similar to: Asterisk-CVS-HEAD locks up on 'reload' from CLI (sometimes)

Hello All, Wondering what sort of real world mileage people are getting out of different internet connecions - i.e. different DSL connection speeds, cable modems, etc... Is it reasonable to hope to carry 10 - 15 concurrent calls on a 768K DSL? I'm not talking about theoretical BW or looking for any difinitive absolute guarantee... With DSL and Cable - there is no guarantee, so I'm

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

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:

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

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

Hi list, Time to time, my asterisk goes down.Verifying with TOP, I see the swap memory of the computer get used totally but, I don't see what the process is using it. Hereis a copy wath I see doing top. Does somebody have an idea ? My asterisk version is ====>>> Asterisk CVS-HEAD-08/18/04-22:30:24 Thanks Angel. 08:49:19 up 5:23, 1 user, load average: 0.50, 0.70, 0.64 35

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

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

I am getting an odd situation when backing up a number of ext3 filesystems and was wondering if it could be caused by journalling. Over the space of a minute the load average will jump from 2 to over 40 and the system will be unresponsive for anywhere from 8 to 25 minutes. I am going to be trying a number of things, but was wondering if anyone could see the reason for the high load given the

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 =>

I'm having a problem with the callerID that the polycom IP600 phones are displaying. I would like to modify the CIDName and leave CIDNumber as exactly what the phone call came in as(provided they aren't hiding callerID). Most of the calls will be going to the queue, but a few will go directly to the SIP phones. I've done a various combinations of using SetCallerID(),

Does anybody know if there are any external telephone-keypads for sale anywhere? (containing the keys 0-9, *, # and onhook/offhook would do) I am looking for a keypad to control a softphone and would prefer the controls to be in the physical world instead of as a window. Sincerely, Markus Hakansson

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

When I want to pick up a ringing line, I dial *8 and hit New Call softkey on my Poly 501. For some reason, if I pick up the hand set and dial *8, it seems to ignore or drop the 8 digit. I've confirmed that this happens with all of my 12 Polycom 501s. Does anyone know what would cause this or how to fix it? Craig Bruenderman Network Advocates, Inc. 300 Envoy Circle Suite 300 Louisville, KY

Has anybody tried to use a Polycom phone (I have 500s and 600s) with a netmask shorter than /24? (A network bigger than 255.255.255.0). We've run out of IPs in our initial /24 network, and I'd like to expand it to 255.255.248.0. When I set it to 255.255.248.0 I can ping the phone while the bootloader has control. As soon as the SIP application starts, I stop getting ping responses. Phone

I really hate to have to make a post like this, but I feel I have little choice but to relay to the group my experience with Teliax, and explain why I recommend against using them as a primary Voip-> PSTN provider. I hope that a letter like this will inspire companies like Teliax to work harder at customer service, as well as circuit stability. We need more companies that offer the types of

(Sent to linux-raid, linux-kernel and ext3-users since I'm not sure what type of issue this is) I've got a test system here running Redhat 7.1 + stock 2.4.9 with these patches: http://www.fys.uio.no/~trondmy/src/2.4.9/linux-2.4.9-NFS_ALL.dif http://www.zip.com.au/~akpm/ext3-2.4-0.9.6-249.gz http://domsch.com/linux/aacraid/linux-2.4.9-aacraid-20010816.patch All three patches applied