similar to: multiline IP hardphone w/ FDX speakerphone?

Folks! Netweb Group, Inc. fully supports connectivity to any Asterisk PBX systems you have and can provide A-Z termination with immediate effect. Any volume is good enough for us, even an amount as small as $1.00 a day will do for us. We will provide connectivity from our Softswitch IP 216.162.116.46. If anyone is interested, add this to your Asterisk IPBX and then email me for setting up a

I'm running asterisk stable 1.0.5 and I'm trying to get the netweb eezee phone version v1.37.008 to talk IAX to asterisk. The pages I saw in the wiki maybe didn't hold my hand quite enough and the information on the eezee phone website appears to be for a different firmware version. If anyone has done this recently and has a working situation I would appreciate some useful hints

Greetings, I just received some netweb-301 phones frm Seshu down in NJ. I cannot for the life of me get it to register with the asterisk server, nor upgrade the firmware to the latest (1.41) i'm still using 1.37. The packets are traversing the router, going into the other subnet, hitting the asterisk box, but not actually making it to asterisk. Nothing in the asterisk logs, but tcpdump

Hi, What is out there in terms of SIP enabled handsfree speakerphones? Looking for something that works well and also fits a low budget. I am used to using a Cisco 7940. It is a great phone but a bit expensive. Thought about the Polycom SoundPoint 300 until I realized that it does not include speakerphone functionality. Thanks, Adi

I'm using Asterisk 1.2.3 and PANASONIC KX-TS208W - Speakerphone does not work with it. It works fine when you pick up the handset. Anyone experinced this problem before, the speaker works fine with Verizon line. The phone is behind a Linsys router RT31P2. -------------- next part -------------- An HTML attachment was scrubbed... URL:

I am not familiar with that phone. Is it single pair? -----Original Message----- From: broadbandvoice@comcast.net [mailto:broadbandvoice@comcast.net] Sent: Sat 4/22/2006 12:13 PM To: asterisk-users@lists.digium.com Cc: Subject: [Asterisk-Users] PANASONIC KX-TS208W - Speakerphone IncompatibleWith Asterisk 1.2.3 I'm using Asterisk 1.2.3 and PANASONIC KX-TS208W - Speakerphone does not work

I'm looking for recommendations on speakerphones for a conference room. We're using Grandstream GXP-2000 which we've been very happy with on all accounts, except the speaker phone. Speaker phones on these units are extremely bad, picking up any and all background as well as having full-duplex issues (that is, when the other end is talking you can't talk over them to interrupt or

I expropriated the right to rip Daniel's disclamer for use in my email too.. DISCLAIMER: You might have to pay royalty fees to the G.729 patent holders for using their algorithm. For easier testing I prepared codec_g729.so binaries and associated libraries and put them on the web: http://kvin.lv/pub/Linux/Asterisk/ arkadi.

Idle curiosity: I like the look and feel of the Grandstreams, but it's been my experience that the speakerphones suck (esp. when compared to the pretty damn flawless Polycoms). I've used the BT-100/101 and GS-2000; have any of their newer models changed that? Thanks! -Ken -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean.

Anyone who has experience with the Polycom 330 know if the speakerphone is loud enough to be heard in a 20 foot x 20 foot room? The context is a classroom where announcements will need to be made. The phone will be wall mounted at the front of classroom. Thanks, Matthew Brothers

Thanks, I was afraid to try and change gains that I didn't know what they did simply because I don't want to blow a speaker or something.. :) I'll try it today. The only thing I haven't figured out is how to set a default ringer in the configuration file, set my time to EST w/ Daylight Savings and when receiving incoming calls if it's possible to see NAME & NUMBER instead

If that was possible, that would make my life easier as well :) -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Michael Graves Sent: Tuesday, August 31, 2004 10:12 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom SoundPoint... Gains - Whichis for speakerphone

John, By chance do you know how to set a default ringer? What I have done is the following: <DEFAULT se.rt.1.name="Default" se.rt.1.type="ring" se.rt.1.ringer="7" se.rt.1.callWait="6" se.rt.1.mod="1"/> As you can see, I want 7 to be the default ringer for line 1... For some reason, it doesn't take these changes.

Hello all, I'm looking to troubleshoot some echo issues and possibly "studdering" while using the speakerphone with the Intellitouch ITC3002. On the hardware end, i have a 2621 setup as the router with some policy maps for qos, and a 2900XL with cos priority set to 5. I have setup created 2 vlans currently (really small office), 1 for corp/data, and 2 voice vlan. Asterisk is on

now that I have finally figured out what I was doing wrong with my polycom phone and got it to read the configuration file Im changing some gains. I successfully changed the gain for the ringer... It was too low for me. Does anyone know which gain would be for the call waiting and which tone would be for the "hands free" mode? Thanks

Sorry, got the subject wrong in the last mail. My e-mail client is playing games with me...

Johan Nilsson a ?crit : > We are using Speex AEC and preprocessor to handle the acoustic echo. The > AEC typically contribute with ERLE of 15-20 dB which should be as > expected from the algorithm. Additionally we need about 30 dB echo > suppression which we hope to get from the preprocessor. So far it seems like you're doing things right. > However it has shown to be very

Johan Nilsson a ?crit : >> There's also a parameter to control the maximum amount of >> suppression allowed: SPEEX_PREPROCESS_SET_NOISE_SUPPRESS : noise >> suppression SPEEX_PREPROCESS_SET_ECHO_SUPPRESS : echo suppression >> when there is no local talk >> SPEEX_PREPROCESS_SET_ECHO_SUPPRESS_ACTIVE: echo suppression in >> double-talk > > Yes, I am

Johan Nilsson a ?crit : >> What happens if you make SPEEX_PREPROCESS_SET_ECHO_SUPPRESS_ACTIVE less >> aggressive. Does it end up with too much echo or it just doesn't realise >> that it's in double-talk conditions? > > My impression is that it does not make much difference on the timing to > set this parameter less aggressive. Depending on the how loud the

>> I think the residual echo estimation is fairly reliable but I do not know >> how to use this to improve Pframe and in that way solve our main problem >> with the gain during near end talk. > > How can you tell that the residual echo estimation is reliable? in any > case, I suspect that the whole Pframe idea might have to be revised > (i.e. computing it completely