similar to: Other Voicemail systems

Is anyone else with ViaTalk experiencing an outage right now? My DID has been down since 5AM (8/20). Asterisk is unable to re-register or connect for outbound calls. I have also tried calling support and their number gives a fast busy.

since delete is a reserved word, what do you name a column in your voicemail options table to allow setting of the delete option for realtime voicemail? Anyone? Sherwood McGowan ViaTalk Level 2 Support VOIP System Engineer -------------- 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:

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

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:

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:

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

Sorry, sent with wrong account....read below _____ From: Sherwood McGowan [mailto:sherwood@viatalk.com] Sent: Tuesday, August 23, 2005 8:34 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: SIP DEADLOCK Anyone using a CVS-HEAD pulled later than 8/13? We're runnign a downloaded CVS-HEAD from 8/13/2005 and getting SIP Deadlocks like crazy.....

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 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

I listened to all the demos you showed. My ear discerns a little muffling and minor "slushiness" in the GSM files you sent, along with a much more narrow bandwidth, mainly on the high end side, and Allison either has a mild whistling s or slushy s sound in her voice or the producer didn't properly compress it to "de-ess" the recording. Or, I could just be rather tired.

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

I am using viatalk as my voip provider and they use dtmf=rfc2833, but asterisk is not seeing any of the dtmf. I am using CVShead as of 8/26/05. Nothing in the logs indicates a dtmf is being seen. If I use my pots line it sees it fine. Any assistance would be appreciated. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici

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

I haven't been able to find an answer....and got no response whatsoever to my previous questions concerning it. Has anyone found a fix for the remote connections to the CLI causing crashes? Also, is there a known limit? I have a huge need for using asterisk -rx in scripts, which seems is kinda why the -x option as added anyway... Anyone? Sherwood McGowan -------------- next part

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

Hello: Has anyone on this list had to configure hairpinning on a Cisco gateway running IOS 12.2 or 12.3 and using a PRI for connectivity to the PSTN? If so could you tell me how it is done? I'm told this is the source of my call transfer problems and yet I cannot find clear instructions for how the configuration is done. Thanks,Steve -- ISC Network Engineering The University of

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