search for: fawthrop

Hi All I'm trying to get my Ultra UPS connected. It connects using a standard USB cable I have --with-drivers=hidups in the configure command and I have lsusb showing Bus 001 Device 002: ID 0d9f:0001 Powercom Co., Ltd Bus 001 Device 001: ID 0000:0000 I have /dev/usb/hiddev0 now matter what I try I keep getting upsdrvctl start Network UPS Tools - UPS driver controller 2.0.3 Can't

Hi All Curious will this work Std. PSTN line ---x------ X100p | ------ Fax Machine Using a standard "home phone" pstn line with a splitter connecting a fax machine and X100 Asterisk Box Incoming Line: Can I have in the dial Plan [incoming] exten => s,1,Wait(1) exten => s,2,IfFax continue to ring, so that the Fax Machine gets it exten

Has any advancement been made to get 3102 operational in either a SIP or H323 asterisk environment. A post back in time mentioned a downloader service. >From the posts and articles I have read, the NCP is acting like a bootp and tftp server which uploads the configuration to the phone?? Am I close? if so, where does one get the SIP image for he 3102 and 2102 phones? I had 8 donated, but they

Hi All I have been looking around and haven not been able to find a working example I have a fresh/new install of Asterisk 1.6.2.0 with dahdi 2.2.1 and libpri 1.4.10.2 I use a sangoma A200 card so I am using wanpipe 3.4.7 If I use zaptel which I read I need for app_rxfax then asterisk crashes with segfaults on startup asterisk[2624]: segfault at 30353466 ip b7eb538b sp bffda26c error 4 in

In trying to get the asterisk and faxing working I had to resolve to using iaxmodem and hylafax. I have incoming working, but outgoing the other fax rings but it would appear from web searches that the fax signals are too low to be "heard" I can read about rxgain and txgain for zapata. my fax setup goes direct from aterisk <-SIP-> SIP Provider <- Fax Machine -> It never

Hi All Could a VPN be used to help with SIP Tunneling and QoS issues. State 1: Two IP Networks Connected via the Public Internet transmitting VoIP Traffic Say a VoIP User and VoIP Termination Provider. Each side can put QoS onto their part, but if QoS does NOT exist between them then call quality will be bad anyhow. State 2: Same as above except a VPN tunnel is setup between each side. Thus

Hi All In Order to place a "call Me" button on a webpage which would you use ? [A] <a href="sip:1234@123.123.123">Call Me</a> or [B] <a mailto:"sip:1234@123.123.123.123">Call Me</a> Then what s/w is required on the users end to make the call so that it rings via an * server to a sip phone (ip phone) registered with the * server

Hi All I'm trying to understand how I would send my fax ? If I use Word or what ever word processor or even an email client to create what I want faxed. I have *asterisk setup with and FXO Gateway that will make the call to the fax number I dial SIP extension 320 is the FXO gateway. How do I now get my email or word document to TIFF to then fax to the FXO gateway or SIP/320 ? I

Hi all How much does configuring a network with VLANs improve or effect quality ? Is there much reason to justify the configuration of VLANs ( I know networking, but not VLANs at all) Would it not be better to find high traffic users and determine why? Your Thoughts Thanks Barry

Hi all Is the use of a VPN between IP-PBX and VoIP Provider a useful tool? Since the QoS and general traffic of the Internet can never be predicted, would the implementation of a VPN between Client and VoIP Provider increase voice quality and/or security or is the converse true ? Thanks Barry

Hi All My first experience with syslinux I have created a debian live bootable environment. Which creates a fat16 format USB-HDD and then uses syslinux to boot up with (Provided the hardware allows you to boot from USB). When I plug the USB Stick into say a windows machine it mounts the USB drive and you can see the linux kernel files syslinux and other files Question: Is there a way to

In having configured my first * server there are a few questions I could not understand or find answers to 1) How does one use ztmonitor to adjust the rxgain and txgain. I have set mine to -1.0 each to get rid of echo on std phones connected on the TDM10B FXS module 2) Is it best to use a TDM card with an TFX and FXO module or is the the FXO = X100p and FXS on TDM a good method? 3) How do I

...akage. (Alexey Ostrovsky) > 5. Re: Where to get 48 volt Power Supplies for > Cisco > IP Phones (Greg Boehnlein) > 6. extensions/sip from database? (Manuel Wenger) > 7. Re: IP local loop? (Shaun Dawson) > 8. Re: 2 Sip phones behind un-natted Asterisk > (Barry Fawthrop) > 9. RE: PRI problem??? (Timothy R. McKee) > 10. Re: Where to get 48 volt Power Supplies for > Cisco > IP Phones (Nicholas Ruddick) > 11. Re: 2 Sip phones behind un-natted Asterisk > (Bruce Komito) > > --__--__-- > > Message: 1 > Date: Mon, 24 May...

Greetings I was getting bad static crackle on a phone, so I reload from the latest CVS and did a make clean ; make install on zaptel, libpri and asterisk Now I get this error [chan_skinny.so] => (Skinny Client Control Protocol (Skinny)) May 28 13:59:42 WARNING[16384]: chan_skinny.c:2541 reload_config: Unable to get our IP address, Skinny disabled Urgent handler [chan_oss.so] => (OSS

I'm having with an echo or delay I connect to the PSTN with a x100p and then connect a std. phone to a FXS module on a TDM10B. The std phone is only 2-wire so I know this is not helping. (yes I have read the 2-wire 4-wire issue) I have tried many echocancel values. The best thing to help was rxgain and txgain. below is my current zapata.conf file All help would be grateful. I have tried

hi i bought a ultra ULT31502 ups someone seem able to run this ups with nut just see Dean thread on: http://www.tigerdirect.com/applications/searchtools/item-details.asp?EdpNo=770423&Tab=11&NoMapp=0 i use suse 10.1, i installed nut the ups is connected directely to my computer with an usb cable lsusb return Bus 001 Device 004: ID 0d9f:0001 Powercom Co., Ltd with lsmod, i can see

...fo/asterisk-users -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050318/1e87c4bd/attachment-0001.htm ------------------------------ Message: 22 Date: Fri, 18 Mar 2005 16:22:11 +0000 From: "Barry FAWTHROP" <brif8@hotmail.com> Subject: [Asterisk-Users] echo / delay problem To: asterisk-users@lists.digium.com Message-ID: <BAY102-F182C235C5E05ED3531C376E84A0@phx.gbl> Content-Type: text/plain; format=flowed I'm having with an echo or delay I connect to the PSTN with a x100p and t...

Hi All Has there been problems with Caller ID and Asterisk 1.2.10 ??? I have a Phone Number from Teliax, I get this chan_sip.c:10468 handle_request_invite: Failed to authenticate user "3523029577" <sip:3523029577@207.174.202.2>;tag=as03efd979 And the call does not go through. I have a CG-410 FXO gateway if I enable caller ID I get the same thing, without CallerID enabled It

Trying to setup samba as a PDC with LDAP on gentoo I have managed the setup and can access and register computers/ join the domain Having trouble with profile access (1) But when I try to connect an XP client in the Log file I get Error Opening File: for ntconfig.pol and login.bat both exist with 777 permissions in /var/lib/samba/netlogon and /var/lib/samba/netlogon/default user the log shows