similar to: Coming-off-hold delay/silence on Sipura 841 and Asterisk

Hello, How good is :SPA 841 form SIPURA. Thanks Varun

I -think- that the Sipura 841 was PoE... and I'd been anxious to find out. However, according to Atacomm.com, it's been delayed until mid-January. *sigh* So: does anyone know of a (decent) phone that meets the following criteria, and isn't too expensive? - SIP - two (or more) lines - some form of TCP/IP-based configuration - 802.3af (power-over-ethernet) - 100 Mbit passthrough (not

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 ! I'am located in France and I was wondering if the cabling in the sipura and in the headset is the same (I mean the order of the cables) or maybe is there something else to

I want to implement a one way announcement and paging facility using Asterisk and Sipura phones. The wiki says Sipura phones only support Auto Answer using the Call-Info header which is no lone shipped with asterisk stable since 1.0.4. I would like to ask if anyone has implemented a similiar facility using Sipura SPA-841 or any other SIP phones. If I could take a look at how

Hi there, I have bought 70 sipura 841 phones for a customer of mine. When following the mass provisioning guide in the admin manual for the sipura, I see it download the spa841.cfg file from my tftp server Sometimes the phone also downloads is phone specific file via tftp, and it works okay then. But, after a reboot of the phone, it is very very likely that it won't startup

Froogle found me one supplier for the SPA-841, not sure I trust them though. Does this phone even exist yet? Does anyone have any experience with it? Does anyone know a vendor other than Atacomm/voipsupply?

I am looking at some lower cost phone to use with Asterisk. What is the ArtDio IPF-2000 or the Sipura SPA-841 like? Also, I see voipsupply.com has an ArtDio IPF-1000 listed, is this a new or an old model? I cannot find any information on it. Adi

I have just bought several Sipura SPA-841 SIP phones, and after some testing I have found out that the volume received by other parties when calling using the handset is very low. I've been able to reproduce this problem in the 3 phones I've tested so far. I've tried tweaking several configuration options but nothing I has helped so far. Has anybody else experienced this problem?

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

Hi Just 2 issues I have with SPA841. 1. I autodial extension 600 then inside an AGI wait for more digits. The digits are transmitted correctly to * but they do not show up on the SPA841 display, only the 600. How do I set the 841 is show the digits after the 600# 2. Is the SPA841 pixel display backlit? Master

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Well, a bit of a newbie using Asterisk and trying to get some Sipura SPA-841 phones to talk to the Asterisk server. Not doing something right apparently.... If I turn on debug for the extension in the CLI (sip debug ip xxx.xxx.xxx.xxx) I see frequent 5 packet attempts by the server to contact the phone, but seems to always be failing. The status

I was browsing through the web config of a Sipura SPA-841 (Firmware 2.0.13) and noticed a setting marked 'paging' under supplementary services on the Phone settings page on the advanced admin login. Anyone know how it might be used? Could it be like the Snom - exten => 10,1,SetVar(VXML_URL=intercom=true) exten => 10,2,Dial(SIP/testuser) Craig

> Hi, I'm a user of Asterisk@Home for a couple of months now and been playing around with it for a while. I'm facing a strange situation which i am not able to solve. I have my * server and a SIPURA 841 phone both behind my router at home (No NAT between them). My * server is registered to 192.168.2.XXX and my phone is at 192.168.2.XXX with a port 5060. Initially i see the registration

Well I just need to say I got my phone last week. Here is my quick review of the phone and hope that someone has a possible fix for it or I will be sending it back. First the phone is nice looking in my view and it's heavy so it feels like a real desk phone. But it has these stick, gummy or I really don't know how to describe the bottoms on the phone. There good size but when you press

Hei! Does anyone know how to configure this phone to autodial the number after interdigit timeout has passed? Rennes

Hi there, I have setup Asterisk with a couple of Sipura SPA-841's and Grandstream ATA's. The problem is that with both of these devices the Unattended call transfer process seems to be just like Attended but instead you hang up as soon as you have dialled the number of the party your are transferring to. The call transfer all works fine BUT as you complete your side of the transfer

Is there a a consensus on which of these is the better phone.I've personally been using an 841 and have learned to live with its shortcomings. I now need to recommend some phones for some sites we're installing. I'm looking at the BT102 for desktops that don't want/need a headset but need a phone for the higher end users (without costing the earth). TIA, tony Zero Effort

Anyone having problems with registration to * from a SPA-841? I got a SPA-841 a week ago. I noticed that sometime it could not be reached (dialed to) and it can't dial. In this case the line LED is yellow. I enabled logging to syslog and there is a hint as to what happens. For some reason sometimes it gets "401 Unauthorized" Any ideas what is happening and how to fix it? Phone

Hi all, The problem is on the volume of the voice sent by the SPA-841. I think the echo cancel algorithm sets a limit to the microphone when detects sounds or noise from the earphone. This problem generates an oscillation on the voice volume sent by the phone and even turns it off completely for very little lapses of time making the communication very uncomfortable. I manage three different

Hello List, I've spent the last day trying to find information on how to call multiple sip phones and have them all answer so I page everbody. When I use Dial( ext&ext&ext... ) the first phone that answers gets the page, but none of the others do. Is there a way to get around this? TIA, Steve