similar to: dial, answered and then hangup

As usual, we'll be jawing about any and all asterisk-related subjects with the usual gang and any new people are always welcome, regardless of your level of expertise. You can even come and ask questions, it's guaranteed to be a more pleasant experience than it will be on IRC ;) http://VoipUsersConference.org/topics.php IRC; Freenode.net #voip-users-conference

Hi all, Is it possible to install more than 1 asterisk in a single server? If yes, what do I need to set and take care? Rgds, ango

HI, Does asterisk will ignore the setting in files if realtime is applied, say asterisk will ignore all the setting in sip.conf if realtime table sip_buddies is applied? ango

This is an email to all New York based Asterisk users. For some time it's been bugging me that we don't have a local contact point/user community. If you are involved in Asterisk and in NY/NJ shoot me an email, I'm going to try and revitalize either meetup.com or some other shared environment for Asterisk users in NY. Shoot me an email and once I get an idea of how many

Hi all, I have a TDM400 with all FXO on it. When I make an outgoing call, I can hear callee but callee claims the volume is too low so that he/she can't hear very clear. Can I adjust to increase the volume in callee side? Is it increase the value of txgain can solve the problem? ango

Does anyone know what options you need to use with "sox" to output the audio in the WAV49 format that Asterisk uses.

Hi everyone, How can I add the equivalent of: exten => s,n,SIPAddHeader(Alert-Info: Ring Answer) in a .call file? This is to support paging to Polycom phones... Thanks for all info! Steve

1and1 dedicated server's service has been down for a few hours , unable to reach them by phone or email. do anyone know what is going on there ? Mario -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070731/74328b51/attachment.htm

Hi all I sold new TDM800P card with 8 FXO ports, someone know if can be use this card on AsteriskNOW or trixbox? What can i do for use this card? Thanks. ---------- RafaelCanchola Product Development Engineer, FonetGlobal Inc. rcm at fonetglobal.com http://www.fonetglobal.com Ph. + 52 800 022 10 21 ext. 214 + 52 442 167 08 00 VoIP 523663899 d00d! cyberalph -------------- next part

Hello,

Hi All; Any one can advise for a good stable open source video conference or video server? Regards Bilal ____________________________________________________________________________________ Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs

hi, all I creat new application app_myapp.c for asterisk 1.4.15. I add this in asterisk/apps dir. to load. after compiling asterisk app_myapp.o and app_myapp.so has been created but when i run " show applications" at cli> . my application not displayed. what's wrong??? any suggestion!!! thanks Bhrugu Mehta

I'm having trouble getting an Asterisk server to make outbound calls on my iDCS Trunks. I have idcs station to asterisk station working I have asterisk station to idcs station working However, I am unable to get Asterisk to utilize any outbound trunks on my iDCS.... Anybody have any ideas? ________________________________________________________________ Sent via the WebMail system at

Does anyone know how I can append to different user recorded voice files within a dial plan? For example Asterisk ask caller a question and records the answer, then ask another question record the answer to the end of the first answer - so when it's played back, all the answers are in one playback. TIA Bart -------------- next part -------------- An HTML attachment was scrubbed... URL:

Morning All, Has anyone here successfully implemented skills based routing within queues? The concept behind skills based routing is fairly straight forward, and I know I could do it with multiple queues, agent penalties and a bit of AGI to put the call into the right queue. However doing this is going to require the addition of several extra queues and isn't a very clean solution. The

Hi all, Im using Asterisk 1.4.11 and I want to proceed some time and date operations in my dial plan. (for a time shifted callback). Should look like: CURRENT TIME + x minutes. Of course it should increase the hours for example in this case: 10.59 + 5 minutes = 11.04 I guess I've to use the math function in 1.4 but how can I manage easily the time operations? Kind Regards, Erik

Hi, I know that Asterisk doesn't support Instant Messaging, but I'm trying to use the AGI function RECEIVE TEXT to implement a kind of IM service. I have a sip softphone that tries to send a message to an active channel and the AGI script that expect to receive the text through the STDIN. Two problems arise: First: How can I say to asterisk to get the message? (I see on CLI console that

Hi All; Is there an GUI for Asterisk that can help in showing the call flow (who is in progress, who is connected, called number, ...)? I was think in AsteriskNow does this? Any advise? Regards Bilal ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now.

Dear all I have asterisk 1.2 configuration and it is working fine but thing is that i have alread Avaya setup and i have intergrate my Linuxbox asterik with Avaya system avaya already use 4 digit dialplan (1644 example ) and in asterisk i have configure 2 digit dialplan ( 44 example ) now i want to configure 4 digit dialplan in asterik without any change in avaya or asterisk so

I have a PRI line and I am having problems setting the ringtimeout on the dial application to more than 29. If I set ringtimeout to 29 on the dial application call and I do not answer the ringing phone then I correctly get DIALSTATUS set to NOANSWER. If I set ringtimeout to any value over 29 on the dial application call and I do not answer the ringing phone then I go to extension h and have