similar to: How to detect answering machine

Hi! A customer of mine wants to connect an asterisk system with 240 to 480 lines to a PSTN switch. To save the costs for E1 cards and the corresponding E1 mainlines he wants to connect the system to the switch by a SIP trunk. Phones will be connected to the server through the same SIP trunk as this will be some kind of a "hosted pbx". Given he finds a provider wich has this much SIP

Hi! I am trying to suppres the transmission of my CallerID when I place a call using a .call file in /var/spool/asterisk/outgoing Callerid: Callerid: "" and Callerid: '' made the call transmit the default number (headnumber+0) Callerid: 1234 made the call transmit 1234 Using *31* in front of the called number in Channel: made the call not go through. Does anyone have an

Hi! I have an automated dialout via a call file to a mobile. Can I detect when the call is not answered but forwarded to the mobile operator voicebox? I would like to stop the dialout if this is the case. TIA, Mike

Hello I have the following problem : The dataframe TEST has multiple lines for a same person because : there are differents values of Nom or differents values of Prenom but the values of Matricule or Sexe or Date.de.naissance are the same. TEST <- structure(list(Matricule = c(66L, 67L, 67L, 68L, 89L, 90L, 90L, 91L, 108L, 108L, 108L), Nom = structure(c(1L, 2L, 2L, 4L, 8L, 5L, 6L, 9L, 3L, 3L,

Hello Asteriskers, :-) We're trying to set up an outbound notification calling for system alerts with Asterisk 1.4.0. We generate a call file in /var/spool/asterisk/outgoing and the outbound call is originated through Zap/1 (Sangoma A200D to a Canadian POTS line). The problem is that Asterisk does not wait for the other side to answer before it starts playing the message. So the

Hi, I was able to implement T122p one port PRI and was able to call out, but I am planning to use TE412p (includes echo cancellation) 4 port digital card (PRI), I wanted to know can asterisk support 3 four port PRI cards (12 PRI connections) with proper hardware like dual core quadcore processor and 8gb RAM in one server? Also I was planning to implement using 64 bit architecture with Asterisk:

Hi, Here is what I want to do: Labour = Class.create(); Labour.prototype = { initialize:function(name){ this.name = name; } } What I want to do is create a class called "Worker" which will inherit from "Labour", and the signature of "initialize" is "function(name, position)". May I ask what should do? Thank you all very much for the

Hi,How to accepts the call and plays a voice message on the line without billing the caller ? This may be necessary for IVR applications that want to explain features of the service offered . Regards _________________________________________________________________ MSN 8 with e-mail virus protection service: 2 months FREE* http://join.msn.com/?page=features/virus

Dear Friends, We have implemented "Asterisk" in our organization. There are 150 members in our organization. At present all are using softphones. Now, I want to buy hardphones for our staff. Can anybody suggest me that what is the best hardphone for Asterisk with low-cost? Thank you. Regards, Chandra. --------------------------------- Ring'em or ping'em. Make PC-to-phone

Hi! Just ashort question - obviously I am too stupid too find the answer on the net. :-) I want to upgrade a running Asterisk 1.4.2 to 1.4.6 - what will I have to do? Just install it over the existing version? Do I need to backup the configuration? Will I need to reconfigure the source or will the new version "import" my old settings? Will I need to update Zaptel and Libpri too?

If Asterisk is configured as a simple answering machine replacement with the X100P connected to PSTN line. No FXS ports in the Asterisk machine. Standard phones are connect in parallel with the X100P like you would a regular answering machine. Can Asterisk detect that a phone has been picked up and cancel the outgoing message and/or voice recording? What about if the phones are connected to

Hi all, I need your lights :) There are many hardware provider for E1 cards on the market, what's your exeperience with E1 and what's the preferred provider for Asterisk out of Digium? Olivier

Hi everybody, Happy New Year ! I'm trying to detect if a call was answered by a machine (linke voicemail systems) or a human. I would like to use AMD (Answering Machine Detect) command, but with my configuration it was not possible get there. Follow my dialplan: exten => _[789].,1,NoCDR exten => _[789].,n,Dial(SIP/${EXTEN}@111,60) exten => _[789].,n,AMD

I am using asterisk 1.4.21 with outgoing call files. I am call a line that is busy as you can see below. How can my AGI ask what the status of the last call was so I can tell if there was NO ANSWER or it was BUSY? Thanks, Jerry -- Attempting call on SIP/401 for smvoice_callprogress at smvoice-dialout:1 (Retry 1) -- Got SIP response 486 "Busy" back from 192.168.1.161

-----Original Message----- <snip> Is this possible with asterisk? Anyone have a sample dialplan? -other than the problem outlined below I would try something like S,1,wait(20) S,2,voicemail(uwhatever) S,3,hangup That should ignore the call for 20 seconds and then leave a message in the unavailable greeting for 'whatever' then hangup That leaves another problem -

HelloI have the following system Asterisk 1.6.1dahdi 2.2.0.2 TE420P card Centos I have noticed that all the four lights are blinking(ie coming red and then off so on)... Previously I also noted that when dahdi drivers are not installed lights blink but one by one in sequence(like in marriage cermonies :P) and after dahdi installation lights get off ... but this time all at same time

I have a Centos 5.4 machine that has, for the past two weeks, apparently been shut off over the weekend. It's just sitting there turned off on Monday morning and when someone hits the power switch it comes right back on and everything works again. This happened last weekend, and again over this past weekend. Here is /var/log/messages from shortly before it apparently shut down this weekend.

Hi all, I have hit a wall configuring a TDM400, I have set these up before without issue but today I just can't seem to figure out what I am doing wrong. On an incoming call the following is produced in the Asterisk console with verbose 4 -- Starting simple switch on 'Zap/2-1' Mar 22 16:12:34 NOTICE[2051]: chan_zap.c:6063 ss_thread: Got event 18 (Ring Begin)... Mar 22

I have looked and understand how to create an outgoing call by putting a file in the spool. After searching the archives, I came up with stuff about predictive dialers, but nothing right on topic. I would like to be able to detect an answering machine in outgoing calls so that the message will restart with the beep. In other words, I want to send a message to all of my customers, and if they