search for: spa841

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

I don't know what's changed, but four SPA841s and a SPA3000 are no longer answering when they get an inbound call from *. This has been a working configuration for weeks. I *have* been fiddling with the server config; however, the configuration is under version control and I've reverted everything to exactly how it was when the server w...

Hello all, I had been using a spa841, and all the sudden it's dead. When you power up, all the lights come on and nothing else. Has anyone experienced this? Thanks, Greg

Guys. I just unpacked on of the new spa841 I orderd and I was changing the ringtone (and listening to the options) when suddently the phone stopped playing back the tones and now the phone doesn't ring, speaker doesn't work and no ringtone play can be heard. Has anybody had this kind of problems?

Hello. I have a customer that needs to downgrade the firmware on their SPA841 to 3.1.1(a). I can't seem to find the firmware file. Google turned up 3.1.2-something and Linksys is taking a while to get back to me. Anyone happen to have that file lying around? Thanks, Nabeel

...o anybody who happens to be within earshot. Any clues where to look at what's going on? My voice mail number (extension 220 in my dialplan) is the only number being dialed. When this happens, show channels looks like this: IAX2/NuFone-1 (None) Up Bridged Call(SIP/spa841-09f083 SIP/spa841-09f08388 220 at inside:5 Up Dial(IAX2/mumble:mumble which looks the same as if I dial it myself. Thanks in advance, ------------------------------------------------------------------------ Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST...

Hello All, I have been using the following phones with excellent success inside my lan: Cisco 7960 Polycom IP600 I have also been comfortable with the SPA-3000. I recently got a SPA-841 and the quality is aweful. Even in stock setup, it's like when I speak into the handset I sound like I am very far away. I want to get a hardphone for europe, but I need to be able to use a STUN server, and

Guess who's here to do an Asterisk demo this week without the power supply for his SPA-841. I have an ATA186 with me. Both phones use a 5v supply. Does anyone know whether the supplies are interchangeable? Thanks in advance; sorry for the noise. B.

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 properly. Several reboots and factory default reset later, it sometimes works. Has...

I'm setting up a new AAH 1.0 box to replace my AAH 0.6 box. Though on the new box, I've installed a generic ebay X100P. I don't have my livevoip or voicepulse accounts set up yet on the new box (can both boxes be registered at the same time?). I've set up one IP phone (SPA841) with the new box. I have my SBC POTS line plugged into the fxo card. I set up everything in AMP. I can make out going calls. The problem I'm having now is the digital receptionist greeting (aa_1). If I set it to automatically forward to an extension it works. But, if I have it play a...

Does anybody have a good link how to connect Sipura with Asteriks, both behind NAT? I'm using FWD but their connection is like a weather (especially IAX), I need something more reliable. I was thinking of using stun and/or proxy but can not find any good link explaining how to setup Linux server -- #Joseph

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

...record the call but have had absolutely no success. Is there a trick to this? In extensions.conf [globals] DYNAMIC_FEATURES=>automon [default] exten => 123,2,Dial(SIP/3000,,wW) ; wW allow one-touch recording During the call, I press *1 but it records nothing. My phones are all Sipura SPA841 SIP phones and I am running the latest * build. David Morrow Technical Systems Lead Autodata Solutions Company David.Morrow@Autodata.net http://www.autodatasolutions.com <http://www.autodatasolutions.com/> Tel: (519) 963-3020 Fax: (519) 451-6615 < Lead, follow or get out of the w...

Hi All; Any one can suggest a TFTP server to be installed in Fedora (same machine that Asterisk is installed) to be used for Cisco IP Phones to download the required firmware and configuration files. Thanks for the help in advance. Regards Bilal

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

via google, I found the reference regarding digit maps for the Polycom phones: http://lists.digium.com/pipermail/asterisk-users/2005-January/082884.html But I don't see any instruction for prepending digits to the number dialed. Does anyone know how to prepend a digit to the number dialed (from the Polycom side, not Asterisk)? I can do this pretty easily on a Sipura. i.e. Say I want to

Hi again everyone ! Was wondering if anyone had any pointers on how to debug voice quality issues in asterisk. I've got a user who either can't be heard on her phone calls (outgoing and incoming) and today someone that called her said that her voice was coming in and out. Any pointers or suggestions are appreciated! Thanks so much again ! This list has been so helpful to me.

I have a SPA-2001 and I didn't realize I could use calling features on an analog handset. Does that mean you can dial *77 and use a VOIP feature? (like forward or hold)? Mike ________________________________ From: Jorge Carrasquillo [mailto:jorge.carrasquillo@gmail.com] Sent: Thursday, June 23, 2005 12:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re:

...: No DTMFmode : rfc2833 LastMsg : 0 ToHost : Addr->IP : 192.168.10.32 Port 5060 Defaddr->IP : 0.0.0.0 Port 5060 Def. Username: 2608 SIP Options : (none) Codecs : 0x4 (ulaw) Codec Order : (ulaw) Status : OK (16 ms) Useragent : Sipura/SPA841-0.9.1 Reg. Contact : sip:2608@192.168.10.32:5060 Sip config for Box B on box A: pbx3*CLI> sip show peer pbx1 * Name : boxb Secret : <Not set> MD5Secret : <Not set> Conte...

Hi.. I am about to replace my aging Nortel Venture system with an Asterisk system and 6 Polycom IP 501 phones, and a couple sipura 841's for less used areas. We have 3 phone lines here. One is SBC, one Vonage, and one Voipjet... One hangup is that I can't figure out how to pass through a voicemail waiting indication from SBC. This is important because my wife and her family all