search for: johnv

Don't buy this book for its content. It's a waste of $40. However, it is useful to wave in front of my customer's faces to show them that Asterisk is real. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040724/5ccf4012/attachment.htm

Hi! My application needs to look up (by spelling) the first and last names of a person and then insert the corresponding pre-recorded audio file to personalize the message. E.g. "Hi, John Brown. Your book is due back at the library." So I need "John" and "Brown" in audio files along with LOTS of other names - Do such databases of sound files already exist or do I

Has Asterisk ever been audited for common security holes, such as buffer overruns? A quick grep through the source for routines that should never be used, like strcpy, strcat, etc., reveals a lot of it. I fear I fear. Has anyone flung pathology at IAX2 to see if it stands up to malformed packets? (This is always an issue when you have a protocol that only a small number of programs use ...) I

I've configured my [*] to dial the pstn which is working like a charm. I've also configured an extension to ring when the PSTN line is ringing which is also working brilliantly, but, sometimes it doesn't detect that the call has been hungup. I've had a look on voip-info and checked the conf files but I can't see anything that I've missed. Cheers Matt

I upgraded to the latest stable version of 1.0 today and am still seeing the *8 problem where the phone that was originally dialed keeps on ringing even after another phone picks up. Are other people also seeing this? Has somebody figured out how to make this go away? Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL:

Here's the scenario: 1. I call out through * using a X100P card to somebody. Then I transfer them to a MeetMe conference and that all works. 2. After the conference is over everybody hangs up but "show channels" shows that the Zap/1-1 channel is still in use by MeetMe and the analog line is not freed up for re-use. Ever. Any clues? Thanks!

I've posted a couple of docs that are intended to be helpful at http://www.rainiernetworks.com/29535/33701.html 1. A case study of one of our customers using a fairly straightforward T1 coming in to the Asterisk box 2. The installation steps we followed 3. The conf files Yes, yes, I know. Put them on the wiki. -------------- next part -------------- An HTML attachment was scrubbed...

I'd like to programmatically do the equivalent of a *8 using either the Manager or AGI interfaces or some other Asterisk interface. In this scenario, a line is ringing, the "pickup" program executes with an argument that is the extension to connect the incoming call to. That extension starts to ring and, when answered, the incoming call is automatically connected to the extension.

Has anyone created any vertical applications, e.g. real estate, for Asterisk? I'm trying to market * in my area (Seattle) and would like to offer vertical apps to my customers. These apps will help me compete with the big guys like Cisco, Avaya, etc. If you have an app, I'd be interested in discussing how my company (www.rainiernetworks.com) might license it from you. The app does not

Anyone else going to this and interested in having an Asterisk birds-of-a-feather breakfast or dinner get together? Open Source Convention, Portland, OR, July 26-30. See http://conferences.oreilly.com/oscon -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040519/190e9bfb/attachment.htm

I moved from 0.7 to Stable recently and my dialing out/in from iConnectHere stopped working. I made no changes to my conf files or anything else (other than the upgrade). Now I get a "Unsupported Media" error even though I'm still using ulaw and alaw. It stopped working with my softphone, my Grandstreams, my Snom, etc. Searched the lists for a recent discussion of this but

I can't talk through iConnectHere. The connection gets made but as soon as any sound is transmitted the call ends and the Asterisk console shows an "Unsupported Media" error as follow: Got SIP response 415 "Unsupported Media" back from 213.137.73.147 My only allowed codecs are alaw and ulaw. My sip.conf looks like: [iconnect] type=friend secret=xxxx username=yyyyyyy

I have a local company that wants to use Asterisk for a small call center. I've told them that they can do so but they'd like to actually talk to somebody that is. Does anybody know anyone using Asterisk in a call center that they could talk to? If you're an Asterisk consultant this might be an opportunity for you. Please respond off-line. Thanks, John V. -------------- next part

Anybody connecting to on-premise modems by dialing in to Asterisk and using an extension to reach the modem? How? Sipuras, FXO cards, other? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040629/92ee113a/attachment.htm

Just received my copy of this book today. Based on a cursory examination, "VoIP Telephony with Asterisk" (VTwA) by Paul Mahler looks like a very decent book for anyone starting out with Asterisk. More knowledgeable readers may also benefit from perusing its 290 pages as well. Unlike "Asterisk for Small Office Setup" (which I slammed in a review a couple of weeks ago) VTwA has

See my previous email but the book is worse than I thought. In addition, to the things I mentioned earlier. 1. The table of contents is not a table of contents. The only chapter heading is "Chapter 1". It is impossible to tell what is in what chapter. 2. The index is not alphabetized! This renders it useless. Any decent word processor will create a better index with a push of the

The configuration I'm building replaces an existing PBX with Asterisk. There are 8 existing modems that people use to call in from the outside to connect to PC(s) on the inside to transfer data, etc. Callers access these modems by calling the main number and then dialing an extension for the modem they want to talk to. What are my options for supporting these modems with Asterisk? Here are