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...> monitor your set-up. The LCD runs on serial, of course. > > As an alternative, you can use any of the many available > relay boards -- $50 gets you this: > http://www.phanderson.com/iom141.html > > > -----Original Message----- > > From: James Bean [mailto:james@hdcs.com.au] > > Sent: Saturday, February 19, 2005 11:34 PM > > To: Asterisk Users Mailing List - Non-Commercial Discussion > > Subject: [Asterisk-Users] External relay triggered by Asterisk > > extension -question > > > > > > > > Has anyone every setu...

Hi, I have postgresql and * all up and running as the latest cvs-250205, although something weird. Every outgoing call regardless of whether or not it is answered or busy or just rings out in the database the entry has the disposition as ANSWERED, instead of BUSY or NOT ANSWERED. As a test I intentionally rang numbers that would be busy or wouldn't be there to answer the call. Anyone got

Scenario, I got some very good help earlier from Joseph getting me up and started but I have a couple of small problems still. Setup: FC2 Asterisk 1.0 Zaptel 1.0, TMP400P, FXS Port 1 FXO Port 4 Analog dialout line and Analog handset plugged in. Problems: 1. Incoming calls work and the phone rings and can be answered no problems, (although I wouldn't mind being able to adjust the ring but

Asterisk 1.0.7 Has anyone got the hint function working, and maybe with the GXP2000. I am testing with 2 GXP2000 phones (firmware 1.0.1.9) at the moment trying to get the LED's to light up. On ext 690, button 1 is setup for ext 691, I did this using both methods 691, and <sip:691@192.168.69.1> On ext 691, button 1 is setup for ext 690, I did this using both methods 690, and

> From: "James Bean" <james@hdcs.com.au> > Has anyone every setup an external open/close relay, off say a serial > interface, and have an extension trigger the relay? The following will do the trick. Just add a 5vdc solid state relay ('cause you can't sink too much current out of the RS232C port). Substitute &quo...

Sorry Newbie asking everyones option. I am setting up a couple of small asterisk phone systems for my work, I started using some snom 190 and bt102 sip phones (the bt102 works really well with iLBC), but the complaint from my workmates is there is no way to see if other people are on there phone or not, or what lines are being used. The snom 190 only has 5 function keys, the snom 220 seems a bit

Hi, I am sorry to be asking this but the wiki is down and has been for a couple of days and I need to get this working before Monday to get my live system setup. Trying to get the Snom 190's and soon to arrive 3com 3102's to use the function keys and for the life of me I can't work it out from the conversations on the archive what I am going exactly wrong here? The snom 190 with

Specs FC2, Asterisk 1.0.0, Zaptel 1.0.0 TDM400P Port 1 FXS Port 4 FXO Standard analogue handset plugged in with pstn line. Problem: When I go to dialout it drops numbers on the outgoing number. Keys dialed from handset were 9 0418800185 I tried hitting the keys slowly as well as at my normal speed, all tones are heard in the handset for all numbers.

Trying to setup asterisk and zaptel on a Fedora Core 3. Its all working after reading up on udev but I still get errors. [root@redhat ~]# ztcfg -vvvvv Zaptel Configuration ====================== Channel map: Channel 01: FXO Kewlstart (Default) (Slaves: 01) Channel 04: FXS Kewlstart (Default) (Slaves: 04) 2 channels configured. Notice: Configuration file is /etc/zaptel.conf line 4: Unable to

Hi Everyone, Merry Christmas :-).... My Asterisk Box doesn't have a sound card, it is running Asterisk 1.02 Zaptel 1.02 Libpri 1.02 Mpg123 0.59r All compiled from source with kernel 2.6.9-1.6 on Fedora Core 2 Any help would be very much appreciated..... The error I am getting is -- Executing WaitMusicOnHold("SIP/snom-james-849d", "30") in new stack Dec 12 00:27:29

Hi, Hoping someone has run into the same issue. I have an * 1.0.5 tdm400p and 2 fax machines on grandstream 486 boxes. When a fax comes in, no problem receives it fine. When you try to send a fax out just as the fax seems to be finishing the send you get a comms error on the fax machine and it fails wanting to retry (tried 2 different brand fax machines same issue). The 486's were

Hi, Asterisk 1.0.7 TE405P - Port 1 - ISDN30 telco - Port 4 - Primary Rate connection to Phone system The system has a mixture of 20+ sip phones and the 50 odd extensions on the phone system connected to Port 4. What I want to accomplish is to be able to denigh access to certain outgoing phone calls by the extension/callerid the call originated from. i.e. Only certain sip and telephone

FC2, Asterisk 1.0.0, Zaptel 1.0.0 TDM400P Port 1 FXS Port 4 FXO Standard analogue handset plugged in with pstn line. Problem: I have 2 analog phones that I use, when plugged directly into pstn line both phones work perfectly, dialing no issues. When I plug the handsets into the TDM400P, one works perfectly the other drops random numbers. Its like the tone is slightly different on the second

Just after some peoples impressions if they have used this phone. It has 10 function buttons which I am hoping can be individually programmed for destination to accept hints from asterisk. Any input would be very much appreciated. James

Sounds like what I had in mind, could you point me in the right direction to an example of agi scripts that might do this :-). I'm not well versed in the ways of the AGI and would flounder significantly. James -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Rob Thomas Sent: Sunday, 12 June 2005 8:45 AM To:

Charles S. Antrim wrote: > I am using a card that has an fxo and fxs module. I am no where near an expert but I have my sip phone working through my pstn line and this is my config. /etc/asterisk/sip.conf [general] port = 5060 bindaddr = 192.168.69.1 context = sip disallow = gsm allow = alaw disallow = ulaw nat=disable srvlookup=no localnet=192.168.69.0/255.255.255.0 subscribecontext =

Sorry to post such a newb set of questions but I have been hammering about trying to get Asterisk running on FC2 machine reading everything available (I think that is what stuffed me, shouldn't have read it all :-) ). Config FC2 running Asterisk 1.0.0, with the h323 compiled in and installed correctly. Amazingly enough I have everything compiled correctly and installed. I am running a

Hi, Running Asterisk CVS-Head latest on a Dual P3 800 1Gb Ram. For some odd reason now that I have the asterisk box almost to the stage I want it, I hit a problem. I have a te405p in the system, Zap/g1 is connected to the telco as an ISDN 30, Zap/g4 is connected by ISDN Primary Rate to an Ericcson BP250 phone system. When calls come in on g1 they go straight through instantaneously to the

FC2 Asterisk 1.0 When I dial a H323 dialup to an existing OKI Voip Router (BV1250), I get an EndedByRefusal yet the OKI Gateway is setup with the corrent reply ip addresses etc etc, unfortunately its an existing multiple voip router setup with g723.1 and g729a, so changing the codec on the router maybe an issue. I have compile in the h323 as per the channels/h323 setup with the listed libraries.

Hi, I have a weird situation where I have a noop command putting the callerid of the caller on my asterisk console so I know who is calling as a test, but it is putting the callerid of my extension in instead of the callerid of the incoming line. My /etc/asterisk/zapata.conf is [channels] context=default ;switchtype=national usecallerid=yes cidsignalling=v23 cidstart=polarity hidecallerid=no