hi Everyone: I have a THREE questions. What is a sip proxy and what is the benefit of having one with Asterisk? I am well aware that we have a sip channel in Asterisk and that we have SIP registration. I am not sure why you would need a SIP server and OR a registration server. Second question, with Asterisk are you able to do video on VOIP video phones? Last question, does

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I've forgotten the command to add a vm box, and searching google and wiki I'm surpriced I cannot find it. I'd love to know where this is written, so I can see how I managed to miss it! - -- Steve "They that would give up essential liberty for temporary safety deserve neither liberty nor safety."

Hello all. Now I aleady installed the Asterisk. I could make communication between 2 XLite client through Asterisk. I tryed to test the voicemail function as follow. 1, I make a call to 1001 from 1002 2, Start ringing 3, Wait untill time out for ringing If no problem, 1001 go to voicemail and unavailable message will be played. But 1001 receive a 403 forbidden massage and connection go

The scons file that I posted yesterday is broken... I've not taken in account the commit of 2 days ago that modified *iterator.in* and friends. I've updated it and now it works again; inside it you can find comments about what files currently are NOT compiled and why (hint appreciated!) Attached to this email you can find it and a new series of patches against current CVS, that I hope

On Sun, 2004-09-26 at 12:42, Paolo Invernizzi wrote: > Attached to this email you can find it and a new series of patches > against current CVS, that I hope are pretty safe. Paolo, I applied most of your patches to cvs. I didn't apply the patches for things under lib/Support. I will leave that to Reid (if you can resend those to this list it would be great). Here is the list of the

Here I am again... I missed that in the previous diff, they are the usual missing <algorithm> and the std:: namespace missing in sort, find, make_pair. Alkis can you please give them an eye when you have time? The next major problem is that VC has only int rand(void) void srand( int seed ) So I donno how to compile the ExecutionEngine/Interpreter/ExternalFunctions.cpp that refers to the

Hi all, I am new to this list... Can I use Asterisk as a SIP Proxy and as a SIP to H.323 translator? I want to implement PC-to-Phone calls in the following topology (for signalling): SIP Softphone --> Asterisk --> Gatekeeper H.323 ---> Gateway H.323 ---> PSTN The RTP audio packets would go direct through Softphone to gateway. Does someone have a configuration file example of

Anyone wanting to go ahead and use VoiceMail2 will probably need this new version of the addmailbox utility.. I have updated the addmailbox utility to create the correct directory structure required by VM2 and copy the required files to the correct locations.. I just called it addmailbox2.. I don't know if it will be added to the CVS that is up to Mark.. Remember to make it executable.. :)

Hello all, Does anyone know how to successfully transfer a call using the GrandStream Budgetone 100 phones? I've read multiple posts talking about hitting flash, the dialing, then flash again, etc. Some posts talk about using the transfer button, then dial, then flash. Anyways, it seems that I am able to put the caller on hold (whereas they hear hold music) by pressing flash or

Hi all I have been using * for a couple of month and now I wanted to install the Asterisk Calling Card. But when I try to install I get the following error: [root@terraserver1 astcc]# make install mkdir -p /var/www/html/_astcc mkdir -p /var/www/cgi-bin/astcc-admin chmod 755 ./astcc.agi chmod 755 ./astcc-admin.cgi echo | ./astcc.agi >/dev/null Detected dry run! ./astcc-admin.cgi

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