similar to: IP of SIP server changing

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

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 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

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 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

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

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

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

Been tinkering and found Zapateller appears to be answering when I didn't expect it to. I have something like so: [incoming] exten => s,1,NoOp ;Zapateller(answer|nocallerid) exten => s,2,PrivacyManager exten => s,3,Dial(${RING},20) ... I have a 1x1 analog * installation with a couple IP phones too. I've got the FXO interface connected to the home phone line. When I get an

In case it helps others, a couple gotchas that I had to dig and beat my head against to get past... - building on FC2, use "make linux26" when building asterisk - also for FC2, a couple links in /usr/src named linux and linux-2.6 and pointing to /lib/modules/2.6.8-1.521/build/ were needed to build - yes, the power connector on the TDM400P needs to be connected to the power

Looking to outfit a 8-person office shortly and am looking for IP phone recomendations. Saw recent threads and the wiki with all kinds of suggestions and notes on configs but little on the "feel" and sound quality. I have a PolyCom SoundPointPro analog desk/speaker phone and the Cisco 7940 here. How to the other business-class phones compare, from a user's point of view, to these?

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:

Seems strange to me that I should be having this problem but I'm stumped. I'm still playing with a Voicetronix OpenSwitch6 card and Asterisk (development CVS). I using a dialplan that's derived from one on another running installation that's using Digium FXS/FXO interfaces instead of the OpenSwitch card. It seems that as soon as a user on an FXS interface dials 7 digits, a call

Hi all, Happy Wednesday! Got a 7940G off eBay to tinker with and, per the wiki, contacted Cisco about a service contract. Being just little ol' me, they won't deal directly and blew me off (I hate dealing with Cisco, Nortel, etc as a small-business person...) Again per the wiki, I hit CDW.com and found CON-SNT-CP7960 for $9.14. Looking for support for the 7940 (not 7960), I just

I've got a single IAXy installed in a little office nearby and got a call from someone on site a finew mintues ago. Apparently they couldn't make a call on that extension. They'd pick up the phone and get nothing; no dial-tone. Running "iax2 show peers" showed the unit still registered and I could ping it too. Didn't try to call it while it was doing this; sorry. The