similar to: Info about F1000G

Displaying 20 results from an estimated 1000 matches similar to: "Info about F1000G"

2005 Jun 24
4
UTStarcom F1000 WiFi IP Phone Review
I bought a UTStarcom F1000 WiFi IP Phone from http://www.luxoncomm.com and tested it with Asterisk. This is a my first impression of the device. The F1000 supports SIP. It looks and operates like a cell phone, and connects to the Internet through WiFi, so you can use it at any WiFi hotspot. I set up a 802.11b wi-fi network with a Linksys BEFW11S4 Wireless-B broadband router with no security
2004 Feb 15
8
Wifi Phones
Hello list, I was going to buy this weekend a Wisip from http://www.pulverinnovations.com/, but jeff got out of stock and he wont have Wisip for the next 3 to 4 weeks. So I start searching for other wifi phones because I was really upset about it and I found IPC5000 from http://www.fahdtel.com/sip_phones.htm, I liked so much the pic that I email the guy and he send me the PDF with all the details
2007 Feb 19
2
UTStarcom F1000 - WLAN connection unreliable
Hi list, I bought two UTStarcom F1000 phones, pre-equipped with the latest firmware, including WPA support. Those are configured to register to an asterisk server on the internet (not LAN), and registration works. Calling and being called also, with transfer and all bells and whistles. After a few minutes up to 5 hours (varies widely), the display tells me that an Accesspoint is not available
2006 Mar 21
3
WiFi phones and WDS (Wireless Distribution System)
I'm about to start working with WiFi phones on my Asterisk installations. Can anyone tell me if they are using WiFi phones on wireless network that is extended with WDS and how well the phone handles jumping from access point to access point while on a call? Do any WiFi phones support WPA encryption or are they all still under the impression they are only being used on public hot spots?
2006 May 09
2
Configuring utstarcom1000 on asterisk
Can anybody help me on configuring utstarcom F1000 on asterisk? Is there a way to do it or is it impossible? ___________________________________________________________________________ Este mensaje se dirige exclusivamente a su destinatario y puede contener información privilegiada o confidencial. Si no es vd. el destinatario indicado, queda notificado de que la utilización, divulgación y/o
2005 Jul 07
4
Sipura SPA-841 Volume Oscillation Problem
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
2012 May 07
6
using Wifi smartphones as SIP clients
All, has anyone any experience in using Wifi smartphones as SIP clients? Does this work properly? What models/brands are optimal for this (in terms of ease of use, battery life etc)? Thx!! B.
2006 Jan 11
1
[suse-isdn] Major Problems UTStarcom F1000 registering -- pls help
Hi there, I am trying desperatly to register my WiFi Phone UTStarcomm F1000 with my asterisk server. I already changed the name of the user to "anonymous" since it looks like the phone sends that name. The WiFi Phone's IP is 192.168.1.217, the asterisk server's IP is 192.168.1.200 What is it that I am missing? Any help very much appreciated!!! The error message I get is: Jan
2004 May 28
11
Asterisk Database
I'd like to be able to add additional fields to the the Asterisk database. I'm using Mysql for most of my data lookup and manipulation, and it seems to work pretty well. In keeping with what I know how to do, it would be very handy to be able to insert say a "call forward number" into a customer record. That way, I could automatically route calls to extensions to a forwarded
2003 Nov 01
13
Quick Question
Apologies if there is a cleanly written and searchable FAQ that I could be directed to. I have no problem to RTFM if I can find the FM... Does Asterisk currently operate under RH9? I have IBM Netfinity 4000R servers that do not support X windows under RH8.x and I prefer not to go back to RH7.3... BTW, where would I find a useful FM? David -- David J. Sussman, MBA email:
2007 Feb 16
0
F1000 web configure
I have 8 - F1000G utstar phones. on a couple of them I can configure them by the WEB interface with no problem. On a couple of the them I cannot. I get no response when I point the browser to them. THe units work. keypad is fine. I can examine the network values - specifically the DHCP address. etc... Just cant get the web interface to do anything. Has anyone had this issue and know how to
2006 Mar 01
9
MOH native files
Where can I find alaw, ulaw, gsm, g729 formats for native music on hold? I have some mp3 files and I have tried to transcode them to above, but it seams that SOX can't do that. Please, tell me where to download some MOH files (in above formats) or how to transcode mp3? Thank you for your time! -- Tomislav Parcina tparcina#lama.hr
2006 Mar 15
6
Cisco phones and Linksys SRW224P
I'm having problem powering Cisco phone's (7940 and 7905) on Linksys SRW224P switch (with PoE functionality). I have tested three phone's, one is working (7905) and two aren't (7905 and 7940). I have plugged all three phones on same switch port with same cable! Do I need to change anything in phone configuration? Is there something wrong with Linksys switch? How can I troubleshoot
2006 Oct 27
4
IAX2 show peers - description
Hi people, pls does anybody know what "(T)" and "(D)" letter means? server3*CLI> iax2 show peers Name/Username Host Mask Port Status SERVER1 xxx.xxx.xxx.xxx (D) 255.255.255.255 9785 (T) OK (29 ms) SERVER2 xxx.xxx.xxx.xxx (D) 255.255.255.255 4569 OK (95 ms) 2 iax2 peers [2 online, 0 offline, 0
2003 Jun 16
8
SIP REGISTER
Hi! I have a new problem with my SIP device.I have done some changes and now I receive continuosly a SIP message: "501" "Not impelmented" back from the SIP Gateway. I can see that it doesn't register to Asterisk. I have in the SIP device: Registrar 1: UnRegistered to: 2222 registrar: 188.208.12.237 5060 expires: 2000 name: gateway passwd: 123 My
2006 Nov 15
2
T38 problem
I have problem with fax machine Panasonic DX600. It's connected to Grandstream Handy Tone 386 which is connected to Asterisk. Asterisk is connected to my SIP provider. To some numbers I can't send FAX, and I get following error on CLI. WARNING[2237] chan_sip.c: Unknown SDP media type in offer: image 31358 udptl t38 I believe that Panasonic DX600 machine supports T38. And when I have
2006 Feb 13
4
Voicemail - direct call
Hi list! How to send a call directly to voicemail recording? When I put this exten => 313,n,VoiceMail,u221 Or this exten => 313,n,VoiceMail,b221 In my dial plan, calling person first hears greeting message (busy or unviable). I would like to avoid greeting message (I would play something with Playback application). Is it possible? -- Tomislav Parcina tparcina#lama.hr
2006 Oct 18
2
Digium on Dell PowerEdge 1850
Does anybody have Digium TE212P interface card on Dell PowerEdge 1850? I'm planning to install * on that configuration so I'm looking for any positive/negative experience. Best regards, -- Tomislav Par?ina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)270248 Mob.: +385(91)1212148 SIP: tomo@sip.lama.hr e-mail: tparcina#lama.hr http://www.lama.hr
2006 Feb 09
5
What ATA should I buy?
I have running * without any Digium (or any other) hardware. Now I need to connect analog FAX machine to it. I think that cheapest and easiest way is to buy ATA. Please correct me if I'm wrong. Now, which ATA should I buy? Local dealer sells those four. I can buy something else (if there is any reason for it), but I prefer something of this. One more question, can I plug two lines in any of
2006 Feb 14
5
Call centre - * hang's up
When agent tries to transfer a phone call (*2 - att transfer) he hangs up. Why? When a phone call isn't from queue then att transfer works fine. In features conf I have *1 for recording, *2 for att transfer and #1 for blind. In queue blind transfer works. For disconnect I have #0. I guess that * is somewhere defined as for hang-up the call, but where? I can't find it anywhere. Any help