similar to: Vicibox vs VicidialNow

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

~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ 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

Dear all, I need to build a PBX based on Asterisk for a call center. I have worked with raw Asterisk but it's hard to work for big implementations think. Also I have worked with Trixbox CE for a small bussines and it was prette good, but I have not have many features like ACD. I know there is another version called Trixbox PRO -specially Call Center edition- that's not free but has got

Hi, I have 4 bind9 dns installed on centos 4. My primary dns server went down and all of my domains doesn't resolve even if the 3 slave dns is up and running. Im not sure where to configure this is it in my domain registration or in bind? -- Regards, Mark Quitoriano Blog | http://mark.quitoriano.org VicidialNOW! | http://www.vicidialnow.com APUG! | http://asterisk.org.ph --------------

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

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

Dear all, I have to install an E1 card in my Asterisk 1.4.23 server and here is my short question: Is it necessary to install or update any Asterisk/Zaptel/Any extra module or the default installation is good enough to just plug and run the E1 card ???? Thanks a lot Alejandro

Hi, I need to configure BRI 4span card in dubai in vicidialnow for dialer perpose. in that i have small confusion which is NT an TE mode . that was i am setting perfectly but dubai telco what they are use for this i dont know which parameters are use for that . please help me. -- Best Regards, Mahesh Katta *BUZZ**WORKS* Business Services Private Limited BANGALORE | CHENNAI | HYDERABAD |

I just want to make a SIP call from 192.168.1.3 to 192.168.1.4; or not even a call. Ring? Beep? Ping? Some sort of "hello world" connection. 192.168.1.1 netgear router 192.168.1.2 asterisk (vicidial) 192.168.1.3 ubuntu client 192.168.1.4 mac OSX client (not shown) Do I have a firewall problem which would impact a soft phone from establishing a connection?

I have Cent OS 5.1 I also have http://phpsysinfo.sourceforge.net/ I have asterisk running . Now when I look at System Information , I see that "Physical Memory" keep increasing and at one point it reaches 96%. Then my sever get hang and then I have to restart it. I have 4 GB RAM. Processors 2 Model Intel(R) Core(TM)2 Duo CPU E7200 @ 2.53GHz CPU Speed 2.53 GHz Cache Size 3.00 MB

Dear All, ? ???????? I am trying to using E1 PRI Connection with vicidialnow setup..Calls are landed in asterisk .From Asterisk?CLI ?i can see the caller?id from where the call came ..but the calls are rejects. ? ?????? So what should i do now for forwarding this call to a agent who?is using softphone.Is there any Dialplanning required? ? Plz Help me to sort it out.? ---tareq

Hello, We've released another update to our VICIDIAL/astGUIclient call center suite: 2.0.5 http://astguiclient.sf.net/ The call center suite client applications run on most modern web browsers on almost any GUI-capable operating system, and it includes the VICIDIAL call center suite. This package is free and AGPLv2. This package is geared towards Asterisk installations with SIP,IAX or Zap

== Manager 'sendcron' logged off from 127.0.0.1 vicidialnow*CLI> dial 919545090201 -- Executing AGI("OSS/dsp", "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("OSS/dsp", "SIP/19545090201 at sip203||tTor") in new stack -- Called 19545090201 at sip203 Feb 2 13:36:38

Dear, I have asterisk 1.6.2.13 in a small hardware PBX (512 GB RAM and Celeron), and last days when I call from one extension to another of the same PBX after I dial the number the rings sound after 20 seconds. In the CLI log, when I debug the AGI, I see always goes good until dialparties.agi, and after that there are 20 seconds without any log, and so the ring sound. I've read

Dear, I always had one E1 card with one span, so I've never had any problem in set it up through /etc/dahdi/sustem.conf and /etc/asterisk/chan_dahdi.conf because I put span=1. But now I have a PBX with two E1 cards with 4 span (8 span in total). How do I have to define both card in system.conf and chan_dahdi.conf, and how do I have to refer each span to the corresponding card ??? Thanks a

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

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

Dear all, I have an Asterisk SIP server in a LAN environment and I want your opinion in order to decide the use of an audio codec: What audio codec is better, G.711a or G.711u ??? Which suites to my LAN voip calls ??? Thank you !!! Alejandro -------------- next part -------------- An HTML attachment was scrubbed... URL:

Dear, I need to create a custom device extension in order to dial a local queue. Suppose my queue number is 8888, how can fill the Dial field from the custom extension ??? Because if I put just 8888 or Local/8888, I don't succeed. Thanks a lot

HI, Does the mysql in centos compiled with multi processor support? i have a mysql in centos 5 running in a core2duo processor and it seems that it only uses 1 processor. -- Regards, Mark Quitoriano Blog | http://mark.quitoriano.org VicidialNOW! | http://www.vicidialnow.com APUG! | http://asterisk.org.ph