similar to: SPA841 Utterly Horrible, are there any good stun hardphones?

Hello All, Has anyone configured bri to answer for only one msn? In essence, when the primary is busy I want to have channel 2 ring. I am using an eicon diva server bri I know I saw it in the windows interface, but don't see it in the linux setup. Regards, Greg

Hello All, I recently put in a zaptel 1fxo/1fxs card. I am experiencing heavy static. Even with the pots line disconnected, if I do a dial I still get static. This way I know it's not the line, but rather something on the card. I tried alternate pci slots. This card has a power connector, does anyone know what the power requirements are? The unit is in a small case with a 2.4ghz p-4 and

Repost, first one never made it to the list... -----Original Message----- From: Gregory Wiktor - ADCom Corp. Sent: Thursday, September 29, 2005 3:20 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Voice Prompts, what do you think? Good voice. Should we record a new prompt-set? Hello all, I have someone working for me who has a nice phone voice. I looked at

Hello All, I am looking for a sip phone that is capable of automatic nat. The Cisco ata186 for example works fine for natting with iconnecthere, but as for asterisk, both my 7960 and polycom ip600 require you to set the nat ip on the tftp. Does anyone know a good phone (or ata) that can do this automatically? For example, I want to give a phone to my brother, who is going to europe. His ICH

Hello There's a lot of information on VoIP at www.voip-info.org ... but there's also a lot of outdated information there as well :-/ Since SIP is a pain to use when NAT is involved, especially when both the Asterisk server and the remote phones are behind NAT... I'd like to try IAX to see how it works and if it solves the issue. I'd like to start with a softphone (Windows

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 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

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

Hi, I have configured a certified asterisk 13 server with chan_mobile and res_pjsip. I have a Cisco 7940 hardphone and I use ekiga as softphone client. Now the problem is, using the hardphone I'm able to call the softphone and hear everything properly. But when I call from the hardphone to some number that has to be dialed via chan_mobile, I'm not able to hear what the other side says (I

Hi, I've been asked sometimes to tailor call history features embeded in SIP hardphones. For example, a cutomer wanted internal call to be taken out. Another wanted calls to sorted according specific criteria. 1. Have you identified a phone offering the possibility to display as Call History, an XML list produced on a distant web server ? With this feature, you would simply have to tell the

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.

I've a problem. I've some spa3000 and spa2100. Asterisk 1.2.4. Prefered codec g711u in both. Calleng from a fxs of spa2100 to the fxo of spa3000, all works ok. Then I call from a sip phone configured for using g729, to the fxo of spa3000, it also works ok. The problem is that after this, when, making again a new call from spa2100 to spa3000, spa2100 receives only white noise. I suspect a

Hi, Is anyone aware of a wired sip hardphone supporting 802.1x authentication ? I've been told some Avaya and Alcatel ip phones supported 802.1x. As 802.1x is widely used with wireless hardphones, I'm wondering whether or not, 802.1x could also be valuable for wired environments. Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi list. I want to know if there is any console o secretarial hardphone that works with asterisks. I mean a phone in witch i can see the state of the extensions, the phone lineas, etc. Can hold o transfer easly a call, etc. Thanks Alvaro Parres

Hi SIP is such a pain to use when NAT is involved that I'm willing to buy an IAX hardphone for someone who works remotely over the Net and needs to get calls from our Asterisk server, itself behind a NAT. Are there good, affordable IAX phones you would recommend? Thank you.

Hi, When a SIP hardphone is transfering a call while ringing (caller and callee don't speak to each other) using phone's Transfer key, it seems BLINDTRANSFER remains empty. Though I can see a 302 MOVED TEMPORARILY message coming in. Is there a work around or something obvious I'm missing (it's the first time I'm playing with Dialplan transfert features. context mylocal {

In trying to get my services up and running, I've encountered the usual spate of first-time issues. I was wondering if there was a good FAQ or Howto on troubleshooting NAT issues. The equipment that I'm using is typically a Sipura ATA or hardphone (SPA-942) sitting behind either a Linux box as firewall (using IPtables and Arno's firewall) or else a Cisco IOS device running 12.4T

What hardphones support xml/html/xhmtl/microbrowser? I need an inexpensive SIP hardphone that can run simple applications (queue status, etc). The phones I know of: Aastra 480i, 9112i, 9133i (though limited by 3 LCD lines on the 91xx seems kind of silly) Cisco 79xx Mitel 5235 Polycom IP601 Any others? -Dan

Sorry if you see this message repeated twice. I would like to set up hard phones and softphones with the same extension so that when anybody in the company dials an extension, each user's hardphone and softphone will ring at the same time. I've tried setting this up before, but I noticed that the last sip device to register with the same extension is the only one that rings when the

I'm having issues with echo. My setup is Polycom IP501 phones connected to an Adit 600 via T100P. 13 Lines going to the Adit. All echo so far is on the local side (Employees hears own voice, but only on some calls). Watching the channel with ztmonitor, I notice that TXGAIN is pegged out most of the time. RXGAIN is anywhere between 45 and 60%. Is it safe to assume that playing around