similar to: ITSP in Atlanta?

Saw an ad in the latest Linux Journal for Sangoma's FXO/FXS analog cards. it was part of an ad for a reseller. I can't find anything on the resellers site or Sangoma's site either. Did the ad jump the gun or someting? Is this for real? Paul -- Paul Dugas, Computer Engineer Dugas Enterprises, LLC paul@dugas.cc phone: 404-932-1355 522 Black Canyon Park

Got a wierd one that's reminding me of a problem mentioned in an earlier post but for the life of me, I can't find it. So... Inbound calls via a Voicetronix interface on my Asterisk box are being properly detected and routed to my dialplan as expected. It's a simple plan right now that rings a few internal Voicetronix and SIP stations. When the inbound line rings, it's ringing

Been using a few different FXO interfaces (X100P, Voicetronix, and SPA-3000) at a couple sites and they've all run into nasty issues at times. I'm looking at moving one of my offices and am in the market for a solution for connecting 4 FXOs. I got excited about recent offerings from Digium and Sangoma in the way of new cards but I hear they're causing troubles for more than a few.

I have a SPA-3000 that is failing to hanging up pretty often; almost every day now. The weird thing is that an almost identically configured (same FW, different HW rev) second unit right next to it isn't having the same problem. Swap the lines and the problem stays with the unit. I've been going round and round with Sipura and their latest message told me to use a procedure detailed in

I had a problem in the past with a SPA-3000 acting funny that Sipura helped me with by telling me how to factory reset it. They responded in less than a day to my email request and the unit has worked fine since. I've had similar turn around on requests related to a batch of SPA-841 phones. They were all handled by real people who appeared very knowledgeable on the products. This appears

This is a wierd one. Can't figure it out. I have an SPA-3000 at the house handling my incoming line. It's setup to direct the incoming call to asterisk. Works great 99% of the time. A few times a day, I'm getting calls that ring once internally and are then hungup. I managed to get a detailed log [1] of what's happening today and it looks to me that the SPA is acting wierd.

I've had an spa3k in service here at the house for a while now. After some initial wrangling, it's been working okay. I've had to reboot it a couple times and have noticed something rather annoying though. My setup is pretty simple and, dare I say, common. I have the SPA-3000 "inline" between my incoming POTS line and the internal house phone. It's setup to deliver

I've got a context in my dialplan like so but pressing 0 doesn't seem to be working. Instead of dropping out to the "o" extension, it's just returning to the start of the direcotry app. Same with star. Anyone see where I've gone awry? [attendant] ; <snip> exten => *,1,NoOp(Attendant: Directory) exten => *,n,Directory(default,attendant,eb)

I've got hints setup for my MeetMe conferences like so: exten => _60X,hint,MeetMe:${EXTEN} and they show up in "core show hints" like so 600 at dialtone : MeetMe:600 State:Unavailable Watchers 1 _60X at dialtone : MeetMe:${EXTEN} State:Unavailable Watchers 0 I'm wondering why they're Unavailable instead of

I've got my 1.2.x Asterisk server registering with a SIP provider using their servers DNS entry, not their IP address. My server is behind a NAT. The setup works like a champ for weeks then I get a call reporting inbound calls are failing. When my server isn't registered, inbound callers get a "disconnected" message; very bad. When it happens, I've found that the IP

Hi all, Anyone got any VoIP traffic shaping rules for m0n0wall that they could let me look at please? Thanks -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com

Hello, How good is :SPA 841 form SIPURA. Thanks Varun

Hi, I still have the same problem trying to configure ITSP failover in extensions.conf for a connected PRI. Any comments thoughts or direction would be greatly appreciated. I sympathize with wanting inbound DID failover. If we have a client with multiple DIDs we will spread them across two or three ITSPs so that all inbound connectivity will not be lost if one of them has an issue. I

Am I doing something wrong here? GOt a SPC-841 the other day and have it registering properly. Can place and recive calls as expected but when on the phone, a second call is immediately dumped to "busy" voicemail. Does this thing not support call-waiting? Or, have I just got my configs wrong? Paul -- Paul A. Dugas Dugas Enterprises, LLC paul@dugasenterprises.com

Running * CVS with a SPA-841 (0.9.5) and can't seem to get the message waiting light to come on automatically. There is a control in the web interface to turn it on and off (seems rather curious to me but whatever). Has anybody got an idea as to where I may be going awry? TIA, Paul -- Paul A. Dugas Dugas Enterprises, LLC paul@dugasenterprises.com 1711 Indian

> Just wanted to let you know you were just left a 2236:45 long > message (number 1) in mailbox 1000 from an unknown caller, on Monday, > February 21, 2005 at 05:38:14 AM so you might want to check it when you > get a chance. Thanks! > > --Asterisk Hmmm... Call came in either via a SPA-3000 or an X100P Zaptel FXO interface. Got 2200 minutes of beep-beep-beep-...; the caller

Announcing a new Meetup for Atlanta useR Group (R Programming Language)! > > What: The Innagural Atlanta useR Group Meeting! > > When: November 5, 2008 7:00 AM > > Where: Click the link below to find out! > > Meetup Description: Hello All! > > I am putting this on the books for the first Atlanta useR meeting. I > apologize fro the delays, but I think everyone will

Hi, I've tried to make toll-free DID work for the last 2-3 weeks. Apparently only the IAX.CC/Sixtel personnel can make a call to my toll-free. Anybody else just gets a busy signal. It takes for them about 5-6 business days to respond to my request. It seems they are looking in the once a week to the High priority tickets. It also seems that they are permanently closed ! No matter when I

I really hate to have to make a post like this, but I feel I have little choice but to relay to the group my experience with Teliax, and explain why I recommend against using them as a primary Voip-> PSTN provider. I hope that a letter like this will inspire companies like Teliax to work harder at customer service, as well as circuit stability. We need more companies that offer the types of

Hi, I'm hoping someone has a suggestion for us. We have an ITSP that sends inbound traffic to us. Unannounced to us last week they started alternately sending traffic from two IP addresses, instead of the one we knew about. Some calls would pass, and others would be dumped as unauthenticated. I added the 2nd IP to the sip.conf file to allow for this, and everything was fine