search for: clickforacall

I need to decide on the best way to add a voip SIP or IAX client to a website. I'm thinking that I'd like it to be inline, like an aplet, on the page. I've got some asterisk servers running to connect up to, so the real challenge is finding an easily integrated open source client. Any suggestions from those who know? Jason

All calls through the system are being dropped when they are bridged (Asterisk, Linux, pure VoIP system). The calling party here's half of the word 'hello' for instance and the call is dropped. I've noticed that hangup() was being called for some time now when the call was bridged, but the call was still continuing. Any thoughts on where to start debugging? Jason

Hello, I have some extensions that are using variables loaded by an AGI program. Everything works fine and I am able to use NoOp to see the value of my variables when using IAX, but the same variables don't work when using SIP. I can provide further details, but right off of the bat does is there something I need to know about the use of user defined variables in with SIP channels vs. IAX

Hello, I have the following simple application... 1. Call is answered, and Dial() function is used with a macro to dial out to a number. 2. 'Called' party answers the phone, and hears a message (this is a function of the macro) At this point I'd like for the 'Called' Party to be able to make a decision and press 1 or 2 to hear some additional information before

Hello, I am using macros to give the ability to a call-receiver to 'accept' a call. However, any keypress connects the caller. Anyone have any suggestions about how to re-engineer this so that the receiver can deny the call, or press other keys to do other actions, without connecting to the user? Thanks, Jason Wolfe

Hello, In watching the console where two calls are natively bridged, Asterisk shows a hangup for each channel (both using IAX). This isn't correct and causes my CDR records to show shorter calls than what actually occurred. Before I go into higher detail, does anyone have any ideas about this? Jason

Hello, I'm using Realtime to select extensions out of a database so that we can provision inbound tollfree on the fly. Once I 'catch' the inbound, I want to "get out" of realtime and use the regular extensions again. I thought I could just use the goto statement and go to another context/entension in the 'non-realtime world'. Is this not possible? Is it all or

Hello, I have some extensions that are using variables loaded by an AGI program. Everything works fine and I am able to use NoOp to see the value of my variables when using IAX, but the same variables don't work when using SIP. I can provide further details, but right off of the bat does is there something I need to know about the use of user defined variables in with SIP channels vs. IAX

Hello, I need to determine what numbers in a call list are cell phone numbers and remove them from the list. I'm sure this is nothing new to most of you. Thanks for the help if you have time to give me some direction. Jason -------------- next part -------------- An HTML attachment was scrubbed... URL: