similar to: ivr menu not hanging up call

Displaying 20 results from an estimated 7000 matches similar to: "ivr menu not hanging up call"

2010 Jan 02
4
Help getting info from caller
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 =>
2010 Jun 15
2
a2billing for residential voip usage
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,
2009 Nov 26
1
Unable to open sound file error
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
2007 May 23
0
IVR Loop on invalid input
We are running 1.2.14 with an IVR in the dialplan. If I connect to the IVR with a SIP phone (Polycom or Xlite) and press a couple of digits very rapidly (I found this with 33 on a sticky keypad) which are an invalid response, Allison will go into a loop saying 'I'm sorry, that is an invalid response, please try again.' over and over. This does not happen with a commercial
2010 Aug 03
1
chinaroby fxo card - never heard of them
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.
2009 Nov 16
1
can't call through voip provider
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
2009 Dec 12
1
how to randomly use provider?
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.
2009 Dec 13
1
Unable to open file...
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..............
2007 Jul 30
0
Trouble getting sound from a call
Having some issues with getting sound from a call. I have 4 systems. 3 main systems which handle calls for our 3 locations. The 4th system is the central voice mail system. When an inbound call gets passed to someones voice mail its done with an IAX2 connection. The same happens after hours when we have our night mode set. If you dial the main number after hours you are passed straight to the
2009 Dec 15
3
Best way ro run 2 or more asterisk servers?
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.
2005 Sep 09
0
OT Humo[u]r IVR Menu sample
Some one on another list I subscribe to had a session with an annoying IVR system at their doctor and posted this link. http://www.pendulum.org/humor/humor_psych_hotline.html -- Dave Cotton <dcotton@linuxautrement.com>
2005 Sep 04
1
Option 1 in IVR menu
Hi all, I'm trying to setup a simple IVR menu in a context in extensions.conf. So far, I have: extension s for playing back the menu # to repeat it * for directory 0 for operator 1 which goes to another context: exten => 1,1,GoTo(option_1,s,1) Here is what I have in extensions.conf: [incoming] ; main greeintg exten => s, 1, Ringing exten => s, 2, Wait(10) exten => s, 3, NoOp()
2010 Apr 08
2
IVR menu sound processing for AMR and GSM + live test available
Hi! We are in process of setting up an audio guide that will cover notable places of our capital Riga, Latvia. The target audience are tourists that dials a free phone number from a mobile handset to listen to a 3 minute introduction to historic place. All audio, 10+ languages are recorder in studio at 44KHz. The audio is stored on server in A-law 8KHz because we'll be pushing it through E1
2009 Dec 01
6
Question about g729
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
2004 Aug 12
10
H323 problems
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
2005 Jun 10
11
/etc/network/interfaces
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
2009 Oct 08
4
No sound on voicemail from analog line
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
2010 Apr 13
0
Problem with Callfiles
Hi! I am trying to do a callfiel for autodialing but when I move the callfile to outdialing folder asterisk seems like if did the call but it doesnt. I put here my callfile and that I get when asterisk begins to do the call If anybody has idea, pls. Tell me TIA ;;----CallFile----- Channel: Zap/g1/8093908270 Callerid: 8093908270 MaxRetries: 2 RetryTime: 300 WaitTime: 45
2009 Sep 17
1
Changing or Adding a Line to the Extensions.conf in Asterisk
I have a Asterisk PBX System with Redhat Linux Fedora 4, Webmin version 1.400 and I am simply trying to configure into the "Extensions.conf" script an entry that will add to the "Auto-Attendant" a line that will allow a "Caller" to enter a "0" (Zero) will then ring the extension(s) of the "Operator" to speak directly with the "OPERATOR"
2010 Feb 16
6
Asterisk listens on all NICs
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