Displaying 20 results from an estimated 110 matches similar to: "Dialplan problem - incoming calls get MOH, not ringing."
2004 Jan 10
2
Record all phone calls
I want to record all phone calls made inbound and outbound. I'm new so
having a hard time getting this started. Here is what I have so far but
isn't working. Can someone help me out? Thanks,
[macro-record-on]
exten => s,1,SetVar(CALLFILENAME=${DATETIME}-${ARG2}-${ARG1})
exten => s,2,Monitor(wav,${CALLFILENAME})
[sip]
include => macro-record-on
include => iaxtel
exten
2005 Jan 30
4
Processing incoming calls with multiple contextst over PRI
So I have a problem. A customer of mine wants a PBX, owns an office
building. I want to sell him on asterisk. He has 4 tenants. I am using
my asterisk box to simulate it. My asterisk box has a TDM400P card, not
a PRI card. Don't know if it makes any difference.
Anyway, I want to route incoming phone calls to different contexts based
on the phone number being called.
Here is my
2004 Jan 10
0
Record calls where to put line?
Here is what I have now. Where should the line " exten =>
_.,1,macro(record-on,${EXTEN},${CALLERIDNUM})" go should it be under [sip]?
Right now if I call sip to sip monitoring starts and the calls connect but I
only get 44 byte files. If I call and iaxtel number monitoring starts but
call never gets placed and again 44byte files with nothing in them.
Thanks for the help.
[iaxtel]
2003 Sep 22
1
Can't get simple config working!
Hi all.
I'm trying to get a simple configuration working so I can later expand it to
something more interesting.
I'm using kphone to call an extension on the * server. When I try to connect,
I get this error:
DEBUG[81926]: File chan_sip.c, Line 3562 (check_user): Setting NAT on RTP to 0
DEBUG[81926]: File chan_sip.c, Line 527 (__sip_ack): Stopping retransmission
on
2004 Aug 27
1
Problems dialing out with T100P and Adtran
I have a T100P card connected to an Adtran and then a T1.
I have added the following configurations to Asterisk...but, when I dial
9 and then a local phone number, it bounces between the dial tone and
silence and the *error* light on the Adtran blinks.
zaptel.conf
span=1,0,0,esf,b8zs
fxsks=1-8
loadzone=us
defaultzone=us
zapata.conf
[channels]
context=from-sip
signalling=fxs_ks
2004 Aug 25
2
asterisk & chan_sccp
ive got asterisk running with chan_sccp and three cisco phones (2 7910's and 1
7960). lots of bugs. when i press the speed dial button on either 7910,
asterisk dies. also, if i dial from the 7910 to 7910, everything works fine.
i can dial from or to the 7960 once, and then one of the 10's and the 60 die
and try to reregister.
if i take the 7960 out of the mix and remove its
2004 Nov 29
2
Problems with conference on FreeBSD 5.2.1 w/* 1.0.1
Hello,
I'm trying to set up a conference room. When I dial it's extension, I
get an audible error saying "Not a valid conference room, please try
again" followed by a disconnect. I've got debug sip peer 1001 (my
X-Lite client) and I see this in the logs: (I'm pretty sure it has
something to do with ztdummy, but I dunno... I have the port
installed, but I
2003 Aug 07
1
MWI bug ?
Hi Lee,
You need to specify the VM context that you are using..
so using your examples..
extensions.conf entry..
exten => 1000,1,Dial(SIP/1000,20)
exten => 1000,2,Voicemail2(u1000)
exten => 1000,102,Voicemail2(b1000)
exten => 1000,103,Hangup
should be..
exten => 1000,1,Dial(SIP/1000,20)
exten => 1000,2,Voicemail2(u1000@sip)
exten => 1000,102,Voicemail2(b1000@sip)
exten
2008 Jun 20
1
Voice only works from one way.
Hello, everyone.
Right now, we are trying launch our own PBX system based on Asterisk(Fedora)
with Cisco 2611. Cisco has 2 port FXO card installed on it.
For testing, I have 2611 hooked into phone line with number of xxx-xxx-xxxx
fine. (I'll call it F). Using softphone, I can dial in extension 1001 on
asterisk, which should talk to cisco. After initial connection to Asterisk,
I have try to
2003 Nov 11
1
Unable to use voicemail
Hello all.
Now I aleady installed the Asterisk.
I could make communication between 2 XLite client through Asterisk.
I tryed to test the voicemail function as follow.
1, I make a call to 1001 from 1002
2, Start ringing
3, Wait untill time out for ringing
If no problem, 1001 go to voicemail and unavailable message will
be played.
