search for: spa3ks

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

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

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

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

I need to place a call using a "pin code". To access an external line, the host PBX (a Ericsson MD-110) will require that I dial *72*pincode#phone_number to complete any (trunk) call. When I send the number, my Sipura 3000 will reject the call with "Forbidden - wrong password on authentication for INVITE" (see below). All other calls sent to the Sipura box without the

My apologies for the off-topic post ... No matter what settings I try, when I dial in to the SPA-3000 on the PSTN line, it picks up the call and immediately gives me a fast busy tone then hangs up. The info tab says under PSTN Line status: Last PSTN Disconnect Reason: PSTN Disconnect Tone which seems to indicate that the SPA thinks the caller has hung up. Since I am in Japan, it is possible

Can anybody show me a working Sipura 3000 setup please? I need to setup one to my * box, ... What are the variants you can setup? Advantage - disadvantage. bye Ronald Wiplinger

Arrggghh. Tearing my hair out here. I'm trying to set up the spa3000 in the UK for my home, and want * to control the dial plan I've googled to no avail. I've read the manual to no avail. Can someone, please let me know what the parameters is the spa and * are to a) receive a call from the pstn b) make a call to the pstn from the phone attached I can make sip to sip calls (i.e. I

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

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

Hello, Just want to tap the collective wisdom of this list as to experiences pertaining to the Handytone HT488 and the Sipura SPA-3000 adapters... Basically I'm looking for a FXO/FXS/LAN ATA and these two seems to be the top of the pick..Any comments and experiences esp. with Asterisk compatibility would be great, before I plonk in the bucks. TIA. /wai-sun

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

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

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

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.

If I have multiple Sipura 3000 device how can I dial out properly? I can receive call without any problem and that's working really well. Caller ID is shown and when someone call he get's the welcome message the same way I have it configure with the X100P card. I don't seem to have any echo problem with the Sipura 3000 (but I do with X100P cards) My main concern is for

I can't seem to get music on hold working, it tries to work, but I just hear strange noises on the extension.. Here is some debug info. Looks like mpg123 starts fine, but I hear nothing. I'm on todays CVS build. -- Executing Answer("SIP/2203-062c", "") in new stack -- Executing MusicOnHold("SIP/2203-062c", "default") in new stack --

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