similar to: Sipura 3000 x special dialling pattern (pin code)

freebsd quite current ports tree 1.01 asterisk spa3k at 2.0.11(GWg) for calls in from the pstn side of an spa3k to asterisk, i get sporadic short beeps. they are not related to sip re-reg time, which is all that has occurred to me so far. calls in from the fxs side of the spa3k and out through nufone do not exhibit the beeps. calls from the fxs side of the spa3k out the fxo side do have the

My SIP phones can dial out through Sipura SPA3k to POTS for local and 911 calls _but_ incoming POTS calls are being swallowup somehow. Am I on the right track with the code snippit below? sip.conf: --------- In sip.conf the following code is _supposed_ to ring the SIP phones when a POTS line call comes in through Sipuara to Asterisk. [spa3k-pstn-in] ; Pots-line-in from Sipura ; If

In my seemingly endless search for the cause of echo on my SPA3000, I wired it up in the following configuration: Analogue Handset <--> (FXS)SPA3000(FXO) <--> PAP2 And set the Line1 dialplan on the SPA3k to '(<:@gw0>S0)' which means that as soon as I pick up the handset I get linked straight through to the PAP2, which gives me dialtone. Even in this configuration, with

I'm trying to get app_swift (v0.9.1 from http://www.loopfree.net/app_swift/) working, but it's having issues (see below). I'm running 1.4.0beta3 on FC6. Any thoughts? *CLI> -- Executing [100@internal:1] Answer("SIP/spa3k-fxs-08e884b0", "") in new stack -- Executing [100@internal:2] Swift("SIP/spa3k-fxs-08e884b0", "Diane^your text

Hello, I have a Sipura SPA-3000 connected to my PSTN line and forwarding calls to my Asterisk box. It is a SIP peer "pstn-spa3k". I have setup "call-limit=1" in the peer config. When a call comes into Asterisk I get the correct "inuse" values but the hint isn't updated: sprite*CLI> sip show inuse * User name In use Limit * Peer name

Thanks Rich, I have an SPA-3000 laying around, so I will attempt to set it up in a little more conventional manner (although your method looks like a winner for a home test PBX). Would you mind posting or PM your current config to me, maybe screenshots if you PM. If I start with that it will take less time to get to the point where the SPA-3000 is a true FXO-FXS gateway for *. I will be happy to

Voxilla.com has a great config wizard for the SPA-3000 and * http://voxilla.com/spa3kasterisk.php I took the output from this wizard and dumped it on my test box with an SPA 3000 (with some mods to match my * contexts) and everything worked great. Calls from the PSTN to the spa3000 are routed to dialplan #8 on the spa3000, which dials * Both the FXO and FXS port register with * The SPA3000 is

Using a spa3000 with asterisk cvs head, and the spa3k is config'ed with a dialplan that essentially routes any call starting with an "8" to asterisk. All other US 7 and 10 digit calls, 911, etc, route via the spa3k's fxo port. Is there a way in extensions.conf to: - inspect the dialed exten number, - if first digit is "8", drop the 8, - continue through each

Does anyone have personal experience with a 5.8gig wireless phone (system) that has an MWI that WORKS with asterisk via fxs (in my case spa3k) generated MWI. I know the spa3k does stuttered dialtone but not sure if it generates FSK MWI. I see some that state they do but I also see reviews that say they don't. Doug

I'm trying to dial for example *67 to block caller ID but is it not working. I have SPA-3000 an dial plan: exten => _9.,2,Dial(SIP/${EXTEN:1}@pstn-spa3k,60,tr) If I try exten => _3.,2,Dial(SIP/*67${EXTEN:1}@pstn-spa3k,60,tr) it gives me congestion. How to pass "*" in a context when dialing a number? -- #Joseph

How to implement some of the function into asterisk like *67 "call number blocking" or *57 "call trace" ? I'm connecting to sipura SPA3K outside line by dialing 9+number. Is there a way to get outside dial tone in SPA3K PSTN-Line by dialing "9"? How to program the extension? -- #Joseph

Hello all, I have an ISDN termination box (TR1) that converts ISDN(Bri) to 2 normal analogue lines. The same number is assigned to these lines. These lines are connected to 2 spa3k registered to my asterisk box. When calls arrive, TR1 try to pass call to the first spa. If spa not takes the call immediately then try to pass to the other spa. The only configuration I found works is to put the

Is anyone using a cisco router as an ISDN gateway with Asterisk? As you might have seen from a couple of my threads, I have been looking at Fritz! and Cologne cards, both of which require development against a specific version of asterisk/zaptel (e.g. chan_capi), which is intrusdive and causes a lag in deployment. I was thinking a better approach might be to use a seperate gateway, such as a Cisco

I'm using "insecure=invite" with two different dial plans, it it working with one dial plan but not with the other. What other parameters could influence "insecure=invite" In sip.conf below "insecure=invite" is working OK [pstn-1270] type=friend secret=spa3k username=voice-1270 mailbox=369 host=dynamic insecure=invite canreinvite=no disallow=all allow=ulaw

Hi, I use * release 1.0.9 with differents phones and softphone, i've got a problem with my Cisco 7940G (last SIP Firmware). Sometimes, when i but a call on hold, the caller has got the music, but when i "resume" the call, then the caller does not hear me (and nothing at all)... I must wait for 10, 20, sometimes 60 seconds before he could hear me again. Any body already had

Anyone tried to muck around with using the 488 for both fxs and fxo with asterisk? I've been playing with one for the last couple of days, and it looks like its a little lower quality then the spa3k. No gain settings, echo canceller is less then ideal on long analog pstn loops, etc. Anyone with good experiences?

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 need to be able to connect an old PA system to an asterisk box, which basically works as a couple of amplifiers taking an analogue phone signal and playing whatever it produces out of some speakers. There is no on-hook state in the whole setup. Obviously If I just connect the input to a port on an ATA, I'll just get a dialtone played through the speakers. Can anyone think of a way I can

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

Hi, I have my Sipura SPA-3000 setup with Asterisk as follows: [spa3k_line1] type=friend context=home secret=PASSWORD host=dynamic dtmfmode=rfc2833 dissallow=all allow=ulaw When an incoming call comes in, I have a Zap interface in Asterisk which just does a Wait,15 then answers with voicemail. The SPA-3000 detects the PSTN call and makes Line 1 ring - so I can answer the phone if