similar to: SPA-841 spontaneous voicemail problem

We're investigating audio quality issues in our system; maybe someone can help. We're using Asterisk as a basic PBX, with a single PRI on one side and SIP phones on the other: Sipura SPA-841's. We're experiencing several audio effects which seem to commonly correspond to network failures (packet loss, high jitter, etc manifested as "robot voice", dropouts, periodic

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 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

I started out a few years ago with some SPA-841 sets, because the Grandstream 2000 I thought I wanted was perpetually delayed. The GS had more call appearances, and I didn't want just the 4 max that the SPA offered. As it turns out, with the greater flexibility of VOIP, I don't need 'dedicated' CAs the way I needed them on ISDN previously, so 4 is actually adequate. Along the line,

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

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?

Has anyone gotten call-waiting to function on the 4 line SPA-841? I've seen some documents that say it can do it, some say no way. If yes, can you share configs / SPA-841 settings? If no, did you work around it? I can call out just fine on all 4 lines, however, if I am on the line, another call coming in does not ring the 2nd line...it just goes to busy / VM. -Darren

I'm running the latest firmware on the SPA-841 and have a problem with echo. The echo occurs on all calls (PRI ISDN on a E110p or SIP) and is not present when I use the SNOM190 phones so I can def. isolate it down to the SPA-841s. The codec used is g711u and the phones are on their own dedicated 100mbit switch with no other traffic. The server is a 3Ghz PIV sitting at 99.9% idle all

Running * CVS with a SPA-841 (0.9.5) and can't seem to get the message waiting light to come on automatically. There is a control in the web interface to turn it on and off (seems rather curious to me but whatever). Has anybody got an idea as to where I may be going awry? TIA, Paul -- Paul A. Dugas Dugas Enterprises, LLC paul@dugasenterprises.com 1711 Indian

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

-----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

Am I doing something wrong here? GOt a SPC-841 the other day and have it registering properly. Can place and recive calls as expected but when on the phone, a second call is immediately dumped to "busy" voicemail. Does this thing not support call-waiting? Or, have I just got my configs wrong? Paul -- Paul A. Dugas Dugas Enterprises, LLC paul@dugasenterprises.com

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

Hi. I am thinking about building an asterisk system for a small business and want to be able to page through the phones. It seems like to do this asterisk needs auto-answer support in the phone. I know the SPA-841's support this, as do Cisco phones, but I have been unable to determine if the new Cisco/Linksys SPA-941's do. Does anyone here have experience with trying to use auto-answer

Hi list, I have a couple of questions related to asterisk billing and the generation of cdr logs. I've searched the wiki but have not found my answers, hopefully you guys can help. 1) When are asterisk CDR logs _normally_ generated? When the call arrives, when the call hangs up, or both? I have looked at the records created and it seems to only generate it at the time the call is

Hello, Today I restart the MySQL-DB (/sbin/service mysqld restart) and I could no longer connect to asterisk (/usr/sbin/asterisk -r) for a few seconds. There is now a core dump present in /tmp : -rw------- 1 root root 88M Aug 26 08:07 core.sip.pbx.tld-2011-08-26T08:07:35+0200 How can I get usefull information about what went wrong ? Because a spontaneous reboot of Asterisk has never

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, has anyone else also noticed spontaneous reboots ?! I noticed this today and also yesterday. Can't really see if there is a fixed time between the reboots. Normally al of my SIP peers are registered. When I put up the CLI today I saw that a lot of SIP accounts where UNREACHABLE and needed to register again (what they slowly did). These are realtime SIP peers that reside on

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