similar to: AGI Programming

Hi all, I'm seeing a lot of these messages: [Apr 30 20:28:57] WARNING[5402]: chan_sip.c:12543 handle_response: Remote host can't match request NOTIFY to call '2e4a02807750b7024d5ff09c668fa0f7 at 10.0.0.2'. Giving up. [Apr 30 20:28:57] WARNING[5402]: chan_sip.c:12543 handle_response: Remote host can't match request NOTIFY to call '0755ad8f40b9d09d491b635e70bb8905 at

I'm just trying to make a real simple Survey via php. Just want it to play the Question Files, wait for a response, save the response into the correct variable and then email it all. I have no issue playing the audio or emailing. But I can't get it to wait for digits or to properly capture those digits into the variables. I know the code is technically right since the emails have this

I lurk and comment a little on here and have been playing with * for a short while. I am interested in hearing about the pros and cons for using a database backend to Asterisk. My current setup is simple, out of the box with config files in /etc/asterisk and logs etc going into /var. I notice a great many of the contributors here seem to use a db backend (is this also called Real Time

Hi, Matthew, all nice to hear about interest from Packt Publishing for LLVM I think there are a lot of professionals in this mailing list, who can be an author of the book. But writing a good book demands a lot of time, so that's why there are not so many responses (and also maybe because of well-formed and updated online documentation, which can be a good replacement for the book).

Hi, I have three Snom M3s at the moment but getting pretty fed up with the issues :( I am UK based and would be interested to hear of other peoples recommendations. Key features :- * VM Notification * Good Range * G729 codec support * Common/Private Address Books per Handset(s) TIA, Best Regards, -- This message has been scanned for viruses and dangerous content and is believed to be

Sorry if this appears twice - I sent it yesterday and it hasn't yet arrived so trying again: ====== Hi all, I have a Polycom C100 communicator and from what I can find about it, the real work done to make the audio as good as it is, is done in the software they install on Windows XP. It is a USB connected desktop handsfree mic/speaker device and it does work on Linux O.K. but I get some

Hello, Min. I suppose I should ask you this: what do people need to know before working with LLVM? Could a book work with that and link it to LLVM specifically? Thanks, Matthew On Sat, Aug 15, 2020, 7:06 PM Min-Yih Hsu via llvm-dev < llvm-dev at lists.llvm.org> wrote: > Hi Matthew and all, > > Also I feel like one of the obstacles would be the fast-changing APIs. > Since to

Dear Sir, Can someone help me please to find a free ebook talking about AGI scripting through asterisk? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090218/a59fc299/attachment.htm

Hello All, I am Sumeet Sawant, an Author Relations Executive with Packt Publishing. We specialize in publishing IT related books, e-books, and articles that have been written by experts in the field. We have a new book project, _*Getting started with LLVM core libraries*_, and we are looking for people with the right expertise. Could you please let me know if you anyone of you find it

Dear All, Did anyone read these 2 books? https://www.packtpub.com/application-development/llvm-cookbook LLVM Cookbook | PACKT Books Over 80 engaging recipes that will help you build a compiler frontend, optimizer, and code generator using LLVM Read more...<https://www.packtpub.com/application-development/llvm-cookbook>

Hi there, in case anyone is interested, I've just taken ownership of a small home network (3 handsets) of the brand new Siemens DECT PSTN/VOIP phone. It works great with Asterisk. Here's my overview and review so far... http://www.theopensourcerer.com/2008/04/27/siemens-gigaset-685ip-phones/ Cheers Al -- The way out is open! http://www.theopensourcerer.com

Anyone has an idea what happend to voip-info? it stopped working about 24 hours ago. Nir S -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070314/23cdc0f6/attachment.htm

Can I do this? I have a x100p card on my PSTN line and I have an incoming context for these calls which uses the "s" extension. I'm wanting to set up a simple IVR and would like to be able to test the dialplan as I go. But having to dial-in on my PSTN line each time is going to cost me money. Can I connect to my zap_incoming context from my locally connected SIP phone? I'm

Hi All, I am writing to you for Packt Publishing, the publishers of computer related books. We are planning to extend our catalogue of books based on Scientific Computing Tools and are currently inviting authors interested in writing for Packt. This doesn't need any previous writing experience. Just an expert knowledge of your subject and a passion to share it with others is all that we

Hi All, I am writing to you for Packt Publishing, the publishers of computer related books. We are planning to extend our catalogue of books on Open Source System and Network Administration & are currently inviting asterisk experts to write for us. So, if you love Asterisk and fancy writing a book, please write to us with your book ideas at author at packtpub.com. Even if you don't

Hi Everyone, I'm writing to announce my new R beginner's guide book and answer questions related to it. The primary focus of Statistical Analysis with R is helping new users become accustomed to R and empowering them to apply R to suit their own needs. It is a beginner's guide written for a broad audience and should be well received by businesspeople, IT professionals, researchers,

Hi All, Had anyone got an idea of there exists an LDAP backend for SIP and IAX? I've read that there is a patch for LDAP realtime, but I hadn't seen any type of relevant configuration information. Any information on the above would be highly appreciated. Regards, Nir S -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, On my asterisk I have a voicemail for my extension and my Soft SIP phone tells me when there is mail for me. I also have other voicemail boxes which are not tied to any specific extension - rather they are for incoming callers when they interact with an AA, such as "to register for the event please press 1 and leave your details"... Can I chain or aggregate voicemail boxes so

Hi all, Ok, I've been bashing my head for a few hours now on this, trying to figure out if I've done something wrong, but everything seems to me hunky-dory. So here's the deal: 1. I've compiled the spandsp 0.0.2pre10 source code successfully and also the asterisk application associated with it. 2. Receiving a fax at asterisk works fine, at least appears to be working

Hi All, Stupid and silly question - is there a way to limit the number of concurrent calls an IAX client can make? something in the similar sense of incominglimit and outgoing limit on SIP? Regards, Nir S