search for: landysaccount

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. 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 =>

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

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. 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.

...t; hi, > > I am using this card and IP phone about 6 months. There is no issues > > at all. > > > > Installation procedures are same as Digium analog card. > > > > Hope it helps, > > Ashik > > > > On Tue, Aug 3, 2010 at 6:28 AM, Landy Landy <landysaccount at yahoo.com > > <mailto:landysaccount at yahoo.com>> wrote: > > > > 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 o...

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..............

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.

I am testing an ivr but I'm having problems. The call keeps looping and it doesn't hangup the call after passing three times through the menu. Here's my conf: exten => s,n,NoOp("Here's Count") exten => s,n,NoOp(${COUNT}) ;123,n,Set(COUNT=$[${COUNT} - 1]) exten => s,n,GotoIf($[${COUNT} = 4]?33,1:44,1 ) exten => 1,1,goto(tech-support,s,1) exten =>

Hello List. I am having problems retreiving voicemails on my system. I noticed when someone leaves a message through the pstn line I can't hear anything. I tested leaving a message from one of the extensions and that can be heard. I don't know if is the type of card I'm using for analog ( cheap X100p modem ) calls but, can't hear any message coming in from that line. Any

Hello. I'm trying to set TTS with Cepstral and Swift but can't get it to work. I get this error when testing it: -- <SIP/101-00000000> Playing 'welcome.gsm' (language 'es') -- Executing [702 at local-calls:3] Swift("SIP/101-00000000", "Hello this is ceptral") in new stack [Jul 28 18:29:16] NOTICE[5191]: app_swift.c:304 engine: Text to

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

Hello. I'm trying to test an Asterisk server by using a VOIP provider for international calls but, I'm having problems trying to get my server communicate with theirs. I don't know if I'm having all these issues becuase I'm behind NAT or what. I have the following in my server's sip.conf: [provider] type=peer host=<theprovider's server> username=<username>

Hello List. I am puzzled and how asterisk listens to calls or connections from clients. When I do a netstat -nat I don't see asterisk listening on port 5060. Now, I'm testing a server with three network interfaces: two to the internet doing load balancing and the other to our LAN. I would like asterisk to only accept connections coming from our LAN but, can't find where to configure

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