similar to: Bullet-proof FXO?

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

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

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

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.

One of the suggestions I got for dealing with my analog FXO woes was to look into using a Voicetronix card instead of the Digium. They're more expensive but if it worked better it may be reasonable. They indicated they were compatable with the CVS HEAD version of *. Can someone share with me their experiences with Voicetronix OpenLine and/or OpenSwitch cards with *? I'm specifically

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 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

Anybody know of a VoIP (preferably IAX) carrier here in the Atlanta area? Using ones in Chicago and Denver is hit and miss these days. Traceroutes seem to change often and the latency is inconsistent. I'm wondering if someone more local would improve things. -- Paul Dugas, Computer Engineer Dugas Enterprises, LLC paul@dugas.cc phone: 404-932-1355 522 Black Canyon

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

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

Hello all, I hope this is not off-topic, if it is please let me know. I'm currently playing with an Asterisk at home, in order to get to know it's ins and outs. Very very impressive indeed. I've got it hooked up to my home phone line via a Wildcard clone board (Intel modem with Ambient chipset), and it works like a charm. Zaptel picks up the card as a generic clone, and works with it

In my search for a better analog FXO interface, I'm looking into the Voicetronix cards. Can someone let me know if they've been using the OpenSwitch cards with Asterisk? Also, can anybody comment on the differences in FXO performance between the OpenCall and OpenSwitch cards? Thanks in advance, Paul -- Paul A. Dugas Dugas Enterprises, LLC email:

I have searched google and came up with too many options and packages that may or may not work for my needs, most articles seem to be for setting up routers. Maybe someone on the list can give me some better insight. I have monitoring turned on my "shift eight (tm)" (Asterisk ;-)) box for all calls. We have over one hundred agents and tons of recordings in wav format. I also

> 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

While setting my first couple IP phones, I set their voicemail buttons to an extension that runs VoicemailMain. exten => 8500,1,Wait(1) ; voicemail exten => 8500,2,VoicemailMain ; exten => 8500,3,Hangup ; I would like to be able to pass the mailbox number allowing each phone to go in directly but I'd rather tno have

New to asterisk so please be gentle. I'm guessing I'm among a number of recent additions to the list after the article in Linux Mag. I gotta say I'm *very* intersted in the project and will be doing lots of reading shortly. A couble quick questions first... How suitable is Asterisk for use as an IVR providing callers with textual data out of a database? Can it be combined easily

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