similar to: [OT] SPA-3000 loudness

My BT100 won't register with my Asterisk server, it always comes back with a 403. I've included my sip_additional (only one to to have the username 2201) and a portion of the sniffer trace (packets 27 & 28). This has me puzzled as I have my SPA-3K working (incoming and outgoing). On my BT100 I get no dial tone, I can't call it (asterisk says the extension is busy) but I can call

I recently lost my setup (bad drive) and I'm now trying to get my setup back. I have Asterisk setup to a BT100, a Cisco 7960 (7.2 SIP) and an SPA-3000. I can call the phone extension, I can call from the phone on the SPA to other extensions and I can call out to the PSTN. What I can't do is to call from the PSTN to through the SPA to Asterisk. It rings twice then I get fast busy. I

I'm afraid I'm quite confused by what I've found on the Wiki. I have the following dial plan that works: exten => 2201,1,Dial(sip/2201@gs1.uucp,20,) exten => 2201,2,Voicemail(u2201) exten => 2201,3,Hangup exten => 2201,102,voicemail(b2201) exten => 2201,104,hangup When the phone is in use it goes to voice mail as busy. When not picked up, as

OK, this may seem to be an obvious question but where do I find the reference docs? I'm getting this error message: Timeout, but no rule 't' in context 'home' about this line: exten => 2201,1,Dial(${PHONES1},20,Ttm) I know the problem is with the 't' but I don't know what the parameters mean. I looking for a man page basically. -- Linux Home Automation

I'm seeing all sorts of problems and it's probably more of my lack of experience than anything else. I have a BT100 running 1.0.6.7 code. When I go to the status page it says it's not registered (hmm, that's not good). I also can't get dial tone but I can dial! I'm afraid I'm lost any good pointers? I've done a sip debug and all I'm seeing for the BT100 -

I have IAX2 (FWD) partially working. I can place calls from my Asterisk box but I cam unable to receive them (comes back as busy). I have my firewall forwarding the udp ports 5060, 4569, 5036 and 10000 thru 20000 to my asterisk server. I think I have the firewall correctly setup as I can forward other services to their appropriate servers. I have no mail box on the one account (the one I'm

I had my setup working properly under 1.2 and after a disk crash I decided that I wanted to try Asterisk 1.4. So far I can transfer between phones and I can dial out. What I can't get working is to get an SPA-3102 to 'send the calls' to Asterisk. I have the device added to the sip.conf file and it shows up in users and peers (but not in registry). Where do I start with debugging? --

Does the ${BLACKLIST()} function allow for values other than 1 to be returned and if so how can I use that is the AEL? Can I use the function in a switch statement? -- Linux Home Automation Neil Cherry ncherry@linuxha.com http://www.linuxha.com/ Main site http://linuxha.blogspot.com/ My HA Blog Author of: Linux Smart Homes For Dummies

There is finally an introductory book about Asterisk! It looks like Paul Mahler at www.signate.com wrote it with a lot of help from Digium. I looked at the sample pages, it looks great. __________________________________ Do you Yahoo!? New and Improved Yahoo! Mail - Send 10MB messages! http://promotions.yahoo.com/new_mail

I heard Sipura had really awesome documentation on the SPA-2000 and SPA-3000, but you have to email them for it. When I did, they said I had to get it from a reseller. It's been a while since I bought my units, I don't even remember where or who they were bought from. Can somebody email me the documentation for these devices? I'm quite interested in knowing what every one of their 200+

I was finally able to compile asterisk with cdr_odbc.so. But now for some reason I get that error: *CLI> load cdr_odbc.so Jun 22 16:38:53 WARNING[-1084309376]: loader.c:240 ast_load_resource: libiodbc.so.2: cannot open shared object file: No such file or directory Unable to load module cdr_odbc.so But the file is there... # ls -lag /usr/local/lib/libiodbc.so* lrwxrwxrwx 1 root

Hello I have just spent a happy day (!) configuring Exim4 to accept authenticated connections from full blown Outlook clients. That works. Unfortunately, by ticking the "Use SPA" box in Outlook it now demands SPA on the POP3 server as well. The existing POP3 server doesn't support this style of authentication. I believe Dovecot will. We're already using it as an IMAP

Hi, I have a SPA-3000. My telco sends no disconnect indication whatsoever, so I am having trouble making the PSTN line detect disconnect. What I want it to do is detect the very loud off-hook tone that is generated about 20 seconds after disconnect. It's not perfect, but it's a lot better than silence detection. I noticed that the zaptel drivers can detect this with busydetect, but

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

Hi All, I'm on of those brave souls who bought into the preproduction beta of the Sipura SPA-3000 FXS/FXO adapter. I've had the unit a few days and am exploring it's workings. I really want it mostly as a straightforward FXO adapter, to replace an X101p. Let me be clear, I'd love to support Digium in every way possibe, and will likely buy a TDM40 card shortly. But, the X101p has

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

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

Hi All, I am using a Sipura SPA-3000 as an FXO gateway to bring calls in and out of Asterisk. I am using "PSTN Ring Thru Line 1" (on the "PSTN Line" tab) so Asterisk answers the call rather than the SPA-3000. It is all working perfectly except I can't get the SPA-3000 to pass caller ID to Asterisk. It passes "Display Name", "User ID" and any "PSTN

Hi List Is there a cron that I con do to replace this, as the fx0 card doesnt hang up properly phonegc:/home/samantha# asterisk -r Asterisk CVS-05/30/03-17:17:07, Copyright (C) 1999-2001 Linux Support Services, Inc. Written by Mark Spencer <markster@linux-support.net> ========================================================================= Connected to Asterisk CVS-05 currently running on

Greetings, We are currently testing a Sipura SPA-3000 as a gateway from our Asterisk system to a PSTN line for 911 access. We have a number of locations and want to place an SPA-3000 in each, connected to a PSTN line that will provide the correct ANI/ALI information to 911 for each location. It all works great, except for a reasonably significant (4 seconds) delay between when the SPA-3000