similar to: Variables losing their value????

Alrighty. I'm having a hell of a time and I need help. I'll try to give as much information as possible. ---------- HARDWARE ---------- MB: Supermicro PDSBM-LN2+ Intel 946GZ ICH7R + Intel? 82573 Memory: Crucial 512Mb, 1Gb, 2Gb modules matched to above MB. Processors: Intel Celeron 420 Conroe-L, 1.6GHz 512KB 64-bit Intel Core 2 Duo E6320, 1.86Ghz 4Mb 64-bit Intel Pentium E2140

Alrighty, Got it patched and working (at least in my one setup). You can pull the patch from: http://denetron.com/~dballenger/ezstream-program.patch (I can post the pre-patched source too if people would like) You'll have to add a "<filetype></filetype>" line into your config files, just put "file" if you want it to work just by reading that file or playlist.

Hi All I have a problem with mixmonitor in 13.4.0 doing the following: 1. Caller phones in 2. Reception picks up 3. Talks to caller 4. Does attended transfer, talks to manager to screen the caller wanting to speak to him 5. Complete the transfer by putting down her handset so the caller can speak to the manager 6. Caller talks to the manager The problem is that mixmonitor only records

> The intent (not yet reflected in the autoconf stuff, and I'm not certain the > code is completely ready for it) is that curl should be optional. If you don't > have it (and have a sufficiently recent version of it), icecast _should_ still > work, but will not have YP support compiled in. All other functionality will > remain. > > This is one of the major todo items

I recently configured a SIP peer which i must specify my fromuser as my ten digit "DID". I send calls to this peer, but whenever Asterisk sends an options message, the fromuser is "asterisk". Is this a bug? Or is there some other config I must make ? register = 2155551941:123456 at 10.0.138.226/2155551941~600 [peer](!) type=peer context=inbound qualify=yes

> On May 4, 2017, at 4:53 PM, David Blaikie <dblaikie at gmail.com> wrote: > > Alrighty, a little fuzzy on how best to implement this - Adrian, you've probably got the most context here as to how to wrangle this. > > My first attempt was in IRMover.cpp, IRLinker::linkFunctionBody - after metadata is copied over, create a new subprogram derived from Dst.getSubprogram,

Good afternoon, I'm trying to write a simple callback context, but i need to hangup an incoming call and then call the origin number back, the problem is that asterisk stops processing the call after Hangup() application then it is not able to dial the origin number back. Sorry for the grammatical erros. -------------- next part -------------- An HTML attachment was scrubbed... URL:

I am building a new Asterisk server here at the office, and I'm wondering if there are any downsides to creating my dialplan with AEL. It seems more intuitive (to me), but I'm not sure if there are any pitfalls I need to be aware of first. We use this for internal extensions, 8 pots lines, and our answering service which gets about 500 incoming calls a day down our T1. Also, one more

Hi! Guess I am doing something totally wrong here: Some smart person could maybe plz tell me what.

I'm currently using the Linksys PAP2, and since there's a shortage I'm looking for different devices. I'm mainly looking at the Sipura SPA sets since they are the base of the pap2. Anyone else have experience using them, and which one? Thanks Sherwood McGowan -------------- next part -------------- An HTML attachment was scrubbed... URL:

In my Asterisk CLI I get Ring/Off-hook in strange state 6 when i make a call into the system, the system claims to answer the call, and do the things in the dial plan, but I just hear ringing on the phone I'm calling in from. I am using a Sangoma A200 4 Port Analog card. my wanrouter version: WANPIPE Release: 3.3.6 asterisk -V: PBXtra Core fon_o_1.2.17 Any ideas? Daniel Lockard

On 04/05/2011 03:06 PM, asterisk-users-request at lists.digium.com wrote: > Message: 12 > Date: Tue, 5 Apr 2011 13:36:21 -0500 > From: Sherwood McGowan<sherwood.mcgowan at gmail.com> > Subject: Re: [asterisk-users] Iptables configuration to handle brute, > force registrations? > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users at

Has anyone else had problems with users being able to press key tones during a voice prompt? I have a few users complaining that some systems will not recognize key presses during them. using current CVS-HEAD, linksys PAP2 UA's, rfc2833 dtmf mode. Thanks Sherwood McGowan -------------- next part -------------- An HTML attachment was scrubbed... URL:

I need help translating extensions.conf to AEL: [default] exten => _X.,1,Set(DID=${EXTEN:6}) exten => _X.,n,Goto(continue,1) exten => _1X.,1,Set(DID=${EXTEN:7}) exten => _1X.,n,Goto(continue,1) exten => continue,1,Noop(${DID}) exten => continue,n,Set(GROUP(IAX)=incoming) exten => continue,n,GotoIf($[${MATH(${GROUP_COUNT(incoming at IAX)}+${GROUP_COUNT(outgoing at

I'll keep this brief because I don't want to come across like any more of an a$$ than I absolutely have to, especially since I know I've blown my stack before..... Gentlemen (and Ladies, if you're out there), If someone gives you advice on this list, and ESPECIALLY if they give you advice offlist, have the courtesy to (AT THE LEAST) to let them know when/if you get your question

Dear All After various failed attempts at using Wine I have installed this from the Mandriva 2006 DVD. I have version ?Wine 20050725?. I have installed Paint Shop Pro as follows: As a user I typed: ?Winecfg? and when in the configuration programme I went to the Application tab > Add Application button > browse to the installation executable: ?/mnt/cdrom/autorun.exe? > Open button

Guys, I've been poking around trying to find a good answer for this via voip-info, google, etc... Haven't found anything that helps, so maybe you mates could. A lot of my customers are using Linksys UAs (router/ATA PAP2) and some using Sipura SPA-2002s. Every once in a while, the customer will get one-way audio. I've read that this is commonly caused by the outgoing RTP port not

Anyone have a good tool(s) to use for simulating a bunch of calls? Benchmarking or stress testing? I only need SIP protocol, and do appreciate any replies...I realize I could google it, but I am looking for opinions as well. Sherwood McGowan -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hey all, I'm trying to create redial on busy for my users, but haven't the foggiest on how to make asterisk route depending on the status code returned over SIP (483, Busy Here?). . . anyone know how to do this? Thanks Sherwood McGowan -------------- next part -------------- An HTML attachment was scrubbed... URL:

Since I can't seem to get anything figured out for the Comedian system, are there any other systems out there that we can hook asterisk into? Sherwood McGowan ViaTalk Level 2 Support VOIP System Engineer -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050912/32ae9d25/attachment.htm