similar to: Asterisk and RTP streams

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:

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 run a bunch of the Linksys ATA's.. I always use rfc2833 for DTMF. works very well and have never had a problem with it. ..o-------------------------------------------------------o. Brian Fertig NOC/Network Engineer Planet Telecom, Inc. Tampa, FL Office -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of

I am trying to setup one time caller id block on my system(activated when an incoming call matches *811XXXXXXXXXX), and I have had little to no luck. Could you take a look at my context/macro definition and help me figure out what I am missing? Here is my context for my dialplan: include=default plancomment=user-default

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.

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

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 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

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:

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

Alrighty Gents, let's see if any of you have encountered this one...Variables losing their value...I'm setting a variable with four underscores (used to be two, had same issue) so it can be inherited by child channels, and then the next line in the dialplan I use it but it appears to be empty...I've googled and found nothing stating this kind of weirdness.. Asterisk 1.8.2.2 (upgrading

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:

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

Hey all, sorry if this has been covered, but I've not found anything after a couple hours' worth of googling. I can see (and I'm familiar with) all the usual MySQL addon apps once I install Asterisk 1.8.x, but I cannot find any reference to MySQL and the new CEL logging tool other than ODBC. Is this the only method available to use MySQL with CEL at this time? Thanks, Sherwood

Gentlemen, I have googled, searched the mailing list archives, and even spoke on the IRC channel, but have not found an answer to the following problem. I am attempting to retrieve multiple columns in an ODBC query using ARRAY per the solutions offered by many individuals. My dialplan code is as follows: exten => _.,n,Set(ARRAY(var1,var2,var3)=${ODBC_LOOKUP(${KEYVAL})}) exten =>

I've recently finished an add-on module for CEL logging to MySQL, and it needs to be tested. The feature is being tracked at https://issues.asterisk.org/view.php?id=19058 And the patch is available at https://issues.asterisk.org/file_download.php?file_id=29110&type=bug Thank You, -Jonathan Penny -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hey Guys! We had migrate asterisk 1.2 to 1.8 now big issue is queue system. Before we had 3 queues and we were using AgentCallbackLogin but now its quite difficult to use AddQueueMember. Is there any easy way to logged into multiple queue using AddQueueMember ? and restrict agent for specific queue ? -S -------------- next part -------------- An HTML attachment was scrubbed...

Hey all! I'm trying to figure out how to have a queue accept an inbound caller's key press to action on. At first I'm just trying to implement a "Press 1 to leave a voice mail" announced and at any time in the queue, the user can press 1 and go to the queue's voicemail. Later I'd like to have it accept "Press 1 if this is an x issue, press 2 if this a y