similar to: no voicemail on pstn line

Group In voicemail.conf I would like to having the following setup per context not per-mailbox settings serveremail userscontext fromstring usedirectory emailbody pagerfromstring dialout sendvoicemail callback review operator volgain nextaftercmd forcename forcegreetings tempgreetwarn Can this be done? Thanks! -------------- next part -------------- An HTML

Hello. Happy New Year to everyone. I have a small WISP and would like to have customers to call our number to check their balance. I am planning on writing an AGI with php so it can get the customer info from the customer database. I don't know how to interact with the caller while in the agi script so this is what I have in mind: [test-agi] exten => 33,1,Answer() exten =>

>From voicemail.conf: ; Maximum number of messages per folder. If not specified, a default value ; (100) is used. Maximum value for this option is 9999. ;maxmsg=100 ; Maximum length of a voicemail message in seconds ;maxmessage=180 ; Maximum length of greetings in seconds ;maxgreet=60 I would like to configure these parameters on a per mailbox basis using Realtime voicemail. I

Hello, I am struggling with what seems a common unresolved problem, changing the password from voicemailman when using a realtime engine (adaptive_odbc in my case, connected to mysql). I have seen messages dating back to 2007 with this problem and the last one was bug 5168, reported as closed, but without explaining the fix

Hi List, I am using asterisk 1.2.14 with real time and I am trying to send the email to more than one email address. In that field I put in user1@domain.com;user2@domain.com. When the call goes to VM I see in the CLI: uniqueid => 17 customer_id => 0 context => techmast mailbox => 14 password => 1234 fullname => Sales and Service email => user1@domain.com email =>

Hi i have installed a new Asterisk server on Fedora. My first server use Asterisk 1.6.x with a MySQL CDR and realtime. I have a small problems, when i configure on the new server, the same information in MySQL, we have a error: [Jun 3 16:27:59] ERROR[3140] res_config_mysql.c: MySQL RealTime: Failed to connect database server SSI on myhost.myserver.com (err 2003). Check debug for more info.

Hello List. I just installed a2billing with asterisk 1.6 and got it working. The only problem is that I'm trying to setup something to manage who's using the most minutes in the house. I noticed a2billing only works for callin cards setups, or maybe I didn't configure it correctly for what I want. Can I use a2billing for "?VoIP residential services"? if yes, how? if no,

Hello. I have a question regarind sound files in asterisk 1.6. I have a sound package in ulaw format and I would like to know if I have a sip extension with allow=alaw would asterisk convert that file to the codec the user is allowed to? I am having a problem playing a file that exist in /var/lib/asterisk/sounds/es/good.ulaw but asterisk is telling me it doesn't. Here's what I get when

Hi everyone, I've set up a little Asterisk system with a Digium TDM400P and everything works splendidly except for the messages callers leave. Every message that a caller leaves is very faint. I've already set volgain=6.0 in voicemail.conf, and that seems better, but to be at a good volume I estimate I may need to go up to 40.0. Is that reasonable? One interesting artifact is that at

Hello. I'm looking to buy a FXO card to do some testing with two phone lines I have at home and was looking in ebay some and found some cheap ones but, the I've never heard of the brand or manufacturer: chinaroby. They run for about $99 plus shipping. Have any one used these? or please recommend one... Money IS an issue. Thanks.

Hello. Sorry to repost this message but, I don't have the original message in my inbox nor in my sent box. Well, last week I posted a problem I am having trying to use an asterisk server use a voip provider and a pstn. Pstn works fine but, I cant even connect to my provider's server. I don't know what I'm doing wrong. I tried using a soft phone and I'm able to register and

Hello List. I would like to know how I can use two or more service providers with asterisk to be used randomly for ei, if an user tries to make a call I would like to randomly use a provider. It doesn't matter where the call is destined to. Thanks.

Hello all, This is my first attempt at posting a patch. So if I screw this all up, my apologies and please someone let me know without beating me up too bad. To use this patch. You need to have an extra line in /etc/asterisk/voicemail.conf that looks like volgain=10.0 The 10.0 gets passed to sox which you will need installed on your system. 10.0 is what works for me, anything over 1.0 will

Hi List. Don't know if I already posted about this problem but, if I have I apologize for the double post. I am trying to test a time of day extension dialing 80, all I'm trying to test is if is morning I would like asterisk to say "Good Morning" but, when I run the test I get the following error message saying that the file doesn't exist and it does: Night..............

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I' try to implement an automatic callback mechanism, just for local SIP calls.. Callback on busy and on no answer. If the other party doen't answer, it should be possible to press 5 to place an callback. Here is my dial: exten => _X.,1,Set(EXITCONTEXT=callback) exten => _X.,n,Dial(${DIALNUM},${ARG2},dtT) And here the script for

Hello List. I have a question regarding connecting two asterisk servers. I'm trying to learn how asterisk comunicates from server to server. I already have a server running smoothly now, I'm installing another one to test it along side the actual one. I would like to run different scenarios: 1. Have one of the boxes at a different location outside the LAN and have them communicate. 2.

Hello. I am currently testing an asterisk server using the default codecs, I have allow=all, and noticed everytime I test it in a wireless lan the latency rockets off the roof to over 1000ms. I would like to test g729 since it uses less bandwidth but, read somewhere I have to buy a license per every channel I have. Does this means if I have my server connected with 10 sip clients I need to buy a

All, I have a problem with H323 the call disconnects when answered. The debug shows -- Executing Dial("SIP/sj1-4ff7", "H323/0797617729") in new stack -- Called 0797617729 -- H323/0797617729 is ringing -- H323/0797617729 answered SIP/sj1-4ff7 == Spawn extension (default, 0797617729, 1) exited non-zero on 'SIP/sj1-4ff7' -- Executing

If I''m using eth1 as my lan zone on my router box, it needs a static ip... what do I set the gateway option to in /etc/network/interfaces since this computer is actually the gateway for the rest of the lan? Itself? My "net" NIC''s address? Something else? My lan isn''t getting internet access using the default Shorewall config file (edited per

Hello. I have a server installed with asterisk 1.6. I have a PSTN line that comes in to one of those clone cards. Everything seem to be working fine. The only problem I have is that I can't get voicemails coming from the PSTN line. All other: SIP, IAX work fine. I can hear those ok but, when it comes to a call that comes in from PSTN I get no sound. What can cause that problem? Thanks in