Displaying 20 results from an estimated 11000 matches similar to: "How to show caller number ?"
2013 May 06
3
Joining an astablished call
Hi,
I don't know how to call this functionality, but what I want to do is join
an already established communication between PSTN---FXS_connected_phone
using my SIP phone (I have an asterisk v11 with digium TDM400P at home)
Is it possible? What I don't want is using the conference sound and
menu.... It's just a normal call between to channels that I have to join
for few minutes.
2012 Jun 23
2
Can't make call with TDM410P
Actually I can start and receive SIP calls (PC client, iphone client)
but?I have an issue with calling external number throught PSTN
(certified-asterisk-1.8.11-cert2).
I'm having this ?error when making a call:
*CLI> ? == Using SIP RTP CoS mark 5
? ? -- Executing [9000 at local:1] Dial("SIP/3000-00000006",
"DAHDI/1/4384019357,10") in new stack
[Jun 23 16:18:09]
2004 May 20
3
two-way synchronization accross a firewall fails
machine O is outside firewall, machine I is inside (machine names changed to
protect the innocent :-)
firewall allows ssh connections if inititiated from I to O, but not if the
other way.
both machines have an /etc/rsyncd.conf of:
[rt]
path = /tmp/rsync_test
comment = Test area
O runs rsync daemon, I initiates a rsync cammnad like
rsync -rvvv --delete --rsh=ssh O::rt /tmp/rsync_test
2013 Sep 03
1
How to use Skype ?
Hi,
I want to recieve calls to my Skype account and forward them to a SIP/FXS
line. I searched for chan_skype for asterisk (v11), but found it only
available for asterisk 10
I know that Digium gives no support for this module, but I am sure that
someone somewhere did write some tool to allow such connectivity.
Do have any idea if I can use Skype with my asterisk v11 ?
Thanks
--------------
2009 Apr 07
3
Logging Asterisk console
Hi all, in witch way can I put in a log file the asterisk console?
I have tried with some settings in file logger.conf but the log not
contain the same debug that I can see with "asterisk -rvvv".
I need it in debugging purpose for tracking some bug.
Thanks Enrico.
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2011 Oct 31
1
Starting asterisk turns bash console text white in rxvt
<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
<meta http-equiv="content-type" content="text/html; charset=ISO-8859-1">
</head>
<body bgcolor="#ffffff" text="#000000">
<font face="sans-serif">Hello all,<br>
<br>
I've googled
2016 Mar 18
3
Incoming INVITE with Portability Info and LRN
On Fri, 18 Mar 2016, Trey Hilyard wrote:
> I thought this would be as easy as
> exten => _XXXXXXXXXX\;rn=+19136630000,1,Goto(from_pstn,${EXTEN:0:10})
Have you tried the '_!.' pattern?
--
Thanks in advance,
-------------------------------------------------------------------------
Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST
2008 Oct 05
5
asterisk, phpagi and singleton
Hello,
I've this situation: 300+ simultaneous calls and dialplan like this:
exten => _X.,1,Answer()
exten => _X.,2,DEADAGI(check_status.php)
exten => _X.,3,Dial(SIP/other/${NUMBER})
exten => _X.,4,Hangup
exten => h,1,DEADAGI(cdr.php)
When project is running , I had a lot of defunct php scripts (I've exceed
mysql connection limits and so on, deadagi help a bit). The
2012 Apr 04
2
Asterisk 1.8 and DeadAGI
Dears;
In asterisk 1.8, it is not more possible to use DeadAGI?
Also, I found the below commands in the a2billing and I would to ask why it set the sequence 1 for the Hangup()? Maybe because it is related to the NoOp? How?
[a2billing-callingcard]
exten => _X.,1,NoOp(A2Billing Start)
exten => _X.,n,Answer()
exten => _X.,n,Wait(2)
exten => _X.,n,DeadAgi(a2billing.php,1)
exten =>
2010 Aug 28
1
Play a number of files to a caller
I want to be able to allow a caller to dial a ddi, system to verify
identity etc (this is all done)
I then want them to sit listening to music, until an event happens.
