similar to: TDM400P FXO problems

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi!. I am having problems with getting asterisk to detect when someone hangs up. I have a TDM400P with one FXO module connected to my telco, and also a FXS-module connected to my phone. The FXS-module detects hangups just fine, but I can't get the FXO to detect them. I am pretty sure i have disconnect supervision on my phoneline since when I

Hi there! First mail on the list :) 1.- is it possible to use an spa2102 to make and revice calls from a "normal" phone? I mean, I know I can use it to connect an analog to an asterisk server, but I want to know if it can be used to connect asterisk to the analog phoneline. 2.- I'm trying to unlock the spa2102 with no succes at the moment, any links or hint will be very

Hi all, I was wondering whether any of you have experience/info on Cable and/or ADSL modems that would come together with a SIP phone adaptor. What I am interested in is something that would plug directly into you ISP's cable (be it ethernet or adsl/phoneline), would combine a modem/router/nat such that on the other you could simply plug in your RJ-45 cable for your PC and a RJ-11 cable for

I use both ports on my cisco ata-186. I run them using ulaw. Today I made numerous calls using my analog phone on port 2. I picked it up about an hour after the last call I made and the line was dead. There is no power at all over the phoneline to the phone, and the red light doesnt light up. The configuration is verified as unchanged. Has anyone seen this problem before. I was

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello everyone, I'm relatively new to the subject - so pleace don't punish me for idiotic questions. ;-) Can Asterisk act like a normal Sip phone and e.g. connect to another sip-gateway? Background: There is a new german company at: http://www.sipgate.de (sorry German only page) They offer a a gateway between a real telephone number and

https://bugs.freedesktop.org/show_bug.cgi?id=38673 Summary: all object is black Product: Mesa Version: unspecified Platform: x86 (IA32) OS/Version: Linux (All) Status: NEW Severity: major Priority: medium Component: Drivers/DRI/nouveau AssignedTo: nouveau at lists.freedesktop.org

It said 'include zapata-channels.conf', where this line wasn't commented bij the ';'... Could you post me a working example of such a config (or a part of it, for the X100P cards...? Thanks guys! Message: 9 Date: Sat, 19 Mar 2005 18:04:26 -0500 From: "Jeff Glassman" <jrglass@columbus.rr.com> Subject: [Asterisk-Users] newbie question To:

Hi, I'm calling asterisk with a swedish PSTN-phone line with caller presentation (DTMF) activated. I'm using asterisk 1.4.20.1 and cannot upgrade unfortunately, so I have to stay with this release. I use a TDM800P 8 channel PSTN card working as answering phones (I connect a phoneline with carrier signal to my TDM-card). Using zaptel-1.4.12.1. I verified that the DTMF tones of the number

Hi All, Maybe someone knows this, im using dahdi in combination with a TDM400, where 2 analog PSTN lines are connected. The weird thing is tho that when someone calls the analog lines it goes perfectly fine, the line comes in and all works ok. Except: Sometimes the callerid from the caller is not the complete number, but only a few random numbers from that phonenumber, and sometimes its complete.

Telstra (Australian Telco) has recently introduced a feature to allow the sending of SMS direct to fixed analogue lines, with an appropriate handset. As best as I can figure out, this appears to use CID type signalling, or at least on a line that otherwise has no CID on it, CID is sent, but with a standard modem I can only receive the date, time, and phone number (eg normal CID info). After that

Last week I was able to do some debugging of the problem I'm having with IAX2/GSM, residential-grade broadband, and VOIP. To summarize, I am having a great learning experience with * and Zap cards, SIP and IAX2. I hit a wall though, when I registered with iaxtel and tried doing VOIP. I spend the better part of a workday with the jitterbuffer and all sorts of settings and finally started to

Hello, One quick question. Does anyone has experience implementing unified messaging (UM) using Asterisk. Does Asterisk has support for UM ? Thanks, Tarun ___________________________________________________ Medicine meets Marketing; Dr. Swati Weds Jayaram. Rediff Matchmaker strikes another interesting match !! Visit http://matchmaker.rediff.com?2

Hello, I have configured my TDM01B Card (1 FXO Port ) as follows (below) but it will not pick up an incoming call. Any suggestions/tools to see what the problem is? I have looked at zttool where this line changes but I don't understand what it means (The last digit changed from 0 to 1) Total/Conf/Act: 4/ 1/ 1 /etc/zaptel.conf fxsks=4 loadzone=nl defaultzone=nl

Apologies if this messages appears more than once, somehow sending it via gmane's nntp-server doesn't work as expected. Been trying to send this message for 2 days now, without success so far... --- Hello, I'm exploring the capablities of Asterisk and must say I'm really impressed! However, I don't need most of the options, but can't figure out the things that appear

Alarms Span RED wildcard X101P Board1 OK wcusb/0 0 ive got my asterisk server up and running and working correctly, the first time after a reinstall and reboot everything was fine - i had both alarms OK and i could get the USB extension ringing when i ran the house number from my mobile. as soon as i tried again i got a red alarm on the wildcard board. now im using the sample

Hello, I'm a little confused on DTMF. A sip peer is registered on two Asterisk servers. No dtmfmode is set for them, the sip peer is 999 on Asterisk 1 and 999 on Asterisk 2. They both register on each other. A call comes in on Asterisk server 1, provider 1, dtmf=inband. Then the call is transferred to Asterisk 2: RetryDial(/var/lib/asterisk/sounds/connecting,15,10,SIP/12351 at

Hi there dear users and dear developers of Asterisk! I've got a maybe strange idea, let's see if you laugh or think it's reasonable J I'm using Asterisk with Digium TDM800P cards with 24 lines (as an answering machine). Each analog line can be reached through a phonenumber, since they are each connected to my telephone provider. Yes expensive I know J The challenge: I'd

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi! I am having problems with the PlayTones application and VoIP softphones. I have the following in my extensions.conf: exten => 123,1,Answer exten => 123,2,PlayTones(Busy) exten => 123,3,Hangup But when I connect with gnophone(IAX) or kphone(SIP) and dial 123 the call just hangs up immediately. I get the following on the console: --

I have an ATA-186 in a SIP configuration (following Shawn Djernes how-to), but I get the following error at the asterisk console when I try to call the phone connected to the ATA: ioctl(ZT_LOADZONE) failed: Inappropriate ioctl for device Failed to register zone 'United States / North America': No data available Everything works if I remove indications.conf from /etc/asterisk -

Hi, I have VoiceMailMain on extention 1001 so it would be nice to get that sent to the phone instead of asterisk@[IP Address] when there is a message waiting.. Is it possible to change the extention that is sent to the phone when MWI is lit up?? Thanks.. -- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by