Displaying 20 results from an estimated 400 matches similar to: "AAH: DNID not set if caller suppresses CID?"
2005 Jan 18
0
AMP and Asterisk PSTN extension config
Hi,
I have configured an Asterisk server with TDM01P (1FXO) for testing purpose. The interface I'm using is AMP. I want to configure my extension so that when I dial from my mobile phone to the asterisk line, I want it to transfer the call to any extension, say 3042 and after a particular number of rings, transfer the call to voice mail so that I can record my message.
My Zaptel.conf is as
2005 Jul 11
1
ASterisk@home + Broadvoice = Almost working installation...
Hello Guys,
I'm somewhat of a newbie and am desperately seeking for some help...
I've managed to get asterisk up and running on my server, and signed up with a
broadvoice account...
I'm having no problem dialing and communicating between extensions, but whenever
anyone tries to call my broadvoice account, they are greeted by no ring or
anything, but rather simply a direct to
2005 Feb 28
2
Fax Failing
Hello All,
I am trying to set up faxing using Asterisk@home 0.6. I have followed
the instructions to the best of my knowledge. When a fax comes in, the
system seems to detect OK but does ot manage to make the fax to pdf to
email leap. Here is what I saw in the CLI when I tested. Any help
would be appreciated.
Thanks!
Wiley
-- Starting simple switch on 'Zap/2-1'
-- Executing
2005 Aug 21
0
Problem with auto-attendant config, I think..
I never heard anything on the AMP list, so figured maybe someone here might
be able to help me sort this one out..
I was making some updates to my attendant config, which is really very
basic, and now incoming call processing stopped. Not sure exactly what the
heck happened, but figured maybe someone could help me with a clue as to what
broke. Now incoming calls are not being answered at
2005 Aug 04
1
no ring to callers?
OK, i've got asterisk @ home 1.3 up and running with Broadvoice.
BUT I have one nagging problem to sort out. When you call my BV # the
calling party gets no ring indication, just silence until either I
answer the phone, or the call bounces over to voicemail. below is the
console output when a call is recieved. what am i missing here?
thanks
Bernie
-- Executing
2005 Sep 28
0
Trying to cut out the paper work...
Hello everyone,
Ok. I am at a bit of a loss.... and would like someone to point me in
the right direction...(btw www.google.co.za did not give me ANY solutions).
The issue at hand is simple, I get asterisk (1.0.9) to answer the
incoming call with no problems... it does the fax detection thing with
app "Answer" and well it goes to the perfectly right context and sets
the varibles
2005 Jan 26
1
Inbound analog Telco line not answered
I have an X100P clone hocked up to an analog line of my PRI. I can use it
to dial out.
but when I call the extension it answers and says "GOODBY"
I have a Livevoip DID which successfuly rings to ext 202
I am using asterisk@home and through the AMP inface the line should ring to
ext 202
Below are Asterisk Messages, Extensions.conf and Extensions_additional.conf
Extensions.conf
2004 Dec 21
5
AMP - Fax Detections
Does anyone know of any obscur reference for detecting an incoming fax.
I currently have AMP running and everything else is working great.
Installed the spandsp patches and software... using the default AMP
extensions.conf, I start sending a fax, I hear it pick up and transfer
to voicemail after 20s.
Fax is set for system... Here is the detail from the extensions.conf
[global]
FAX_RX = system
2006 Feb 28
1
FW: Re: Delay on Phone ringing
Skipped content of type multipart/alternative-------------- next part --------------
asterisk1*CLI> soft hangup Zap/1-1
Requested Hangup on channel 'Zap/1-1'
== Spawn extension (macro-exten-vm, s-BUSY, 2) exited non-zero on 'Zap/1-1' in macro 'exten-vm'
== Spawn extension (ext-local, 220, 1) exited non-zero on 'Zap/1-1'
-- Hungup 'Zap/1-1'
--
2006 Apr 07
1
Inbound PRI calls drop after 5 seconds using Sangoma A101
Hi Folks,
I'm have Asterisk version 1.2.1 with a A101 PRI card. I'm working with the
CLEC to bring up the PRI and inbound calls are hanging up at his end after
a few seconds. I ran PRI debug but it only gives me minimal insight.
