search for: coulthurst

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: http://lists.digium.com/pipermail/asterisk-users/attachments/20050518/06a4baea/attachment.htm

...T 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050912/a31a082a/attachment.htm

...*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 initiate the call. Ok, I redundantly described what it does. Anyone have a way to make it happen? Thanks much... Chris Coulthurst chris@shuksan.com

...###############* Rx: 4452 ( 4452) Tx: 46 ( 63) Is this good? Normal? Any suggestions, or a point in the right direction for the right documentation would be appreciated. P.S. This is CVS-HEAD Zaptel on a P3 550, Host bridge: Intel Corp. 440BX/ZX/DX - 82443BX/ZX/DX Host bridge (rev 3) Chris Coulthurst chris@shuksan.com

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

...XXX 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 caller to hear a phone number seven digits long, while an extension hears just that, an extension. 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/ccede439/attachment.htm

...al(Zap/4/911) exten => 911,104,Hangup() exten => 911,203,ChanIsAvail(Zap/5) exten => 911,204,Dial(Zap/5/911) exten => 911,205,Hangup() exten => 911,304,SoftHangup(Zap/5-1) exten => 911,305,Wait(2) exten => 911,306,Goto(204) Did I get the Priority + 101 idea right here? Chris Coulthurst chris@shuksan.com

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: http://lists.digium.com/pipermail/asterisk-users/attachments/20050527/7cac5743/attachment.htm

...uot;, 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 chris@shuksan.com

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

...0 SWN 0.0 0.0 0:00 0 ksoftirqd/0 Most recent add-ons have been Speex and h323. I just installed h323 today and this has been going on for about a week, so I know its not that, but I can't remember if this was happening before Speex or not. Anyone have any similar happenings? Chris Coulthurst chris@shuksan.com

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

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

...was connected on LAN. Timing is an issue, if you doing everything on LAN it should not be a problem. Just make sure you use G.711 protocol. > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Chris Coulthurst > Sent: Tuesday, 17 May 2005 19:12 > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > Subject: [Asterisk-Users] Asterisk and Credit Card Machines > > I am planning to deploy an Asterisk server at a local restaurant and was > thinking: I hear a lot of troub...

...one has some positive results with a similar competitor, I'd love to hear about it. In the meantime, I have to change back over to PSTN lines temporarily, since I can't rely on service from Teliax. I hope any/all of you that use their service have better luck than I have with them. Chris Coulthurst chris@shuksan.com

Hi people, what is the current status of send/receive fax on asterisk extensions, i dont want to receive the fax and send an email or viceversa, i want to connect a standard fax machine to a Linksys' ATA (FXS RJ11 port) . Webdoc?, pointers? Thanks

Hi, I have lookedaround and don't see this anywhere. Is there a way to tell the ip500 to not make the aural MWI blips?