Displaying 20 results from an estimated 1000 matches similar to: "No subject"
2009 Jan 28
5
Inbound Call Disconnect in 3 seconds
My Inbound calls lands , but line get disconnect in exactly 3 secs.
Here goes my extension.conf setting :
[from-ipkall]
exten => 901835,1,Ringing ; call ringing
exten => 901835,2,Wait(1) ; Wait 1 second for CID delivery from PRI
exten => 901835,3,Answer ; Answer the line
exten =>
2009 Feb 02
2
Invalid Extension
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
CLI Output :
~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
vicidialnow*CLI>
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from
2005 Mar 22
1
NEWBIE: MWI on 7960
Situation:
* New Install of Asterisk
* 7960 w/ SIP 7.4 Image
* 7912 w/ SIP040406A
* 3 Lines Defined on the 7960 (5104,3100,2100)
Questions (configs are below):
* Why won't the MWI light on the Cisco? I've tried:
* mailbox=2100
* mailbox=2100@default
* mailbox=2100@wvlandsales-voicemail
* Does anything look goofy overall? :-)
Thanks,
George
Sip.conf
[2100]
2009 Jul 20
0
No subject
timeout to be set.<br>
I'm hoping to find an option along the lines of the Dial() ringtime,<br=
>
but no luck.<br>
Gosub() looked interesting, but I don't think quite fits my needs eithe=
r<br>
<br>
Could someone please offer a little insight on this situation and<br>
point me towards the right command to be playing with?<br>
2007 Jul 20
2
priorityjumping not working, Dial goes to n+1 not n+101
Priorityjumping is totally ignored by my asterisk (tested 1.4.4 and
1.4.7.1 on FreeBSD 6.2)
[general]
priorityjumping=yes
With n+101:
exten => 1337,1,Dial(SIP/zytek,5,Ttj)
exten => 1337,102,Dial(SIP/zytek,${RINGTIME},${OPTIONS})
exten => 1337,n,Hangup
-- Executing [1337 at firma:1] Dial("SIP/113-087a3000", "SIP/zytek|5|Ttj") in new stack
-- Called zytek
2008 Aug 20
1
vicidial mysql problem
I installed asterisk, astguiclient, php and mysql. but when i dialled one
number to another number my asterisk server give the following error:
> /var/lib/asterisk/agi-bin/agi-VDAD_ALL_inbound.agi
> install_driver(mysql) failed: Can't load
>
'/usr/lib/perl5/site_perl/5.8.8/i486-linux-thread-multi/auto/DBD/mysql/mysql.so'
> for module DBD::mysql: libmysqlclient.so.15:
2004 Apr 06
0
quad BRI. Outgoing calls droped in 10 seconds.
We have quadBRI configured 1 port in TE mode 2,3,4 ports in NE mode.
We are trying to place a call from the phone connected to BRI card port #4 to
city number through ISDN line connected to port #1.
Number successfully dialed. Person on the other end answering the line. But
conversation can't last more then 10 seconds.
Below is a log of such call.
Its not clear for me why we appear in
2007 Nov 01
3
Outgoing PRI CID?
We have now got our new PRI line (10 channels, 100 numbers) connected
and everything is working except the outgoing caller ID. Whatever
SIP phone I'm using, the CID that's shown is the very first number...
----- s n i p -----
[default]
include => outgoing
include => priin
[outgoing]
exten => _NXXXXX.,1,Macro(dial,08${EXTEN},${RINGTIME}) ; Local number (w/o areacode) -
2019 Jan 09
2
Switched from Asterisk 1.8 to 13 - CDR ringtime now always zero (Joshua C. Colp)
Regarding this I've read the specs linked to in detail, but I can find no mention anywhere of any change that implies or states that no ring time will be recorded anymore in Asterisk 13 and that all times in start and answer columns will now be equal for all calls.
Can this be because I nowhere use the Answer() application in my dialplan when dialing out?
-----Original Message-----
From:
2008 Jul 29
0
Fallback on a fallback
I have two sites running Asterisk PBX. Normally the inbound calls go
through a 3rd (colocated) server and are routed via IAX to the site
(the site registers with the main server)
I created a macro that tries to ring one location and then another.
Each site explicitly Answer() the call even though it will only ring
all the sip phones at the relevant location. When fall back is in
effect it goes to
2000 Oct 30
3
ssh-agent and ssh-add with openssh-2.2.0p1 on Redhat 7
Hi all,
i'm trying to figure out if i'm being silly or if there is a genuine problem.