" Ext: 1 Cause: Unknown (16), class = Normal Event (1)"
He ran a trace and the only difference he is seeing is a
"ISDN interface explicitly
2006 Feb 22
0
debugging asterisk configuration
I'm trying to create a new contex for incomming calls from certain
trunks. My problem is this calls are not checked through ext-did (for
incoming routing). The calls from standard trunks are filtered
correctly but these ones are not. Is there some way to debug what
file/line is being executed by asterisk? My custom context is this:
[from-pstn-nofax]
include => from-pstn-custominclude
2006 Jan 30
8
Analog with channel bank - Inbound works, outbound doesn't
I am experimenting with an asterisk setup in my office. The last bit I have to test is working with analog lines. I have TE411p digium card, with an ISDN line plugged into the first, a channel bank plugged into the second port, and the last two ports empty. I have the following setup in my zaptel.conf:
span=1,1,0,esf,b8zs
bchan=1-23
dchan=24
span=2,0,0,d4,ami
fxsks=25
And in zapata.conf, I
2005 Aug 02
12
WHat does it take
How many times do you ask for help here before getting a respone? Every
single thing I do No matter what I get busy extensions. I am willing to pay
someone to help here. Anybody got a clue?
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2005 Feb 11
2
transferring a IAX call into a conference
When I make a call out on the Faktortel number I am then able to
transfer to call to my asterisk meetme room of 801 by hitting 'transfer'
then '801' then 'send' on my grandstream phone.
This connects my faktortel trunk (and who ever is on the other end) to
my conference room I can then make another call using my local pstn
service and set up a 3 way (or whatever number
2000 Apr 16
0
Trying to get cmdline going...
Hello all,
Another johnny-come-lately courtesy of the Slashdot effect... I have
taken a moment to browse the archives, so I'm not asking "Why doesn't
the cmdline utility compile?" yet again. I have been doing some
fiddling with it and found a few thinkos to fix, but some of the more
esoteric getopt() things are stumping me.
Ferinstance, it seems that
<
2005 Jul 08
1
Serial port DTMF Caller-ID reciever /w scripts for X100P/X101P/Clones....
Since the X100P/X101P/Clone cards does not work in all countries that
use DTMF based
Caller-ID systems, I've developed a hardware that you connect to a
serial port and the PSTN.
You then run a perl script "cid_logger.pl" as a daemon, and modify
extensions.conf to call
an agi script whenever a call comes in, and if it's on the X100 card it
will get the caller id
information
2006 Mar 30
3
asterisk doesn't wait for whole extension
Hi,
maybe a dumb question, but it seems that some calls are directed to our
central dial in number despite the extensions the callers say they dialled.
E.g. they dial 1234-567, asterisk recognizes 12345, it says this is an unknown
extension, where it is right, and redirects the call to the central dial in
extension 1234-0. This only seems to happen when the numbers are dialled
manually. When
2010 Aug 30
2
help with dialplan
Todd
How do you have the context in the phones sip configs set?
Bryant
From: "Todd Reese" treese65 at gmail.com
Hi all,
I've been have problems with getting this system on line and would like
to acquire some help with the extensions.conf.
My current problem is that the phones won't dialout.on the VOIP lines
listed as dialout1, dialout2, dialout3. This version of asterisk
2004 Oct 23
7
Asterisk and Broadvoice, no incoming voice
I just signed up for the BroadVoice service a few hours ago, but for
the life of me I can't get any incoming voice. The incoming
connection is fine as it rings my extension from outside, but I can't
hear anyone talking. Outgoing voice is working fine though.
I've been looking through the archives, but I haven't found a solution
to the problem yet. I even tried another router
2006 Oct 20
2
getting DID info..
This might be a newbie question... I'm using a SIP trunk and trying
to get DID line information on an incoming call. All I hear is a
nice lady saying 'Zero' - then the call continues... Any suggestions?
thanks
Todd
exten => s,n,Set(DIDID=(<${FROM_DID}>))
exten => s,n,SayNumber(DIDID)
or
exten => s,n,Set(FROM_DID=${EXTEN})
exten =>