similar to: how to tell

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

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

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! How come that the logfiles in my /var/log/samba looks like this: -rw-r--r-- 1 root root 0 Feb 24 04:02 log.data -rw-r--r-- 1 root root 3223 Jul 2 18:04 log.dell -rw-r--r-- 1 root root 0 Jun 22 04:29 log.dell.2 -rw-r--r-- 1 root root 0 Jun 1 04:02 log.dell.3 -rw-r--r-- 1 root root 0 Jul 3 04:09 log.gggg

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

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...

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'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

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 =>

Hello I was just stuck around as to how I configure my Asterisk to access extensions from Mysql. I have made all the necessary changes in the extconfig.conf, the extensions.conf, res_mysql.conf, res_config_odbc.conf,res_odbc.conf as they have mentioned on the site www.voip-info.org <http://www.voip-info.org/> . But still I am getting the error as May 13

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

I'm attempting to set up my SoundPoint 501 with my Asterisk server. I've configured DHCP and TFTP and successfully updated both the BootRom and SIP application. I've also created a custom cfg file for this phone's MAC address and the settings seem to be taking just fine. I can see that the phone registers with my Asterisk server but 'sip show peers' reports that the 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

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

Hi, I'm trying to write a tool, which shows me the state of the current calls. For this purpose I'm reading from Pipe the Asterisk output and parse it... asterisk -vr | mytool However, the problem ist how to get the information about who got this call in the group. The Zap channels are assigned dynamical. Only thing I can see which channel is connect to the caller but not who is

Greetings, I have Asterisk@Home 1.1 with an X100P, which is my only outgoing line. Call waiting will not work when an extension is dialed out on that line. The only way it will work is if an extension has dialed another, and someone calls in on the ZAP line. Then, the flash key will switch between the other extension and the ZAP line. I cannot get my SAP-2002 to tell the ZAP to flash