Running on the notorious Redhat 7, 2.2.16-22 #1, X86.
[user at host]$ ssh-agent -s
SSH_AUTH_SOCK=/tmp/ssh-XXYFcFR6/agent.2101; export SSH_AUTH_SOCK;
SSH_AGENT_PID=2102; export SSH_AGENT_PID;
echo Agent pid 2102;
[user at host]$ echo $SSH_AUTH_SOCK
[user at host]$ echo $SSH_AGENT_PID
[user at host]$
2005 Jun 15
0
Re: Asterisk-Users Digest, Vol 11, Issue 100
Jon, thanks for your help, but I'd rather not do it using agents and
queues, ideally what would happen is it would simply play the message
and wait for the person to press a button, if nothing is pressed, it
just keeps going down the list. Any other suggestions?
asterisk-users-request@lists.digium.com wrote:
>Date: Wed, 15 Jun 2005 00:53:14 -0500
>From: Jon Gabrielson
2005 Jun 16
1
Newbie question about pressing a key to, be connected to the caller
Jon, thanks for your help, but I'd rather not do it using agents and
queues, ideally what would happen is it would simply play the message
and wait for the person to press a button, if nothing is pressed, it
just keeps going down the list. Any other suggestions?
asterisk-users-request@lists.digium.com wrote:
> Date: Wed, 15 Jun 2005 00:53:14 -0500
> From: Jon Gabrielson
2009 Jul 20
1
callforward with asterisk-gui.problem with stdexten
Hello, i am trying to enable call forwarding on asterisk 1.6 with
asterisk-gui
If i set my stdexten as follows (with the lines i marked) everything seems
like working.
But if i make any change on asterisk-gui and apply it.. it recreates the
macro-stdexten and deletes my configuration regarding to it.
So where should i add my call-forward configuration???
Where am i making a mistake??
2009 Feb 18
1
Accumulated call time
Hi All,
Asterisk 1.4.12 CentOS 5
My ISP account includes nearly 500 minutes of VOIP calls per month but
the service is expensive for unbundled minutes. So I'm trying to find
a way to keep an accumulated total of calls made through that trunk so
that I can automatically switch to a lower-cost provider when my
bundled minutes are used. The plan is to store the accumulated time in
AstDB and
2008 Dec 02
1
Need help for transfer
Hi All,
I need to stop the transfer feature on particular sip user.
I am using linksys phone and it has set the forwarding enable to another
user.
I have three users 2101, 2102, 2103.
2102 is registered in linksys phone with forwarding enable to 2103.
But is there any procedure in asterisk that we can not allow 2102 not to
forward on 2103.
and also i want to prevent the SIP/2.0 302 Moved
2005 Feb 17
1
(Kphone) Registration Failed: Forbidden
I just can't get kphone to register with asterisk, i can make calls to
the demos and even get into the mailbox but kphone cannot register.
Here's my story. Can you help me?? Please
I have installed asterisk on debian using apt-get install asterisk.
I have configured an extension in extensions.conf as follows
exten => 8003,1,Dial(Sip/8003,${RINGTIME},rt)
exten =>
2003 Aug 26
0
TDM10M && Siemens Euroset 2015
Hi all,
--------
I have installed a TDM400 with one active FXS port (TDM10B) an connected
it to a Siemens Euroset 2015 analogue phone.
I have installed some smom IP phones to the network as well and
configured them as usual (sip.conf). For configuring the TDM10B I have
used FXO signalling in /etc/zaptel.conf and in
/etc/asterisk/zapata.conf. I definded the TDM channel and the Snom
phones to the
2012 Dec 19
1
Dialplan - working out when users answer
Hey guys,
I've got a part of my dialplan that dials multiple people:
exten => direct,n,Dial(${QUEUEEXTS},${RINGTIME})
Multiple extensions are in the ${QUEUEEXTS} from an external script - e.g. SIP/100&SIP/101&SIP/105 etc
This works great, however I want to see if I can find a way to work out (and run an AGI script) when the call is picked up by someone.
Thanks all!
2020 Feb 04
0
Always Be Conferencing v16e - pure AEL-based dial plan solution
/****************************************************************************
* *
* Always Be Conferencing (ABC) *
* *
* Creator: chris @ Penguin PBX Solutions *
*