But 1001 receive a 403 forbidden massage and connection go
2015 Nov 24
2
subscriber state before dial
Hi All
After a Dial() I get:
WARNING[7964][C-000075a8]: app_dial.c:2437 dial_exec_full: Unable to create
channel of type 'SIP' (cause 20 - Subscriber absent)
if the subscriber is not registered.
Is there a way from dialplan to know, *before* Dial(), if a destination
Subscriber is
a) not registered or
b) busy ?
I need to redirect a call to some other Subscriber if (s)he is not there
2004 Jan 19
3
Getting correct CDR info
I'd like to know how everyone else is going about getting correct CDR
information for calls. Right now I notice that if a call come in and gets
parked the CDR info doesn't how the correct info on who picked that call up,
also when someone transfer a call there isn't a new record being made so the
duration of the call shows up for who received the call and transferred it.
I started
2005 Jul 19
2
No sound when bridging two single FXO cards
Wow ! No reply... May be I must talk about X100P instead of X101P ?
Is someone has yet encountered this kind of "no sound" problem when bridging
two FXO lines like this (first as input, second as output) ?
Any idea ?
TIA.
Best Regards,
Francois BERGERET,
France.
----- Original Message -----
From: "Francois BERGERET" <f6hqz-m@hamwlan.net>
To: "Asterisk Users
2005 Feb 25
1
cascaded ringing
Hi,
I intend to let several SIP-phones on my asterisk ring cascaded on
incoming calls.
First only phone 1 should ring, after 5 seconds phone 2 should ring in
addition and after additional 5 Seconds phone 3 should also ring.
How can I realize that correctly?
Currently I do use
Dial(SIP/1,5)
Dial(SIP/1&SIP/2,5)
Dial(SIP&1&SIP/2&SIP/3)
But this seems not to work correctly on
2004 Jan 17
1
Registering multiple FWD accounts
Can multiple FWD accounts be registered?
I have the following output in my sip.conf file:
register=74928:xxx@fwd.pulver.com/74928
register=75160:xxx@fwd.pulver.com/75160
register=74573:xxx@fwd.pulver.com/74573
[fwd-74928]
type=friend
secret=xxx
username=74928
host=fwd.pulver.com
[fwd-75160]
type=friend
secret=xxx
username=75160
host=fwd.pulver.com
[fwd-74573]
type=friend
secret=xxx
2011 Oct 12
3
FXS ports on TDM410P card...
My analog card, uses a PCI slot and a 12V power connector, is configured
with 2 FXO and 2 FXS modules. I can ring handsets connected to the FXS
ports but I can't dial out from them. Is extensions.conf where I need
to make changes?
[root at robin asterisk]# cat chan_dahdi.conf
[trunkgroups]
[channels]
[phone](!)
usecallerid = yes
hidecallerid = no
callwaiting = yes
usecallingpres = yes
2005 Mar 21
2
Ext matching problems
Hello everyone...
I'm trying to get up a testing pbx installation. Following instructions
of what've read from the handbook and from asterisk's wiki, I wrote the
dial plan as follows:
[general]
;
;
static = yes
;[globals]
;
[default]
;
exten => 0,1,Answer()
exten => 0,2,Playback(fcopba1)
exten => 0,3,Hangup()
exten => *0,1,Answer()
exten => *0,2,Record(fcopba1:gsm)
2003 Apr 09
7
Caller press "0" in Voicemail
I would like to add the ability for our users to be able to press "0" whenever reaching someone's voicemail box to re-reroute them to the auto-attendant.
Here's a sample extensions.conf:
[incoming]
include => ciscophones
exten => s,1,Wait,1
exten => s,2,Answer
exten => s,3,DigitTimeout,5
exten => s,4,ResponseTimeout,15
exten => s,5,BackGround(auto-greeting)
2004 Jan 31
0
Dial app does not indicate ringing to calling party
I hope somebody has seen this before...
I'm trying to use a Dial command on a inbound call to ring multiple
destinations. The calls come in to me from the provider on IAX2, and one
of the destinations I try to ring is a IAX2 to call to my cell phone.
When I add the IAX2 destination into the Dial command, the setup I am trying
to achieve works (i.e. my Zap, SIP, and cell phone all ring) but
2005 Aug 04
0
Calls not cleared down if extra destinations or dial commands added to extension
We have a weird situation where if the external called hangs up the call
before it is answered asterisk seems not to handle it if the original
dial command is replaced following a timeout.
We are trying to pass the call to the main reception, but if there is no
answer then it should ring another extension in addition to the first
extension the idea being that we don't end up with people