search for: lcrcomputer

Hello, I have a current asterisk 1.8.13.0 asterisk-addons 1.6.24 asterisk-sounds 1.2.1 dahdi-linux-complete 2.6.1+2.6.1 libpri 1.4.12 and asterisk-gui 2.1.0.rc1 (not trying to use the gui, want to do everything by hand) with a TDM410 with 2FXO and 2FXS. I have my POTS (PTNS) line plugged into port 1 (FXO) and a analog phone connected to port 3 (FXS). I compiled asterisk with asterisk

I'm looking into getting a new phone and wondering what the difference in functionality is between a single line phone with call waiting and a real 2 line phone (either a real SIP phone or an analog 2 line phone and a 2 port ATA) is. Why would I want the real 2 lines vs. just being able to take an incoming call via call-waiting? Cheers, b. -------------- next part -------------- A non-text

Hello everyone.... Searching the archives and google always comes up with entries regarding the "dyn" dns option in the 7960, but I can't find answers to my specific question.... My 7960 is connected via cable modem and is NAT'ed (everything is working fine). On the 7960 under SIP configuration\NAT Address I have the public IP of my cable connection. Comcast gives me a

I am putting together my first *. I had it running with two other pc's running xlite and setup voicemail and a couple of menus and submenus and had that running well. I had order a couple of oem x100p cards from digitnetworks. I installed them as they said with their voicepet2.2.zip drivers and did the modprobe on zaptel and wcfxo and then ran ztcfg -vv and got this: Zaptel Configuration

I just spent the last hour or so trying to get this firmware to work across a NAT with no success. I have a GS BT101 working through the same NAT, so I don't think it's the NAT itself. I have a STUN setup in * and pointed the UIP200 to it and I tryed several combinations of nat= in the sip.conf and in the config files for this phone. No luck(yes, I did a reload now with each change in

The orginal did not make it to the list... Spam filter issue??? No repeat of the lockup yet. Lyle -------- Original Message -------- Subject: voicemail locked up Asterisk 1.4.13 Date: Thu, 01 Nov 2007 20:57:27 -0500 From: Lyle Giese <lyle at lcrcomputer.net> To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> I am running Asterisk 1.4.13 with libpri 1.4.2 and zaptel 1.4.6 on openSuSE 10.2 (64bit kernel) with an AMD dual core 64 bit processor at 2ghz and 1g of ram. Motherboard has a VIA chi...

...; > Subject: Re: [asterisk-users] Telemarketer Torture.... > To: asterisk-users at lists.digium.com > Message-ID: <20080316143700.2e1952af.g.stewart at horwits.co.uk> > Content-Type: text/plain; charset=US-ASCII > > On Sun, 16 Mar 2008 08:50:50 -0500, Lyle Giese <lyle at lcrcomputer.net> > wrote: > > > I just forward them to one of those two extensions. If callerid > worked > > more reliably I would automate it. But I get a lot of caller id > failures > > on my incoming POTS lines, esp when calling in from my cell phone. > > The way I w...

I'm connecting a T1 PCI card to a Nortel Option 61 switch T1 card. My Sangoma A102D shipped with 2 T1 cables - which I assume are straight through. Do I need to make crossover cables for this scenario? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071026/9cea5e74/attachment.htm

I am facing one way audio problem in sip trunking between asterisk and avaya. +-------------+ +----+ | avaya sip |-------| P1 | +-------------+ +----+ | | | +-------------+ | Asterisk | WAN

...> ? 4. Re: Securing Asterisk - How to avoid sending, "SIP/2.0 603 > ? ? ?Declined" (Paul Belanger) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Sat, 23 Jul 2011 09:29:26 -0500 > From: Lyle Giese <lyle at lcrcomputer.net> > Subject: Re: [asterisk-users] use dahdi for local terminal modem > ? ? ? ?access? > To: asterisk-users at lists.digium.com > Message-ID: <4E2ADAC6.4010101 at lcrcomputer.net> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > > On 07/22/11 22:4...

Hi All, I hope I'm posting this to the appriopriate list, and that cross posting to two lists is OK. (If not, I'm sure I'll hear about it quickly :)) I'm running Asterisk on my (new) VIA EPIA-V motherboard. This seems to be the ideal platform for a home version of asterisk - its small, quiet, low power, and should have plenty of computing horsepower if only it would work!

I am starting to play with * and put it on a SuSE machine and had it on a NATed ip and I could dial 500 successfully from the console. I changed to name of the host and the domain name on it and put a public IP on it and now I cann't dial into the digium test number. Is it down or have I hosed up the system in someway that I have not discovered? I am behind my own Cisco router and all port

Not even sure how important this is considering the state of many of the online docs... I have doxygen installed as is noted for the requirements for 'make progdocs', but the make doesn't find dot. I have no idea where dot went, is or should have been... I am installing und Suse 9.0 and it's rough. If you forget something duringthe initial install, adding the package later

I am trying to get distinctive ringing going again with these phones, depending on the outside line the call comes in on. I had a working 1.0.x Asterisk setup using: SetVar(ALERT_INFO=<http://127.0.0.1/Bellcore-dr2>) Which used the short quick rings. In Asterisk 1.4, I have tried several things, but I think the correct syntax is: Set(_ALERT_INFO=<http://127.0.0.1/Bellcore-dr2>) But

I had a working system using * 1.0 and then 1.2 and now Asterisk 1.4.13 with addons 1.4.4, zaptel 1.4.6, libpri 1.4.2. I have a mix of Grandstream (GXP2000), Uniden uip-200, Linksys Wireless G, and analog phones via Adtran chan bank. When I went to * 1.4.13, the Uniden phones stopped being able to login to voicemail. All phones are on same lan with Asterisk. I get 'Login incorrect'

I am not that good at regex and it's use in Asterisk. I am running Asterisk 1.4.13 Currently I have this in my extensions.conf for incoming calls on our house phone line: [housemenu] exten => s,1,GotoIF($["${CALLERID(num)}" = "815xxxxxxx"]?s|12); 815xxxxxxx is our home phone number, when caller id fails or is missing that is what is recorded. I want to expand this

I got a Uniden UIP200 and started to configure it and I am lost.... I have a tftp server setup on my * server and have the files unidencom.txt and uniden<mac>.txt there. But it doesn't quite work yet. It registers as a sip phone (sip show peers), but I cann't dial it and the display shows #1 disconnected all the time. It has firmware version BS4.59a in it. I have no idea if I

I am setting up my * for at home office and still have analog phones attached and answer from those analog phones and not necessarily through the pbx. I found that with the X100P cards, they see the 2nd ring and will be ready to answer the line. I used a Wait to pause and allow another 2 rings before * answers. But found that if we answer the line after the 2nd ring and before the 4th, * still