similar to: Software to generate an SRTP key pair?

At 1:07 PM -0700 on 4/5/04, Scott Laird wrote: >[snip] >2. Is anyone working on SRTP for Asterisk? Are there any SRTP >clients out there? > >Scott Yes to both. http://www.voxilla.com/modules.php?op=modload&name=News&file=article&sid=56 http://www.dslreports.com/forum/remark,9818470~mode=flat Read carefully. Sipura is offering crypto in their newest (yet to be

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

was breezing over http://voxilla.com/ Looks like a new ATA from the founder of Komodo Technology (aka the Cisco 186) Sipura SPA 2000 http://www.sipura.com/products/spa2000.htm to join the others Cisco ATA 186/188 http://www.cisco.com/warp/public/cc/pd/as/180/186/ 8x8 DTA-310 http://www.8x8.com/products/home_office/dta-310/index.asp.html Grandstream HandyTone 286

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

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

Problem: SPA3000 phone does not ring for incoming PSTN call although I can dial out. I set up my Sipura with the Voxilla Wizard which is pretty good but leaves out some important details. The Voxilla Wizard for Supura SPA3000 gave me a setting for PSTN Tab -> Dial Plans -> Dial Plan 8 (<S0:66610>) Should I put extension [66610] in sip.conf with a context in extensions.conf that

Can you folks recommend Asterisk merchandise resellers that have good reputation? So far I didn't have much luck except surfcity.com where I've ordered their P104 phone. I would like to purchase Sipura SPA-2100 and SPA-841 and "VOIP Telephony with Asterisk" book. So far I have poor experience with Voxilla and Asterisk Mall. - Voxilla folks have no clue how to ship Warranty

I was having problems getting a Linksys PAP2T-NA to work with Pitney Bowes mailing station so it could use its modem to dial home and download postage/software updates. After scowering the web, I couldn't seem to find a definite how to article on what settings were needed. I finally came up some settings by combining the information from various places around the 'net. I have typed out

I?ve been tearing my hair out trying to get a Polycom phone (behind a NAT) to register to an * box behind a Cisco SIP ALG. With known good credentials configured on the phone and in *, I get 403 Bad Auth when trying to register. If I put the phone onto the same LAN as * it works fine without changing any authentication parameters whatsoever. If I make the secret blank (null) on the phone and *,

I have a SPA-3000 that is failing to hanging up pretty often; almost every day now. The weird thing is that an almost identically configured (same FW, different HW rev) second unit right next to it isn't having the same problem. Swap the lines and the problem stays with the unit. I've been going round and round with Sipura and their latest message told me to use a procedure detailed in

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

Citeren Emilien Kia <emilienkia-guest op alioth.debian.org>: > Modified: branches/ssl-nss-port/server/conf.c > ============================================================================== > --- branches/ssl-nss-port/server/conf.c Wed Jan 5 21:12:03 2011 (r2803) > +++ branches/ssl-nss-port/server/conf.c Thu Jan 6 10:27:55 2011 (r2804) > @@ -178,6 +178,22 @@ > return

Hello all, So i am new to asterisk and very green when it comes to Linux, so don't beat on me too bad :) I just set up * on Red Hat 9.0 last night... everything "seems" to be configured coffectly, I can start * no problem and get the CLI prompt... now here is my question... I have an account set up with VoicePulse Connect! and also have a Sipura Spa-2000... i am trying to get make

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

We often hear about the problems with vendors, but not as often when a vendor does more than expected. Last night I placed an order for a Sipura ATA with http://www.voxilla.com/ Today they called me to explain that they had just run out of inventory, and would be happy to cancel/refund my order, or give me a free upgrade to next day shipping when they receive more inventory next week. This

Anyone tried the new PSTN/FXO port in the new SPA 3102 FXO/FXS adapter ? From a quick test (got mine yesterday), seems like it is not recognizing Caller ID from PSTN/FXO port.. Using the same configuration as a Sipura 3000 (to be sent to mother-in-law POP :-), no Caller ID at all, (I've even extended the PSTN delay to give it more time, but no dice). www.voxilla.com forum has a couple

Hello all, We have an Asterisk server that has worked without issue for a while. Before, only Sipura and Polycom 500 series phones were used. Recently, we've added a few POE switches and 20 or so Polycom 330's. The 330's seem to lock up often. One easy way to do this is by hitting the * key. I looked thru the XML files for dialplan and what not, nothing stood out.

Anybody using a Sipura 3000 for FXO with Asterisk? Mine is working except for one small nit... When a call comes in from the PSTN, the Sipura answers it and then passes it on to Asterisk, which plays extension ring tone. I'd prefer for the POTS line to stay on-hook while the extension rings, and to only be answered by the Sipura when the extension answers. Has anybody made this work?

Does anybody out there have any evidence that Sipura support is still in operation? rgds benjk -- Sunrise Telephone Systems, 9F Shibuya Daikyo Bldg., 1-13-5 Shibuya, Tokyo, Japan. NB: Spam filters in place. Messages unrelated to the * mailing lists may get trashed.

We have extra equipment that was over-ordered or unused. All of the equipment is brand new. The equipment has been highly discounted to move quickly - the last set of equipment sold in 48 hours. If this equipment is of interest to you, call or e-mail quickly. Buy on VOXILLA and SAVE $300 each (Cisco routers & switches): http://store.voxilla.com/customer/home.php?cat=259 For Sale (all new):