similar to: OT recommend an IAX phone or IAX softphone+USB handset?

Ok, I am back from my thanksgiving holiday, and I find there was a big snow storm here in Seattle. Apparently during the storm there where multiple brown out/black outs. I have struggled since day one to get a high quality PSTN gateway configured with my very long loop and Mac based asterisk. I originally tried the HT-488, which had multiple issues, and was unacceptable. I then purchased

Hi, I am not sure if it's a suitable place to ask this question. Anyone here familiar with USB phone please? I want to buy a cheap one to make it work with IAX softphone. Regards. Steven louse _________________________________________________________________ Add photos to your messages with MSN 8. Get 2 months FREE*. http://join.msn.com/?page=features/featuredemail

I have a client that want to try the softphone with USB handsets route to see if hardphones will even be needed. I always push for hardphones (Polycom) so I am not sure about softphones or USB handsets. This is going to be for a 300+ seat call center onsite and many offsite, I plan on using OpenVPN for the offsite machines. Any advice on softphones, handsets, or practical experience with

Since upgrading to 7.2, I've noticed a random problem where I dial a number and hear all the correct tones in the handset, but the display won't show all the numbers I dialed. So you sit there waiting for the dialplan to kick the call off (b/c you heard the proper amount of tones played and think it's all good) but the phone is just sitting there b/c it somehow "missed"

I have problem to change from handsfree mode to handset mode. When I switch from handset to handsfree while waiting for connection I press the green speakerphone button once. It is all well. Once it is connected I don't want to give the called party too much echo and I want to switch it back to handset. If I press the green button again I lose the call. Anyone knows whether it is possible

One of my coworkers needs to get a softphone set up to my Asterisk system; he's a Linux user, so it looks like about the only IAX2 option is iaxComm. For ease of use (he'll be using this a fair bit), I'm recommending that he get a USB handset; I'm just having trouble finding any US retailers for them. :) Could someone recommend a USB handset that's compatible with iaxComm

Breeze to set up, too. To monitor and transfer to SIP/1000 / ext 1000: 1. Insert exten => 1000,hint,SIP/1000 into your default context (the context the extension is in) 2. In the monitoring phone's web interface, click Function Keys, pick a key, change it to Destination and type in SIP/1000. Once you submit the form it will change to a SIP URL, that's OK. 3. There is no step 3.

Hi All Has anyone come across a handset that can somehow replace FOP? Some users don't like FOP unless it is on a dedicated PC. Thanks Garth

Even better you can set your firefly softphone to auto answer so that you don't even need to be near the pc to answer. Cheers, Dean -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Linn Boyd Sent: Friday, March 04, 2005 2:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re:

hello all- has anyone had any problems with the handsets on BT100's. Just picked one up for my lab and the speakerphone works great but I am only getting one way audio (incoming) from the handset. Since the speakerphone works fine, I can't think of any config. reasons why the handset wouldn't other than a faulty handset. Any thoughts or experiences? Jason Kawakami Technical

Seeking recommendations on handsets for use with Asterisk. I've been looking at the Aastra 480i CT because of its cordless handset and also the new Linksys SPA-942. Anyone using either one of these with comments on them? Any other thoughts on good reasonably priced handsets? This is for just a couple of people who work from home offices and will be connecting to an Asterisk server hosted

While I'm asking about the Polycom ip500, the answers for all phones where mic/handset/headset levels are adjustable would be of interest to many I'm sure. For the ip500, the default value for the handset seems to be voice.gain.tx.analog.handset="3" I've noticed that echo all but goes away when one reduces the mic volume on almost any phone. My question is, for you users

After 4 hours of debugging codecs, changing config files, etc. as a result of not being able to capture voice from a Cisco 7960, I eventually found that the mic in the handset appears to be dead. Does anyone know where I can get a new handset (just the part you hold to your head, everything else on the phone works fine)? Or, does anyone know how to open one up? I tried doing a little prying

Is there a way to initiate a transfer using an analog handset? For instance I'm looking for a way to do something like the following: External call comes in and is answered by user A. After talking to the caller they determine that the caller really needs to speak to user B. Is there any way for user A to initiate a transfer to user B, using only their analog handset? Now to make

Does anyone know where I can get replacement handsets for the Polycom SoundPoint IP phones? Or does anyone have any they want to sell? From the looks of it you have to buy a whole new phone to get a new handset. My vendor, TriaTechCOA, told me I had to buy a whole new phone to get a handset, which is pretty ridiculous. Maybe there is a more sane vendor I should be buying from? Thanks, -Ryan

Hi All, I have a Polycom Handset on a front door and I'd like the phone to dial a number as soon as the handset is lifted without having to press and buttons or enter any numbers. I know how to do this on a Linksys but I can't find out how to do it on a Polycom. I would be greatly appreciate is some is able to tell me how this is accomplished. Regards David. -------------- next part

Hi Guys!! I wrote a little patch for asterisk 1.2.5 and I will maintain it for future release unless somebody explains me how we can ask people at Digium to add it to the source tree... We are planning on using Asterisk as our main PBX for the office over the next few weeks. Our current setup uses TDM400 cards to bring our 8 lines into Asterisk, our Telco provides us an option for three

Martin: I had seen your other post and sent you a message off-list, but I never got a response. What do you feel is the most lacking that does not make it ready for a production enviroment. - I've been using a SIP deskphone in my office and usually some sort of ATA at my house, both as the primary phone. I've also had mobile phones from almost every carrier. Each one of these devices

is there anyway to change Sip headers in local channels? if a user sets forward on their handset, calls coming in to the handset get diversion header added: Diversion: "202" <sip:202 at 192.168.1.46>;reason=deflection Then asterisk sends the call to local channel: - Now forwarding SIP/201-00000483 to 'Local/3333333333 at test' (thanks to SIP/202-00000484) and not all

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?