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...und that my dial plan in the extensions.conf file was not allowing me to dial *xx. Once I corrected my dial plan I was able to dial *0, *69, *78, *79, etc. Training the wife on how to actually use it was an entirely different issue. I have not tried to enable 3-way calling via the PSTN. "Jeff Otterson" <asterisk-users@jeff.otterson.name> wrote in message news:<6.1.2.0.2.20050821102616.03277590@wheresmymailserver.com>... > I have been looking for the answer to this question for a > while. Google-ing and reading the archives of Asterisk-Users has not > enlightened me...

I have found that fax detection is returning an error saying that no fax extension is present when I have defined one. The console returns this error: Aug 26 10:58:41 NOTICE[1112745536]: chan_zap.c:3989 zt_read: Fax detected, but no fax extension extensions.conf has: [default] exten => fax,1,Hangup exten => fax,2,Congestion exten => fax,102,Congestion exten => f,1,Hangup exten =>

Can PDAs be used as softphones/clients on asterisk? what i wanted to do is to set up 2 PDAs as softphone(client) which allows them to communicate each other through asterisk server(desktop) devices i have: pda compaq model 3680 pda sharp sl5500 access point desktop(asterisk) can i apply my idea on the asterisk? any guides? thanks in advance :) --------------------------------- Do

I'm having the same problem here. Any solution to this problem? -Manuel (sorry for top-posting, I'm having a stupid mail client here) -----Messaggio originale----- Da: Simon Brown [mailto:Simon.Brown@otterson.com.au] Inviato: giovedì, 1. luglio 2004 02:05 A: asterisk-users@lists.digium.com Oggetto: [Asterisk-Users] Dial plan errors I am attempting to implement the new features added recently where you can have "Goto(s-DIALSTATUS)" in the dial plan. My extensions.conf looks like this: exten...

Hi, Found a nice little video about a prototype phone from broadcom currently sitting in Microsoft WinCE lab. The video is at: http://channel9.msdn.com The video in question is an interview with Mike Hall titled "Windows CE and Windows Embedded Lab Tour". The clip dealing with the VOIP phone is right at the start so you don't need to watch the whole thing (although there is some

Hello all, I was wondering if anybody knows where one might obtain a PCI ISDN card supporting a single BRI for use with Asterisk in Australia (and using something like chan_capi). Because of the Isdn4Linux DTMF issue, I don't want one of those cards. I've already spent too much time messing about with my current card. I'm after something like the AVM Fritz! cards. I found one place

...> > __________________________________________________________________________ > http://www.webmail.co.za/dialup Webmail ISP - Cool Connection, Cool Price > > --__--__-- > > Message: 6 > Date: Wed, 31 Mar 2004 16:37:06 +1000 > From: "Simon Brown" <Simon.Brown@otterson.com.au> > To: <asterisk-users@lists.digium.com> > Subject: [Asterisk-Users] Register vith SIP provider from behind NAT > Reply-To: asterisk-users@lists.digium.com > > I cannot successfully register with, or even make calls to, a SIP = > provider > (such as FWD) with my...

I am configuring an asterisk server and I want to test the incoming configuration with my FXS handsets. I have the FXS lines able to call eachother and they can connect out the FXO lines. I changed the context for the FXS lines to "incoming" so that they would be able to test the setup for incoming calls. For the incoming context I have: [incoming] exten => s,1,Wait(1) exten

Hi all, After a long time having no time, I have finally done some fresh work on my software fax machine. I have replaced the original carrier tracking with something more robust. I have also added 4800, and 2400 bits per second modes, and cleaned up a few bugs in areas like superfine mode operation. I apologise for this update taking so long. At ftp://ftp.opencall.org/pub/spandsp you will

...gt; > __________________________________________________________________________ > http://www.webmail.co.za/dialup Webmail ISP - Cool Connection, Cool Price > > -- __--__-- > > Message: 6 > Date: Wed, 31 Mar 2004 16:37:06 +1000 > From: "Simon Brown" <Simon.Brown@otterson.com.au> > To: <asterisk-users@lists.digium.com> > Subject: [Asterisk-Users] Register vith SIP provider from behind NAT > Reply-To: asterisk-users@lists.digium.com > > I cannot successfully register with, or even make calls to, a SIP = > provider > (such as FWD) with my...

I have just installed one of the new TDM400 cards with an FXS and an FXO module into my * server. I also checked out the latest cvs head. I am using 7940 phones. Now I have some strange problems: 1. When in the VM menus, key presses do not register. 2. When I press "hold" on the 7940, it hangs up. Has anyone got any ideas? TIA Simon Brown

If I try to dial a SIP extension which is not connected/registered with *, I end up getting a busy indication and the call goes through to the "busy" voicemail message. The extension is listed in the sip.conf, but it is not connected at the time. Shouldn't it go to the "unavailable" voicemail message? Simon Brown

I am attempting to implement the new features added recently where you can have "Goto(s-DIALSTATUS)" in the dial plan. My extensions.conf looks like this: exten => s,1,Dial(${ARG2},20,r) exten => s,2,Goto(s-${DIALSTATUS}) exten => s-NOANSWER,1,Voicemail(u${ARG1}) exten => s-CHANUNAVAIL,1,Voicemail(b${ARG1}) exten => s-BUSY,1,Voicemail(b${ARG1}) exten =>

I am attempting to implement the new features added recently where you can have "Goto(s-DIALSTATUS)" in the dial plan. My extensions.conf looks like this: exten => s,1,Dial(${ARG2},20,r) exten => s,2,Goto(s-${DIALSTATUS}) exten => s-NOANSWER,1,Voicemail(u${ARG1}) exten => s-CHANUNAVAIL,1,Voicemail(b${ARG1}) exten => s-BUSY,1,Voicemail(b${ARG1}) exten =>

I have been looking for the answer to this question for a while. Google-ing and reading the archives of Asterisk-Users has not enlightened me. It seems that this question has been asked many times, and many times it has gone unanswered. I have call waiting and three way calling on my PSTN line from Verizon (the local telco). This is connected to a FXO port on a TDM400P. I also have

Can I do this with * ??? S,1,answer call S,2,play "thanks for calling, we'll be with you soon" S,3,play music while caller waits and ring nominated extensions at same time S,101,if not answered go to voicemail I can't find a way to play music and ring extensions at the same time. Any help would be greatly appreciated. Simon

When I start * I get 6 mpg123 processes start as well. Is this normal? Often after a couple of days these mpg123 processes start to consume cpu and I have to kill them off. I do not have a sound card in the server and I have noload => chan_oss.so Simon

I think you will find the functionality you are looking for is in * already. Here is an excerpt from the sample extensions.conf file that is included with the source: exten => s,1,Dial(${ARG2},20) ; Ring the interface, 20 seconds maximum exten => s,2,Goto(s-${DIALSTATUS},1) ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER) exten

I can't see for the trees :) I can make calls out to my SIP provider but get an "unable to authenticate <calling no.> when I try to call in via the PSTN number they have supplied me (where <calling no.> = phone number trying to make the call) sip.conf [general] register => 4316568:xxxxxxxx@sipgate.de [sipgate] secret=xxxxxx username=4316568 fromuser=4316568

I have quite successfully set up the Services button to work on the 7940 running SIP. I have a metric-imperial converter, a foreign exchange rate calculator, a calendar etc available to users. The XML is really fussy though. Simon -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Rich Adamson Sent: Thursday, 1