similar to: How to do barging using asterisk server.

Hi, we are using Asterisk to record and playback. Both services are working well independently but it seems we can't start playback of a file while we are still recording it, even if the file is already in the hard disk. Is it possible to playback while recording the same audio file? Or is there a way to enable it? Regards, Jahnavi. -------------- next part -------------- An HTML attachment

Hi, I have our software with SIP running on it.I configured asterisk server as proxy. How do I implement the call screening features(incoming and outgoing) using asterisk server.Please suggest me how to proceed on this. Thanks & Regards, Jahnavi. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I have xlite registered with a user. Now i dial an extension say 1500 which has the dial plan as follows. exten==>1500,1,AGI("localhost//hello.agi" So when i dial extenstion 1500 the script hello.agi is invoked which in turn plays a welcome message. I now want to transfer the call now to operator. How can i achieve this???Please help me in this regard as this is very urgent.

Hi, I have an xlite registered with asterisk server. When i dial a number AGI is invoked. and in this we are running to threads one to record files and one to play files. So i dialed the extension and i started recording and playing at the same time. On the xlite i m getting an indication when recording my voice and at the same time i could see playing the other file too. But in the directory

HI folks, I have set a barging facility with our production box.Client able to barge a agent but client raise a requirement, they want talk to barge agent but that communication is not listen by customer. It is possible with asterisk or not. thanks in advance. Regards Akhilesh -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I have xlite client registered with a user. Now i dial an extension say 1500 which has the dial plan as follows. exten==>1500,1,AGI("localhost//hello.agi") So when i dial extenstion 1500 the script hello.agi is invoked which in turn plays a welcome message. I now want to transfer the call now to operator. How can i achieve this???Please help me in this regard Thanks &

Hi, I installed asterisk server in my linux box. I configured a user 1000 using xlite and registered with asterisk server in the same linux box. I configured one more user 1001 in other box and this user also got registered with asterisk. But i am facing two issues here. 1. When a call is made from 1001 to 1000 i could see an incoming call blinking but no audio flow is observed. 2. When i made a

Has anyone successfully implemented Asterisk as a voicemail server for a GSM/cellular system and worked out a way to send notifications of new messages to the phones? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100823/0ed9345d/attachment.htm

Hello Users I'm planning to do Call Barging and Call snooping , I saw this Feature in asterisk.org. This Barging and Snooping are test for " is Agents are replying the Answer or not " that I'm guessing Can anybody help me... this Feature ...... How to do Call Barging and snooping in SIP Channels , I;m not using any Zaptel Card -- Thanks and Regards Ravi Prakash Sunkara

Hellow, i need the following requirements with asterisk : 1) Can ACD (Automatic Call Distribution) service work with asterisk, and how to set up ACD in asterisk ? 2) How call barging can set up in asterisk ? 3) How call recording can set up in asterisk? Thanks mahboob System Engr SSL -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello list, how come on my Asterisk 1.6.2.11, I have no help available ?! asterisk*CLI> core show application Dial -= Info about application 'Dial' =- [Synopsis] Not available [Description] Not available [Syntax] Not available [Arguments] Not available [See Also] Not available Kind regards, Jonas. -------------- next part -------------- An HTML attachment was scrubbed...

Just done a clean install of rc5 on a totally new machine and found the following:- /etc/init.d/asterisk start errors on line 109 - there is no 0 before $VERBOSITY as in the other lines. More interesting is that after make samples I have no iax2 available. Dave Cotton

<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN"> <html> <head> <meta http-equiv="content-type" content="text/html; charset=ISO-8859-1"> </head> <body text="#000000" bgcolor="#ffffff"> <font size="+1">Does anyone have links to the most recent audiocodes

Hello list, I need to know how to deal with a redundant network with only one asterisk server, which is receiving registrations from the end points on both of its ethernet ports. This means extension 201 is registering both from eth0 and from eth1. Is there a way/software which can act as a middle man between asterisk and the ethernet ports, and by default sends registrations to asterisk only

I downloaded the latest 1.8 (27922) but got some segmentation faults. The first one was when it loaded cdr_odb, and so I changed menuselect not to compile that one, but the second one was when it tried to load chan_agent and so I stopped there to see if anyone else was seeing this. The agents.conf is all commented out except for [general] . Anyone know what is happening? Thanks. P.S. I deleted

How can I let asterisk immediately dials a trunk when off hook?

I am having a one way audio issue with xlite clients behind NAT. They can connect to the server and make calls but no audio is heard on the other end. my sip conf [general] context=default bindport=5060 bindaddr=0.0.0.0 srvlookup=yes canreinvite=no[tomfmason] type=friend secret=secret callerid="Thomas Johnson" <XXXX> host=dynamic nat=yes canreinvite=no disallow=all allow=gsm

Hi, Is it possible to record say 30 seconds of audio and then have LumenVox convert to text ? or any available tool open source for speech to text . Regards Dhaval -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100914/b56c3d9c/attachment.htm

Hi, My problem is as follows. I registered an xlite client and dialed 1500 extension. In the extensions.conf i set as follows. exten=>1500,1,AGI(localhost// hello.agi. This hello.agi when connected plays a greeting message. Once this is connected from the script i want to transfer the call to another extension say 1600. How do i achieve this. I tried using ChannelRedirect but it didnt work.

Hi Everyone, I have a provider whose DID used to come into the box just fine but recently stopped working. Nothing has been changed on our end. Here is what I get when doing "sip set debug peer PROVIDER": Sending to 123.123.123.123 : 5060 (no NAT) ^^^^ That is ALL I am getting with sip debug turned on. With Allow Anonymous SIP set to YES, then the call comes in properly and you see