Displaying 20 results from an estimated 200 matches similar to: "Calls Declined"
2009 Sep 08
1
SIP Error
*I am getting below CLI in my asterisk :*
== Manager 'sendcron' logged off from 127.0.0.1
-- Executing AGI("SIP/cc101-b7910cc0", "agi://127.0.0.1:4577/call_log")
in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing Dial("SIP/cc101-b7910cc0",
"SIP/Sama203/119545090201||tTor") in new stack
--
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 =>
2010 Feb 14
3
Line DC
My dialer works perfectly , but whenever I dial a number manually from xlite
and press a Key like 6055 for DTMF , line gets disconnected. Line gets DC as
soon as I press any key from xlite
What could be the issues ?
I tried the SAME VOIP from another center and Its Ok there.
I tried the Same dialer Xlite over Static IP, problem is there.
I tried the same number from other Dialer , it works
2009 Apr 10
0
IVR and DTMF
REPOSTED with MORE Info and Modified Subject Line:
--------------------------------------------------------
I am using one of the Minute Provider to dial out USA numbers.
Now in one of my process, we need to Dial IVR and the enter DTMF digit and
then it connects to the automated IVR.
When I dial out the IVR directly using Xlite and VOIP Mins provider , it
works perfectly. but when In try from
2009 Aug 20
12
IPKall and FWD
We all know the FWD is NO more available.
How to set up IPKALL so that my Inbound number rings on my eyebeam or xlite
?
Any alternative for FWD ?
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090820/4206395a/attachment.htm
2009 Jan 25
10
CentOS and BAT File
In windows, we use BAT file to execute few series of command , which help us
in not writing each command manually everytime we want to execute those
commands.
In CentOS, I want to do the same thing.
Any Advice ?
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090125/d67fb239/attachment.htm
2010 Mar 04
9
30 mins GSM file
I need to create 30 mins of GSM file for Asterisk .
Silent / Blank file.
Whats the best way to create it ?
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100305/b92821c0/attachment.htm
2009 May 19
8
Ghost ??
We are using asterisk and sometime when our guys are on call , they hear
some voice of person and amazingly that person is NOT from our center.
Any one faced this kind of thing ?
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090519/7fe54bec/attachment.htm
2009 Feb 24
8
HDD FULLL
I have 320 GB SATA HDD.
When I checked my phpsysinfo, it shows 95% HDD is filled.
[root at vicidialnow ~]# df
Filesystem 1K-blocks Used Available Use% Mounted on
/dev/sda2 301924504 285002780 1337472 100% /
/dev/sda1 101086 11062 84805 12% /boot
tmpfs 1553832 0 1553832 0% /dev/shm
[root at vicidialnow ~]# du
16896 .
You have new mail in /var/spool/mail/root
[root at vicidialnow ~]# df -i
2009 Jan 28
4
Call Recording Alias
Modified httf.conf file and added :
------------------------------------------------------
Alias /recordings/ "/var/spool/asterisk/monitorDONE/"
<Directory "/var/spool/asterisk/monitorDONE">
Options Indexes MultiViews
AllowOverride None
Order allow,deny
Allow from all
</Directory>
Created a folder under vicidial as recordings.
FULL_RECORDING is also enabled.
2009 May 19
9
Hang at 5:34 pm EST
Some at 5:34 pm EST DAILY, all my call get disconnect.
I tried RE-INSTALLATION, I tried Reinstallation on a virgin HDD, but its
same.
I tried changing VOIP provider I tried changing Internet Provider..But no
help..
What could be the reason ?
Here are my enties of crontab :
### recording mixing/compressing/ftping scripts
0,3,6,9,12,15,18,21,24,27,30,33,36,39,42,45,48,51,54,57 * * * *
2009 May 01
9
LoadAvg , Codec and Bandwidth Utilisation
1) If I see the Loadavg more than 4 , whats the immediate solution to get it
under 1 APART from restarting the server ?
2) I get too much of cross connections.
Can Codec be the culprit ? I use g729. Can using GSM will solve the problem
? What could be the other reasons ?
3) Anyway to measure the bandwidth utilisation from the server ?
-------------- next part --------------
An HTML attachment
2009 Jan 13
9
FWD and Asterisk
I have an account with FWD and I have configured my SIP.conf with
[fwd]
type=friend
secret=password
username=901835
host=fwd.pulver.com
But when I am trying to dial out my own DID , I dont see any call landing in
asterisk.
In extension.conf (vicidial) file I have
exten => 2062036895 ,1,Ringing()
exten => 2062036895 ,2,Wait(1)
exten => 2062036895 ,3,Answer()
exten => 2062036895
2009 Jan 26
7
Auto Detect
Which command to run which will auto detect all hardwares present in the
system ?
OS : CentOS
Running Asterisk
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090126/5e064cf8/attachment.htm
2009 Jun 22
6
Learn Asterisk
What the best website and book to start learning asterisk ?
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090622/aabe17b8/attachment.htm
2009 Jan 22
7
Root Password not taking
In one of my center , its not taking root password.
Anyways to recover it ?
In other terms , I lost the control of server.
Any solution or re-installation is the only way left ?
I am using CentOS.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090122/ef95ad6e/attachment.htm
2009 Jan 15
2
Dropping this SIP message, it's incomplete
I am getting this Error on my Asterisk.
How to solve it ?
"ERROR[2654]: chan_sip.c:11355 handle_request: Missing Cseq. Dropping this
SIP message, it's incomplete."
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090115/cb953962/attachment.htm
2009 Mar 07
0
Busy Here
I use Xlite and Asterisk.
Now, everything was working fine till yesterday.
But when my agent tried to login to asterisk through xlite, I see below line
sin CLI :
== Manager 'sendcron' logged on from 127.0.0.1
-- Got SIP response 486 "Busy Here" back from 192.168.0.17
> Channel SIP/cc101-08969e60 was never answered.
== Manager 'sendcron' logged off from
2009 Aug 31
2
List Access
To view the post and reply , I always to use below link..
http://lists.digium.com/pipermail/asterisk-users/2009-August/thread.html<http://lists.digium.com/pipermail/asterisk-users/2009-February/thread.html>
Any better way to access the forum ?
-------------- next part --------------
An HTML attachment was scrubbed...
URL: