similar to: Can not make progdocs

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, 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 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

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

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 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 beg to differ. Digium is hiding from the real world and somebody is going take the software and run with it. My customers lost in excess of $50.000 and cut my pay in half, because of hackers. The hackers figured out how to scan every asterisk for weak passwords or open ports, and bang them real good. We need two things: a) disable in sip.conf the reply for INVITES that have wrong user

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 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

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

My PSTN line doesn't allways hang up properly after it goes to voicemail. The problem occurs when a caller hangs up during the initial greeting. Even though the hangup accured, voicemail continues to record, usually a fast busy and/or a teleco generated "please hangup now" message. After the voicemail.conf 'maxmessage=180' expires the line simply stays offhook. The hardware

Hello, If my understanding is correct, these days it seems that many ISDN BRI lines are configured in energy saving mode in which signalling D-channel is "dropped" until a new call comes in. Is it possible to replicate this behaviour with Asterisk (when Asterisk is in NT mode and is seen as a public ISDN by another PBX, for instance) ? If not, would you it would be a useful addition to

Hi I have come across a problem, with my tdp410 and soekris board (basically pc on a chip amd geode cpu). I am using the box as a firewall/asterisk box. The problem occurs when I drop ppp and I get dead loop dectiotn going, I seem to lose interrupts and get lots of messages in syslog from wctdm24xx saying missed interrupt increasing latency its out lined here

Anyone tried to build * + h323 to rhel3... I have to problems in the process... a) Zaptel would not build - a whole bunch of errors about kernel... b) make progdocs failed with reference to dot - check your installation. Do I need the zaptel ?? I will not be using any interface cards.. I'd like to make progdocs - any suggestions there? Latest cvs on everything.. Thanks

You could accept as the "passcode" the caller punching in their own phone#, then checking that against your whitelist. Lets associates get past the challenge when using someone else's phone, without their remembering some arbitrary passcode. And strangers or barred old associates who abuse it can get an earful about how you're suing them for wire fraud. Preferably after you

I'm running * 1.4 and can successfully restart asterisk from the command line with: /usr/sbin/asterisk -r -x "restart gracefully" However, I have a cron job that tries to restart asterisk and gets this error: No such command 'restart gracefully' (type 'help restart gracefully' for other possible commands) Can anyone think of why this is happening? Thanks

Hello all, After installing Asterisk, i have installed the docs by "make progdocs". But i don't know where to locate this documentation. please Help. Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070810/ceb95948/attachment.htm