similar to: Sipura phone does not ring

I have a sip.conf with stanzas for sip phones that have 'context=sip-incoming for some Grandstream phones and another stanza for a Sipura SPA3000 with context=pstn-incoming. Reviewing the code today, I was dismayed to see that all my outgoing extens were mixed into those two. I have been told this is very insecure. How can I separate the outgoing extens? When I create a context

It _appears_ that the only way to create valid [context] is by a context = line in sip.conf. Is there another way to create a [new_context] in extensions.conf so I can dial from it? Right now most of my extens are in [default] and I'd like to avoid that. Larry -- Larry Alkoff N2LA - Austin TX Using Thunderbird on Slackware Linux

Some people on this group may have understood from messages posted here that the Sipura SPA-2000 is not currently available for shipping. That is not the case. Voxilla.com has the Sipura SPA-2000 available for immediate shipping, and has had them since late November. The price is $109.95, and it comes with a month of free VoicePulse service with activation fees waived (a $65 value). In return

I'm new to the list and to Wine. I loaded wine wine-0.9.7-i486-S10.2 into Slackware 10.1 in order to run the Forte Agent newsreader. It was setup and running under Win4lin which I have discontinued. What would be the syntax to load Forte Agent and specify the old "start-in" directory used under Win4lin? wine ./agent something_for_start-in_directory ?? Does Wine have the

Hi, all I am trying to figure out how to forward incoming PSTN call on SPA3000 to VoIP extension(s). Basically, I have converted my home to VoIP. I have normal phone (connected to SPA3000) and couple of IP phones. All call coming from VoIP DID do ring all phones (analogue via SPA3000 and IP ones). Now I need to do same thing for incoming PSTN calls. I have enabled gateway function in SPA3000 and

Does Wine have the ability to run DOS .com files? Specifically I wish to run 4dos.com which is a command.com replacement. Larry -- Larry Alkoff N2LA - Austin TX Using Thunderbird on Slackware Linux

I have a Slackware Linux box called Tillie with three shares. All at /, user at /home/user and public at /public which is "guest ok" for access without a password. The problem is that Tillie shows up in Windows Network Neighborhood as a single icon. To open that icon and see the three shares Windows demands a password, thus making it impossible to access. Is there any way to get

I have a working Asterisk 1.2 and Sipura SPA3000 combo but I would like to have more control over incoming PSTN calls to the Sipura. Right now such calls come in over the fxo port (sipurafxo1 on the PSTN tab) and are routed by dial plan 8 to the fxs port (sipurafxs1 Line 1 tab) I'm not even sure if the calls are getting to the fxs port and they sure are not going to the specified context.

Voxilla news story: http://voxilla.com/voxstory84-nested-order0-threshold0.html Two new products * A Sipura 2000 in a linksys box: Linksys PAP2 Phone Adapter * A combination NAT router with 2 FXS ports: Linksys RT31P2 Broadband Router Jim James H. Thompson jht@lava.net

Anyone have one of these yet? http://www.voxilla.com/shop/index.php?action=item&id=38 -Dan

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

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

Hi all, I was wondering whether anybody here would help me clarify this minor issue please. If I have the following setup; Asterisk ------ Sipura SPA 3000 (fxo) --------- Pstn Line Would a call coming in on the pstn line be answered by the ATA or just get passed through to the * server (depending on dialplan) to handle? So basically, the caller does

I'm trying to get working a spa3000 with asterisk. My problem is I cant get wroking the incomming calls (I installed the lastest firmware). My problem is asterisk is rejecting the authentication from the spa3000. Asterisk answers forbidden (SIP/2.0 403 Forbidden) and I think I placed the username and password correctly... Sip.conf says this: [linea2] username=linea2 type=peer secret=1111

I have downloaded the source (wine-20011004), compiled and installed with ~/wine-20011004/tools/install.sh and am attempting to start running it. My first try, as suggested in the docs, is /windows/sol.exe from a terminal in KDE and the error message I get is: Gtk-WARNING **: cannot open display: I compiled and installed as root - that may have been a mistake. However, I have a /root/.wine

I have been trying to read everything I can find on Sipura 2000 and Asterisk. I am trying to make the Sipura-2000 act as two analog lines off my asterisk. I have followed (what I believe) the example on http://www.voxilla.com/Article39.phtml and I still can't get my Sipura to register with my Asterisk server. I can re-config my Sipura to talk to fwd, or voice-pulse connect and it works

Hallo! I had ordered a Sipura SPA-3000 in the UK, but the supplier turned out to be unreliable and never shipped. Yesterday I went looking for alternative suppliers and found the Linksys SPA3000 device. It's a different box, but the specs look very similar. Is this the same device? Has anyone used this Linksys SPA3000 successfully with Asterisk? Thanks, Frank

I am trying to setup a simple Samba server using the instructions in the O'Reilly book "Using Samba" first edition on a Slackware 9.1 system which uses Samba 2.2.8a. The windows machine uses windows 98SE. The instructions under Server Configuration starting page 93 and Disk Share Configuration starting page 96 had me setup a very simple smb.conf which should then appear in Network

I would like to show a remark that would show call progress and appear on the CLI screen. The remark should be in the code of a sip [channel] or extentions [context] If I can't send my own remark, what little used 'show' command could I insert in the code? Can this be done? -- Larry Alkoff N2LA - Austin TX Using Thunderbird on Linux

I'm trying to update a bunch of downloaded files from a mounted older disk. This command works from the Kterm command line or in a script: rsync -nuv /mnt/kinda2/home/lba/dl/* /home/lba/dl and a list of the files to be updated appears on the screen. But this line (without a tailing * in source) does not work: rsync -nuv /mnt/kinda2/home/lba/dl/ /home/lba/dl and I get the