similar to: Caller's phone keeps ringing after 200 OK

Hello Everyone, We are looking for a simple open source auto dialer with "polling" capabilities. What we would like is a program that we can upload leads to, and have asterisk: i) Dial numbers ii) Play pre-recorded iii) If user presses one, forward the call to an agent There are so many solutions out there it's hard to make a decision on what works, what has just a limited free

Hello Everyone, We are seeing many instances where we receive 200OK from the vendors however, asterisk still keeps ringing. Is there anyway to stop this from happening? I remember reading something about early media however this seems to be a case of late media? Kind Regards, Nick from Toronto. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello Everyone, We are currently experiencing some higher load on our servers, and since signaling comes into our servers on G729, we would like to implement G729 pass-through. A few questions arise, do we need to convert all the recording to the codec, and what about voicemail? We are also using A2Billing (hope I am not violating any thread rules), and would like to convert all that recording

Hello Folks, I am setting up a call center but we have few agents so one agent is able to handle calls of different languages and different queues. For the agent to identify the caller, I want a popup to appear as the phone starts to ring with the caller's number, language (selected in the IVR), Queue (sales, support etc) and any other information (e.g a URL with parameters) I can send this

Hello, As of today's cvs * snapshot I am able to pickup a ringing (sip) cisco 7960 with *8 but the extension then keeps ringing indefinitely, even though I picked up the call. Is this a known issue? Thanks, -- There are no Debian developers in any part of Hell, because the good karma incurred by being one takes you straight to the pearly gates. Of course, the frequent flame wars you put

Hello There, I would like to play a busy tone (ie BEEP BEEP BEEP) when the underline carrier sends back 486 Busy Here. Looking at Dial parameters ( http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial), it mentioned something about the r parameter as not being very professional or something like that... Then there was: U(x): Executes, via gosub, routine x on the called channel. This is similar

Hi, I searched throug the forum and could not find an answer. Sorry if already posted. I'm using Asterisk 1.4.19 a i686 running Linux Centos. Sometimes when I call and the person picks up, it continues ringing and I can hear the person say hello then it hangs. Has anyone experienced this problem and what is the solution to it. -------------- next part -------------- An HTML attachment was

Hello Everyone, I am required to write a java program that will get our asterisk to: * Query the database for phone numbers * Loop through numbers and dial * Play message * Get dial pressed response - If 1 = Yes - If 2 = No - If 3 = Connect to Agent * AMD Capable * Disposition I am proficient with Java and found the Asterisk-Java API. My questions are: * What is the

Hi, I am new to this group.I searched for my problem in the forum but could not find any solution. So here it goes: In my work place we have an asterisk box. Everything works fine except the fact that when I first call the work phone number from my cell the auto-attendend works fine but If I hang-up and call the same number again, the call keeps ringing instead of auto-attendent answering. I also

Hello, all. I have an asterisk 2.3.2 and a Sangoma interface through which I connected an external PSTN line. I use it as carrier for VoIP calls. I can make successfully calls, but there's one problem, I receive 200 OK with SDP with delay (sometimes more than 30 seconds). So when I make a call through asterisk I receive intially: - 100 Trying - 183 Session Progress, with SDP when the called

Hi asterisk users, I have a inssue with incoming calls with wcfxo card, while receiving a call, I?ve configured my dialplan to forward the call to all mi home voip extensions and that works just fine, but while in the call, after a few seconds, the pbx starts the simple switch once more and keeps ringing the voip extensions log as follows:

Hi, I have one phone on my network that just keeps ringing (when I call it) and does not go to voicemail. If the person there is on the phone, and someone calls it they get the busy message, but they never seem to get the 'unavailable' message... instead it will just ring and ring and ring... any ideas? They are setup with a voicemailbox, and it is set to transfer after 15 seconds of

Hi, Did a test with Local, exten => 1234,1,Dial(Local/1 at agents) [agents] exten => 1,hint,SIP/0317998975&SIP/0317998985 exten => 1,1,Dial(SIP/0317998975&SIP/0317998985) When calling 1234, both 0317998975 and 0317998985 rings when answering in 0317998985, 0317998975 stops ringing, all fine but when answering in 0317998975, 0317998985 keeps on ringing... 0317998975 is an Avaya

Hi, when I press reject on Firefly, another end hears congestion, but I'm still listening to ringing... Also how you determine which version of Firefly you have? I also don't understand licence completely and would like to get this answer - can I distribute Firefly to commercial users if I don't charge anything ? Regards, Rob.

Long story... Would be nice if we can remove this on BYEs X-Asterisk-HangupCause: Normal Clearing. X-Asterisk-HangupCauseCode: 16. Kind Regards, Nick. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140723/09e97fd1/attachment.html>

Hello Everyone, The timezone is set correctly on the OS America/Toronto: mv /etc/localtime /etc/localtime.bak cp /usr/share/zoneinfo/America/Toronto /etc/localtime I even tried adding the timezone setting to sip.conf: timezone=America/Toronto However. Asterisk wants to be in Bucharest? Thinking about it, I want to be in Bucharest! Cheers, Nick.

Hello Everyone. Our environment is a register free setup, and our phones are set as host=dynamic. The problem we are experiencing is for inbound calls: Name/username Host Dyn Forcerport ACL Port Status Realtime 222/222 (Unspecified) D N A 0 Unmonitored Cached RT So when we DIAL 222 we get: WARNING[23103]: app_dial.c:2198 dial_exec_full:

Hi All I've got an asterisk system, using a couple of Xorcom Astribanks to provide FXS ports. (I'm using the zaptel 1.2 branch, if that matters) I've noticed that the ringing volume is a lot louder than on our old phone system, and people are starting to complain it's too loud. (This is the noise the phone makes when it rings, not the noise in your handset when you ring

I have: RedHat 9.0 TDM40B asterisk-0.9.0 compiled from sources zaptel-0.9.1 likewise /etc/zaptel.conf contains fxoks=1-4 loadzone = us defaultzone=us loaded modules zaptel and wcfxs /etc/askterisk/zapata.conf contains [channels] language = en signalling = fxo_ks context = phones channel => 1-4 /etc/askterisk/extensions.conf contains [general]

I try to use a SIP trunk from a VOIP provider to make land to mobile calls. If I do these from a ZAP channel, using an analogue phone, after few seconds of silence (I don't like to generate fake [r]inging) I ear the ringing tone from the mobile operator along with any message the mobile operator decide to say me. If I try to use a SIP phone (or a IAX phone) attached to my asterisk box, I