similar to: Simultaneous Calls at a time

[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>

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

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

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

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

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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

Hi all, For those of you people that use Agents (with Agentlogin, not AgentCallbackLogin) on a call center, I have this need: when the agent logs in, a channel keeps running all the time that the agent is logged in to receive the incoming calls. How do I know which agent logged in (code)? Right now, if I query the login channel, there is no information about which agent is logged on: #

Hi everyone, I was just wondering, does anyone managing production asterisk servers migrated successfully from 1.4.2X to 1.6.0.X? I would like to see if there are some successful cases. Is your 1.6.0.X behaving well, with acceptable stability? Please share your experiences. Thanks, -- Ing. Miguel Molina Grupo de Tecnolog?a Millenium Phone Center PBX: (+57 1)6500800 ext. 1201 Fax: (+57

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

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

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 * * * *

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.

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

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

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

After passing authentication, Then with this line, extent => 3616739999,5,DISA(no-password calls-outbound) As soon as the first digit of the intended number to be called is entered, the system does a Hungup 'DAHDI/1-1' It has done that no matter what I have tried. I am missing the boat somewhere. Anyone have tips? Cary Fitch

Humm... should the list would be magic again? I have just intsalled, using the register, benchmark and downloared the correct codec to my asterisk installation, but I don't have the g729 command at my CLI... Any advice... Do I reboot? ;D

Hello, I have a var like ?blabla? with the ? I need to suppr the ? Is it possible with the ${var:x:y} ? Thank you Cordialement, BERGANZ Fran?ois P Pensez ? l'Environnement, n'imprimez ce mail que si n?cessaire. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi guys, I'm trying to write hangup causes from asterisk into the CDR record. Using version 1.4.24.1 at the moment, but no joy so far. Has anyone implemented this?