similar to: Silly question: Why multiple lines on SIP phones?

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

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

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

I've been struggling with a test * install for a couple months now in a small office and am just about ready to give up on it. It's not that the system itself is a problem. I've got everything (attendant, voicemail, FXS extensions, Cisco and Polycom hard-IP phones, and 2 VOIP carriers) working except for the frigging analog FXO interfaces. These things are driving me completely mad.

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

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

Wondering if someone (Steve?) can clarify something form me. I think the recent "soho fax solution?" thread has mixed things up for me. - Is it possible to get reliable fax reception using a Zaptel FXO interface connected to a standard POTS line and a fax machine connected to station interface? - Is it possible to reliably send outboud faxes in the reverse direction?

Hi there, Do anyone have any experience with SIP phone that support a headset ? We have Budgetone phones but we need headsets. What would you advise ? Thanks Thibault

I'm looking to purchase and deploy a bunch of hardphones for agent use. The phones will have to register with Asterisk and/or SER, depending on where the phones go. They need only one line, G729 codec, and no super fancy features. Preferrably something that is easy to provision. I would think the BudgeTone would be good, but then I've read so many people complaining about them, and some

Has anyone gotten call-waiting to function on the 4 line SPA-841? I've seen some documents that say it can do it, some say no way. If yes, can you share configs / SPA-841 settings? If no, did you work around it? I can call out just fine on all 4 lines, however, if I am on the line, another call coming in does not ring the 2nd line...it just goes to busy / VM. -Darren

>From what i've read i think that asterisk is suppose to generate the dialtone. But when i pickup my analog phone it seems that the Openswitch12 card is the one playng the dialtone. Im almost positive the card is playing the dialtone, can anyone confirm this? And if so how can i change the dialtone?

How long does it take to get a vanity number? I signed up for an account, pre-paid some money, and then placed a vanity number order. I did all of that around Dec. 31st 2004. They said it would take 2-10 business days. It is now Feb. 15th and still no vanity number. I've called them about a dozen times and every time they tell me to keep calling the number to check it and just wait. I'm

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?

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

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

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

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