similar to: Asterisk and Credit Card Machines

Just wondering if there is a reason PPP support is compiled into zapata by *default*: # Uncomment for Generic PPP support (i.e. ZapRAS) # KFLAGS+=-DCONFIG_ZAPATA_PPP Especially since the comments imply that it should be commented out by default... The main reason I ask is because I usually try to re-compile the kernel to only include the bits that I need, and so I don't include PPP...

you can do that. But are u installing qmail and * on same box. i wont recommend that. i use qmail and *. qmail is strictly for internet email. * is on separate server not exposed to Internet. * box also has sendmail. i hv configured sendmail to use smart host (qmail server). This way its safe and secure. HTH, -B ----- Original Message ----- From: "Ing Isianto Istiadi"

Does anyone have success in upgrading Aastra/Sayson 480i CT firmware? All I get, no matter what I've tried is "Unable to upgrade firmware". tftpd is working because the dialplan freshens, and I have aastra.cfg whatevermacaddressfile.cfg and firmware.st in /tftpboot Am I missing something stupid? Is there another way to upgrade it? Chris Coulthurst chris@shuksan.com

Does anyone have a quick-n-dirty context to implement *66 automatic callbacks? I have a few people who like to have no call waiting on their phone (can you really blame them?) It would be nice to have something like *66, and also like 'Camp On', but instead of waiting something like 30 seconds, monitor the channel until it becomes available, then immediately ring back your phone to

Dear All, I have been running an Asterisk 0.7.1 (patched with various agent applications) server for almost 2 years. We have a data center in the USA and a call center in the UK. All calls are routed to a group of central call queues in the USA. Agents from the data center, call center and from remote locations (London, Scotland, LA, Florida, and Maine) can log in, join the call queue and pick

Does someone have a link to step-by-step instructions to making the Line-In on the console sound card a MOH source? I know this has to work somehow. Chris Coulthurst <mailto:chris@shuksan.com> chris@shuksan.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050512/4a3c3025/attachment.htm

Is there a way to make an outside call hear "The person at phone number XXXX is unavail", but when an internal extension calls another extension, they hear "The person at extension number XXXX is unavail"? I swear I've read this somewhere before but I'm not typing in the right search. I probably found it before by complete accident. Of course, we want the outside

Can anyone explain the Polycom Text Messaging features built in to the IP 500/600? Can Asterisk (or something else) talk to it? I've seen vague references to MSN Messenger, and somehow that's mentally disturbing. Chris Coulthurst <mailto:chris@shuksan.com> chris@shuksan.com -------------- next part -------------- An HTML attachment was scrubbed... URL:

Get the new firmware - it's supposed to have changed the callerid display presentation to include name and number. _____ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Chris Coulthurst Sent: Friday, May 20, 2005 1:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Displayed CallerID on

Anyone else unable to get to www.voip-info.org? Site is returning 'connection refused' here. Chris Coulthurst chris@shuksan.com

I've got a nagios plugin making sure the * box is up, but I would like to do more than that. I need to make sure the PRIs connected to my box stay up and I need to make sure calls are not failing for any reason. Are there any * monitoring packages like this? -Daniel

Being one the many people trying to track down echo 'ghosts' I ran across this page: http://lists.digium.com/pipermail/asterisk-users/2004-November/071301.html suggesting ways to adjust the gain. I have a TDM400P 2x2 config with Kewlstart lines configured. I've located a local telco milliwatt test line, and when I call it, the gain numbers are no where near 14844. Now, this

Ever since I upgraded my Polycom 500 to the newest sip.ld (kept the old bootrom), when I dial things like "*98" for voicemail, the screen shows "9*8" and doesn't dial my voicemail system! Is this user error, or errors in the new firmware? Chris Coulthurst chris@shuksan.com -------------- next part -------------- An HTML attachment was scrubbed... URL:

When I try and park a call by transferring to 700, it audibly says to me "701", and then instead of hanging up with me, it puts me on hold. The only way to park the call is to send it blind to 700, but then I wouldn't know which parking spot the call is in! Before I send any .conf files to the list, does anyone recognize this behavior, and have a workaround? Chris Coulthurst

I swear I read somewhere on one of the MANY pages that there is a script out there that can read the extensions.conf file and create the MySQL DB records on the fly. Anyone know where I look for such a thing? Sure speeds up migration! Chris Coulthurst chris@shuksan.com

Lately when I issue a 'reload' from the CLI, I find that it will sometimes hang forever, completely locked up. I can press enter and see the CLI prompt move, but no commands are taken. "top" shows asterisk eating everything up: PID USER PRI NI SIZE RSS SHARE STAT %CPU %MEM TIME CPU COMMAND 20669 root 25 0 10068 9.8M 5392 R 88.4 1.9 1:02 0 asterisk

I have 3 analog trunks zap/1, zap/4 and zap/5. zap/5 is the least used line. Would the following work for 911 calls? [e911] exten => 911,1,ChanIsAvail(Zap/1) exten => 911,2,Dial(Zap/1/911) exten => 911,3,Hangup() exten => 911,102,ChanIsAvail(Zap/4) exten => 911,103,Dial(Zap/4/911) exten => 911,104,Hangup() exten => 911,203,ChanIsAvail(Zap/5) exten =>

Seems to have struck a small problem.. Using a t100p & Zhone channel bank.... one extension ringing another.....the following will appear WARNING[18448]: File chan_zap.c, Line 2685 (zt_handle_event): Didn't finish Caller-ID spill. Cancelling. if we are using defaultzone=au change it to us and the problem goes away..... any possible solutions ?? Gary .

Hello! I am trying to set up multi-link PPP using two T100P cards in one machine, and 1 T405P card (the 4-port one) in another machine. I have previously been able to get PPP working between the two T100P cards in separate machines.... The 4-port card seems to be my problem currently. I am trying to use the tor2 driver (from a fresh CVS download this morning). When I load the driver (or run

Hi all, I have call recording enabled via the Monitor command and it seems, the call stops being recorded after the call is transferred. Is this normal behavior? If so how can I continue recording of calls after they have been transferred....