Displaying 5 results from an estimated 5 matches for "dialphon".
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dialphone
2004 Jun 15
3
Grandstreams randomly go busy with Asterisk?
...g exactly like this.
I have two Grandstream BT101 phones connected to an Asterisk.
Periodically, for reasons that I can't determine, one or the other (or
both) of the BT101s decide(s) to go on permanent busy. Dialing that
phone gives:
-- Executing Macro("SIP/24567-7856", "dialphone|SIP/27654") in new stack
-- Executing Dial("SIP/24567-7856", "SIP/27654|10|tr") in new stack
Jun 15 14:23:41 NOTICE[1343506]: app_dial.c:536 dial_exec: Unable to create channel of type 'SIP'
== Everyone is busy at this time
But dialing in the other direction...
2007 Jun 21
1
TDM400 one way calls
...# more zaptel.conf
fxoks=1-4
loadzone=uk
defaultzone=uk
[root at asterisk asterisk]# more zapata.conf
[trunkgroups]
;define trunks here
[channels]
usecallerid=yes
hidecallerid=no
callwaiting=no
threewaycalling=yes
transfer=yes
echocancel=yes
echotraining=yes
immediate=no
;define channels
context=dialphone
signalling=fxo_ks
cidsignalling=v23 ; Added for UK CLI detection
cidstart=polarity
usecallerid=yes
channel => 1-4
[root at asterisk asterisk]# more extensions.conf
[general]
static=yes
writeprotect=yes
;
[globals]
FAX1 = Zap/1
FAX2 = Zap/2
STREAMLINE1 = Zap/3
STREAMLINE2 = Zap/4
CUSTSERVE1 =...
2005 Mar 08
1
TDM22B in the UK on BT
...rty hangs up the line.
I know i have to use the busydetect stuff but it doesn't seem to be
working.
It is a BT line and my zapata.conf is as follows:
[channels]
language=en
rxgain=0.0
txgain=0.0
immediate=no
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
signalling=fxo_ks
context = dialphone
channel => 1,2
language=en
rxgain=0.0
txgain=2.0
immediate=no
echocancel=yes
echocancelwhenbridged=yes
echotraining=yes
callprogress=no
busydetect=yes
busycount=3
usecallerid=yes
cidsignalling=v23
cidstart=polarity
relaxdtmf=yes
signalling=fxs_ks
group=1
context=incomingfrompstn
channel =>...
2007 Jul 26
1
tdm400p fxs module busy
...voice files
Exten => 501,1,Wait(2)
Exten => 501,n,Record(/tmp/asterisk-recording:gsm)
Exten => 501,n,Wait(2)
Exten => 501,n,Playback(/tmp/asterisk-recording)
Exten => 501,n,wait(2)
Exten => 501,n,Hangup
;
;goto voicemail
exten=>*98,1,VoiceMailMain(${CALLERIDNUM}@${CONTEXT})
;
[dialphone]
exten => 888890,1,Macro(fax,${FAX1})
;
[from-pstn]
;this is linked to zapata.conf and defines where the ddi points
exten => 888800,1,Dial(SIP/401&SIP/402,15)
exten => 888800,2,Voicemail(1000)
;
exten => 769611,1,Macro(oneline1,${FSEXT1})
exten => 769615,1,Macro(oneline1,${LONDO...
2013 Nov 16
3
Make phone ring through webserver using Asterisk
What is the easiest way? And how can it be implemented?
I thought to something like:
1. I request a page to the webserver
2. Perl sends to asterisk a number to dial (Perl and asterisk are
running in the same machine)
3. Asterisk calls the phone
or
1. A Perl sip client registers to remote asterisk server
2. Perl sip client sends to asterisk the number to dial
3. Phone rings