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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

...e '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

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

...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

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

...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

...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

...en => 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

...the number of the caller as well as the name, but so far it just says the name! I can go into the 'received calls' log and look at the info screen to see the number, but that doesn't sound very practical. Any help would be greatly appreciated. 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/20050520/57c32ec1/attachment.htm

...want to see the number of the caller as well as the name, but so far it just says the name! I can go into the 'received calls' log and look at the info screen to see the number, but that doesn't sound very practical. Any help would be greatly appreciated. Chris Coulthurst chris@shuksan.com <mailto:chris@shuksan.com> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050520/3787822b/attachment.htm

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

...stead 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

...re has a clue about this, here are a couple of questions: Can you 'hint' Zap and IAX2 extensions? Can you concatenate extensions together? (i.e. exten => 200,hint,SIP/101&SIP/202&Zap/4) And the big one is..... ...how does this work on a Polycom 500? Chris Coulthurst chris@shuksan.com

Is there a way to check to see if a user has recorded a busy message? If they haven't I would prefer to send them to "u" and play their unavailable greeting. I know I can send them to "u" in both cases, but I would like to send them to "b" if a recorded busy greeting exists. Thanks! -------------- next part -------------- An HTML attachment was scrubbed...

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

How do I tell if Asterisk is receiving a flash hook command? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050612/e0bb14f6/attachment.htm

....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(),

Hi, I have a Asterisk with two Cisco ATA connected with each other . Cisco ATA1 is connected via satellite link Cisco ATA2 is connected via leased line The problem is CISCO ATA2 can call CISCO ATA2 with no problem But CISCO ATA1 cannot call CISCO ATA2 . Any suggestion please Rabii -------------- next part -------------- An HTML attachment was

...tive 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