similar to: Report

Displaying 20 results from an estimated 11000 matches similar to: "Report"

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 May 17
2
Calls Declined
All my calls are getting DECLINED when I am trying from xlite : CLI shows : May 18 00:00:32 WARNING[4617]: channel.c:2781 ast_channel_make_compatible: No pa th to translate from SIP/cc101-b790c1d8(4) to SIP/sip19-090e87d8(256) May 18 00:00:32 WARNING[4617]: app_dial.c:1628 dial_exec_full: Had to drop call because I couldn't make SIP/cc101-b790c1d8 compatible with
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 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 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:
2009 Sep 09
2
Call getting stucked !!
I am using asterisk. I also have an access to VOIPSwitch ver 2 where I can see live calls. Many times I have seen that my calls are getting strucked and then it gets disconneected after 59 mins ( as settings are done accordingly in VOIPSwitch) What could be the reason ? -------------- next part -------------- An HTML attachment was scrubbed... URL:
2009 Apr 16
2
Simultaneous Calls at a time
Double , Triple and sometime 5 calls<http://accessanywebsite.com/search.php?u=Oi8vd3d3LmVmbG8ubmV0L1ZJQ0lESUFMZm9ydW0vdmlld3RvcGljLnBocD90PTc1MzQmc3RhcnQ9MCZwb3N0ZGF5cz0wJnBvc3RvcmRlcj1hc2MmaGlnaGxpZ2h0PQ%3D%3D&b=2> Many time we face an issue where even if an agent is on Call, another call comes in. Sometimes, even if agent hang up the call, call stays back and another come sin and
2009 Feb 07
3
VPN and Asterisk
One of my user was asking, can he use VPN to access asterisk ? What does it mean ? And its possible ? How ?VPN -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090207/325c0670/attachment.htm
2009 Apr 09
2
DTMF
[image: Post]<http://accessanywebsite.com/search.php?u=Oi8vd3d3LmVmbG8ubmV0L1ZJQ0lESUFMZm9ydW0vdmlld3RvcGljLnBocD9wPTI4NjU%3D&b=2#28652>Posted: Thu Apr 09, 2009 8:34 pm Post subject: DTMF and IVR ... Sorry but URGENT[image: Reply with quote]<http://accessanywebsite.com/search.php?u=Oi8vd3d3LmVmbG8ubmV0L1ZJQ0lESUFMZm9ydW0vcG9zdGluZy5waHA%2FbW9kZT1xdW90ZSZwPTI4NjUy&b=2>