When this (external) event happens, I want to play a certain file to
the caller, using playback (so that they have ff / rw etc), and when
finished, go back to the music.
1) I thought of redirecting to an extension that played the
2018 Apr 10
3
withheld caller id
>>> > exten => _9X.,1,Dial(Dongle/dongle800/#31#${EXTEN:1},120,KT)
My suggestion would be to add a pause or two before dialing the phone number
exten => _9X.,1,Dial(Dongle/dongle800/#31#ww${EXTEN:1},120,KT)
D(digits): After the called party answers, send digits as a DTMF stream, then connect the call to the originating channel (you can also use 'w' to produce .5 second
2005 Feb 23
1
Zaptel (Junghanns 4BRI card) to cell phone problem
We have set up an HP DL380 with 3 4BRI cards, Fedora core 2 (kernel 2.6.10)
and asterisk (bristuff-0.2.0-RC7f with asterisk 1.0.5). 4 ports are
configured in TE mode and connected to the PSTN; the other 8 are in NT mode
and connected to isdn phones.
the other outbound calls to PSTN are fine, however, when we call cellular
phones, often audio is one-way (i.e.: the cell phone user can not hear,
2017 Feb 06
3
Call List Campaign to an IVR
> On Mon, 6 Feb 2017, Tech Support wrote:
>
> We were able to develop a feature to send the call to voicemail about 90% of the time. That way, an end user could (1) not be bothered by having to answer the call, (2)
> delete the message without listening to it, or (3) listen to the message when it was most convenient for them. That way, they were in control and things were
2017 Feb 06
4
Call List Campaign to an IVR
On Mon, 6 Feb 2017, Tech Support wrote:
> We were able to develop a feature to send the call to voicemail about
> 90% of the time. That way, an end user could (1) not be bothered by
> having to answer the call, (2) delete the message without listening to
> it, or (3) listen to the message when it was most convenient for them.
> That way, they were in control and things were
2020 Jun 14
2
Any api (agi/ari/ami) equivalent of "core show calls"?
Wow! I've been *-ing for about 6 years and had literally no idea about
that!
I can see a way I could put it to a different use, but it seems to be a bit
of a sledgehammer to crack the walnut of "how many current callers"
compared to one line of (albeit hacky) dialplan.
That's making me sound ungrateful. I don't mean to be!
On Sun, 14 Jun 2020, 22:39 Steve Edwards,
2015 Jun 26
2
Asterisk dialplan best practices syntax
Hi,
I've two yocto questions about the syntax of dialplan:
1. What's the "official" notation of each line: "=>" or "=" ? In the wiki
of Asterisk, I see very often "=>", however, what's the reason for both
syntaxes authorized ? Historical ?
2. To write info in logs/console, you have two commands: NoOp and Verbose.
Verbose seems to be
2010 May 29
6
Best way to limit outgoing calls per trunk
Hi Guys,
I am looking to use System() function along with some bash scripting to
determine if a Trunk is being used during certain time of the day or not.
Here is what I have in mind. Please guide me if you know a better way:
exten => s,1,answer
exten => s,n,System(/tmp/check.sh)
check.sh:
check EPOCH time => do an IF for certain times => Allow mutiple calls in
certain times and
2011 Jun 07
2
What is wrong in m
Hi everyone,
What is wrong in below asterisk application? The output should be content of
field booth_status from table booths:
[extension-status]
exten => _X.,1,MYSQL(Connect connid 127.0.0.1 root password my-extensions)
exten => _X.,n,MYSQL(Query allow_call ${connid} SELECT extension_status FROM
mytable WHERE extension=${CALLERID(num)} ORDER BY id DESC LIMIT 1)
exten =>
2009 Sep 09
1
MySQL question
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2011 Feb 19
1
[1.4] "show channels" in extensions.conf?
Hello
I was wondering if there were a way to use NoOp/Verbose to display
the output of the CLI's "show channels" from within extensions.conf?
Something like that:
===========
[incoming]
exten => h,1,Verbose($[CLI("show channels")])
...
===========
Any hint appreciated, thank you.