similar to: Sipura / Handytone / Cisco

I am trying to get the handytone 286 to make a very simple call to * and having problems. It registers with * just fine, but when I place a call (to echo test, for example), the RTP stream seems to have problems opening. Here is there error I get in *: WARNING[98311]: File chan_sip.c, Line 464 (retrans_pkt): Maximum retries exceeded on call 20d1c411-e210-5f3d-3f88-19035c8fcb26@192.168.2.6 for

I had a Grandstream 286 at my home hitting my Asterisk box at the office, all worked well and I received phone calls fine until the device just up and died. I replaced this unit with an SPA-2000 because I have been impressed with the Sipura devices and decided to use them for most of my needs in the future. Problem is that my phone attached to the device rings shortly after power up of the

Hello list, I have several SPA-2000's and 3000's scattered about the Internet (all behind NATs). Because I do not qualify as an ITSP, Sipura will not license their "Sipura Profile Compiler" so that I can have the units remote upgrade, remote re-configure, etc (via TFTP or HTTP). This is extremely annoying. Right now if I have to make a config change to any of these

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've just made an interesting observation that I'd like to share with you all: the popular Sipura SPA-2100 just doesn't seem to be as great as I'd hoped. I've been trying to get inbound AND outbound faxing working via Asterisk and at least one of my termination services: Voicepulse or Sixtel. In general, inbound has been working flawlessly but outbound has been pretty

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

Hi, I'm looking for a full list of xml provisioning variables of the SPA-2100/3000. Currently the Sipura website has example XMLs only for the SPA-841 [1] and SPA-941 [2]. I'm mostly interested in the CallerID type selector variables and whatever variables control the PSTN<->VoIP settings. Sipura Configuration website form field names are numeral only. :( [1]

I'm considering a situation where I buy about twenty ATA devices. I've played with the Linksys / Cisco PAP2T, and got that working fine with some inbound and outbound faxing. The web GUI was okay. I'm seeing prices around $45 to $50 for this thing. It comes with two FXS ports, but I only need one FXS. I've seen the Grandstream Handytone 286 online. It looks promising as an

I had a problem in the past with a SPA-3000 acting funny that Sipura helped me with by telling me how to factory reset it. They responded in less than a day to my email request and the unit has worked fine since. I've had similar turn around on requests related to a batch of SPA-841 phones. They were all handled by real people who appeared very knowledgeable on the products. This appears

Good evening. I just wanted to take a minute and review my experiences with some of the SIP devices out there on the market. I hope this post will help newbies or someone considering a certain device. I would appreciate any other input on either the devices I am "reviewing" or other devices that I didn't! These devices are deployed in our primary line and small PBX replacement

Hi, We have a page on the wiki just for phone reviews, but I think it needs a bit of format change. Instead of individual reviews for each phone, I think each person should review all phones they have worked with and list the phones they have had access to and rank them in relation to each other. Also each review should have a date so the reader can see how fresh the data is to current.

Hi List, I hope this setup must be done by our astersik users.. I am using Sipura 3000 to receive PSTN calls and forward those calls to asterisk for voice processing and after that, I am transferring call to extension through FXS port on SPA 3000. Currently, media of call is trombone through asterisk. i.e achieving blind transfers on asterisk with SPA 3000. Is it possible to stop trombone

Yes, After call transfer,I don't want to be media go through Asterisk. Is it possible ? Thanks, Karun. -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com]On Behalf Of Dameon D. Welch-Abernathy Sent: Thursday, August 12, 2004 12:07 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Blind Call Transfer using

On Tuesday 10 May 2005 9:45 am, David Masure wrote: > Hi folks ! > > I bought two sipura 841 phones. I used to have GN Netcom headset which > I connect instead of the handset. The problem is that I don't have any > sound coming out the headset and I can't speak neither ! > ... > > Or....can anyone advise me on headset working with the sipura 841 ? I just use a

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

Dear list, I am playing with Sipura 3000 since last week. Through the wiki pages I could get running it reasonably well. My setup is that of a Sipura, linked with a local analog cordless phone, a local PSTN line and the setup to link to an asterisk server located at a remote static ip address. I can dial the cordless phone from other extensions located at the asterisk server; I can dial out

Sorry to bug all of you with this, but I have to assume there are a number of Sipura experts here... I have a Sipura SPA 2000 that I've been using for nearly 2 years now. It's flashed up to firmware 3.1.5. On line 1, I no longer get Caller ID (it used to work, and I can't remember when it stopped). On line 2, I always get Caller ID. To my old eyes, _every_ switch on both lines

Hi all, Anyone able to remotely reboot their password protected Sipura SPA-3000 from command line. I am trying: Sipura SPA-3000 from command line: # wget http://admin:mypassword@192.168.1.55/admin/reboot The strange thing is it works fine when I go to http://admin:mypassword@192.168.1.55/admin/reboot with my web browser... Thanks....

Problem 1 - Outgoing: I am able to call out of the * box using the analog line attached to the sipura 3000 but when the person being called answers there is no audio from either end. * registers that the call was answered but passes no audio. Problem 2 - Incoming: When calling into the 3000 attached to * it never seems to pickup the line. The phones don't ring on the asterisk side. I used

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