similar to: Does the SPA-3000 get rid of echo that the X100P can't?

My SPA-3000 finally arrived and I'm trying to get the FXO port on it to work as if it was a X100P card as far as Asterisk is concerned. I have Asterisk dialing out over the SPA-3000 FXO port no problem. The issue I'm having problems with is having the SPA-3000 automatically forward all incoming PSTN calls to the Asterisk "mainmenu" context (or ext I guess). Currently the

In my hunt to track down my echo issues, I tried disabling all echo cancellation, suppression, adaption, on my SPA-2000 (Advanced section of the config, under Line 1/2). Then calling from one local extension to another. (SPA-2000 Line1, to Line2 on the same device) I was pretty shocked with the results, the echo was HORRIBLE! I even tried 3 different analog phones. Now, once I turned the echo

I recently upgraded from a few month old CVS version of Asterisk to v1.0.1, and dialing out through my SPA-3000 stopped working. Notice right after INVITE, in the old CVS version, it includes the number I'm trying to dial (8019596) which works fine, however in v1.0.1, it doesn't include the number and of course the dial fails. Did a config option change out from underneath me or

I'm trying to send a fax to my asterisk box, however shortly after connecting the fax machine reports a "communication error" and hangs up. Below is the error I get from RxFax: Fast carrier training failed Nothing is written to the file system as far as the tiff is concerned. Any ideas on how to fix this? Thanks ---- Sending Fax Machine: HP PSC 2000 Series 2210

I have a P3-800 with two IDE drives in a software RAID1 configuration. Each drive is on a separate IDE channel. Now anytime there is HD activity, I hear "beeps" and "cutting out" on a call using the X100P card. I ran the zttest program, and discovered HD activity would drop the accuracy down to between 2% and 50%. However I noticed if I disabled one drive in the RAID1

I am having trouble getting transfers to work when a zap channel is part of the call. I have a couple SPA-2000's and some X100P cards as my setup. This is what I'm trying: Dial number from phone: -- Executing Dial("SIP/206-2c61", "Zap/1/#######") in new stack Currently on call: -- Called 1/####### Press flash to place call on hold with SPA-2000: --

This is my setup: SPA-2000 -> Asterisk -> X101P (x4) -> PSTN 3-way calling works fine if I use flash and dial just local extensions. Or even if I use flash and dial one local extension, and one remote party over the PSTN. However, as soon as I dial from my SPA-2000 out over the PSTN, and hit flash the call hangs-up about 50% of the time. The other 50% of the time it puts the call on

Over the last couple weeks I've tried everything I could get my hands on in an attempt to get rid of my echo problems. Using a CVS checkout of just yesterday, I've tried every echo cancellation routine in zconfig.h (including Mark2 w/Aggressive) , as well as the echotraining=800 mentioned on this list just last week. While some things worked better then others, I would consider none

Has anyone seen this problem before? I have a server with a single X100P card. The audio level is a low, but if I raise the gain to more than -2db (Rx + Tx) it starts to oscillate in an echo test. Not at a high frequency but with a noise that is best described as a steam engine starting up. It then starts to clip and crackle. If I bring the gain down to Rx=-2.0 and Tx=0.0 or lower then it settles

I get a C++ preprocessor sanity check error when trying to configure spandsp 0.0.2pre6: checking for dlfcn.h... yes checking how to run the C++ preprocessor... /lib/cpp configure: error: C++ preprocessor "/lib/cpp" fails sanity check See `config.log' for more details. Config.log: -------------------------------------- configure: failed program was: | /* confdefs.h. */ | | #define

I have no idea what this really should be called, so for lack of a better name, I called it "reverse hold". Hopefully someone else can make use of it, or even make it better, as its the first thing of its kind I've made for asterisk. Like most people, I'm very busy, so when I call other companies, sitting on hold really sucks. If you have speaker phone, its not so bad, but then

I just added a second X100P card to my * server, altough it seems to be working * seems to be ignoring it: zaptel.conf: ----------------- fxsks=1-2 loadzone=us defaultzone=us zapata.conf: ------------------ context=inbound-analog signalling=fxs_ks group=1 channel => 1 channel => 2 I created a couple of test extensions: ; test extensions

I was just on 70minute call (IAX2 -> Internet -> IAX2) and during that time I heard several "pops", or "clicks". Each time it happened, I saw the following message: Aug 19 15:36:36 NOTICE[1173711792]: rtp.c:429 ast_rtp_read: RTP: Received packet with bad UDP checksum Any ideas what causes these, and why they turn in to a "pop", instead of just silence, or a

Greetings, What is the recommended settings for using SPA-3000's FXO port for dialing out to PSTN in regard of the DTMF? The voip lan contains SPA-2100 and SPA-3000, with all fxs/fxo ports registered to the Asterisk box with unique username/passwords. The inbound PSTN DTMF works excellently, e.g. people calling from PSTN into the * box are able to pick IVR items with DTMF reliably. The

For some reason, starting just today, 1 out 3 of my asterisk servers is having issues calling 1 other server. The only issue I see is that when it registers with the problem server it is using port 1027, not 4569. ie: Registered to 'Server 1', who sees us as 'Server 2':1027 Server 1 then proceeds to timeout trying to register with Server 2. The way I have each server registering

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've been running with a generic X100P for 5 or so months and every once in a while I have problem receiving faxes. I see that others have the same problems and some worse than I have with these boards so I was wondering if using a Sipura SPA-3000 would be any more reliable. Has anyone had enough experience to tell me if that would definitely fix the random fax error. PS. I have * at

Hello, I've been experimenting with Asterisk 1.0.6 and a Sipura SPA-3000, and I've run into a couple of questions I haven't yet found clear answers to: It appears that the SPA-3000 has no call progress on it's FXO interface? Asterisk considers a dial() as answered when the SPA-3000 has dialed the number on the PSTN line, not when someone has answered a phone on the

Since the 3000 has been out for a little while, has anyone done a review of the product? (couldn't find anything on google for wiki). Can the fxo and fxs ports be used as two independent channels? Is it really read for prime time? Etc. Rich

The party's over folks, the new official cisco/linksys/sipura policy is to no longer sell SPA-3000's to end users. Buy them while you still can :-( -Dan