search for: cameo

Hello Everyone, We are looking for a simple open source auto dialer with "polling" capabilities. What we would like is a program that we can upload leads to, and have asterisk: i) Dial numbers ii) Play pre-recorded iii) If user presses one, forward the call to an agent There are so many solutions out there it's hard to make a decision on what works, what has just a limited free

Hello Everyone, Just getting in a new cisco router, and would really like to get it up and running as soon as possible. Everything is configured from what we can see. This is a NAT setup. After 2 seconds on a successfully established call we reach retrans max, and asterisk disconnects the call. We have no idea why this is happening. SIP and RTP is flowing as expected. Your help is greatly

...f you received a blank email. We re-sent the email to all of you again this morning. So we have decided to have two winners, one for last evening and one for today's mailer. We are very sorry for the problems; the new mailing system isn't perfect. Trivia Question: What movie star made a cameo appearance on the first Late Show with David Letterman, on CBS? Answer: Paul Newman, who was seated in the audience, stood up and said, "Where the hell are the dancin' cats?" Winners: William U John H Thank you for your continued support, Greg Wolters Sun Sales Manager Paladian...

Hello Everyone, Our switch is sending P-Asserted info to asterisk however the information is getting removed when asterisk forks the call. Is it possible to have asterisk retain the P-Asserted on the leg. This is quite important for valid functionality of our network. Tried setting `sendrpid = yes` and still same problem. We really don't want to have to `SipAddHeader` as it is already being

Hello Everyone. Our environment is a register free setup, and our phones are set as host=dynamic. The problem we are experiencing is for inbound calls: Name/username Host Dyn Forcerport ACL Port Status Realtime 222/222 (Unspecified) D N A 0 Unmonitored Cached RT So when we DIAL 222 we get: WARNING[23103]: app_dial.c:2198 dial_exec_full:

Long story... Would be nice if we can remove this on BYEs X-Asterisk-HangupCause: Normal Clearing. X-Asterisk-HangupCauseCode: 16. Kind Regards, Nick. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140723/09e97fd1/attachment.html>

Hello Everyone, Today we observed asterisk sending two invites for the initial call before the call was established (ie, not re-invites). There were no changes made to the configuration for a very long time, and was kind of confused when seeing this action. Can someone please suggest where to look to remove this behaviour? U 2014/08/12 07:34:20.405029 192.168.2.10:5060 -> 192.168.2.20:5080

Hi, I really did my best to understand how I can help improving Wine, but I did not succed. Well, everything started when I installed a win application on my Debian 3.1, Wine 2004.07.16 deb (clean install, no dlls). The application is ALOHA (http://www.epa.gov/ceppo/cameo/aloha.htm) It's a free application from US-Environment Protection Agency used for chemical emergency planning (I am an officer firefighter). On Wine it has a problem that prevents its wide use. After having entered all the needed data (or loading the included prova.alo example in "plann...

hi, Now I have tow computers, one is RH9 linux , the other is windows = 2000. I have configed samba 3.0.7 running on linux, so windows could = share linux computer's files. But there's a problem : when windows using network neighborhood connect = linux samba server, I login into a folder using account and password = for this folder, windows will remember this account and password,

Hello Everyone, I have a new problem where when placing the call, asterisk will automatically go into music on hold until the call is connected (ie, no ringing). It was kind of confusing because sometimes `SESSION PROGRESS` takes longer than others, during this time we are in MOH. The call does eventually connect and the MOH stops. When debugging I saw the following debug message: [Sep 10

Hello Everyone, I am using the following dialplan to allow users to check their messages from PSTN world: ; Internal Routing exten => _1XX,1,Dial(SIP/${EXTEN}, 10) exten => _1XX,n,Wait(1) exten => _1XX,n,Answer exten => _1XX,n,Wait(1) exten => _1XX,n,Voicemail(${EXTEN},us) exten => _1XX,n,Hangup The problem is that when the user presses `*#` to check his/her messages, it adds

Hello Everyone, I was just wondering if the cli command "file convert" supports wildcard or entire directories? I am looking at a very long list right now and anxiously waiting a response :). Kind Regards, Nick from Toronto. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello Everyone, I have a strange issue where the caller's phone keeps ringing even after the 200 OK. I am using the latest version of Asterisk 1.8, and wanted to know if anyone could give me any pointers before posting the SIP debug messages. Kind Regards, Nick from Toronto.

Hello Everyone, Calls that are private name private number have the following TO header: From: "Unavailable" <sip:asterisk at server.com>;tag=as120a1079. Don't tell anyone, but we are trying to put on a "We're big enough to own the pricey softswitch" look. Even though I would pick a OpenSIPS + Asterisk combo over a Metaswitch any day. Three words "Service

Hello Everyone, We are seeing many instances where we receive 200OK from the vendors however, asterisk still keeps ringing. Is there anyway to stop this from happening? I remember reading something about early media however this seems to be a case of late media? Kind Regards, Nick from Toronto. -------------- next part -------------- An HTML attachment was scrubbed... URL: