search for: grunky

...39;d like to configure this so if I deploy this at a customer site it says "caller id unavialable". With the spelling done right.... Any ideas on how this wold be accomplished? -- Rock River Internet Roger Grunkemeyer 202 W. State St, 8th Floor grunky@rockriver.net Rockford, IL 61101 815-968-9888 x102

...lls w/ 9+7 digits and LD calls, 9+1+areacode+number. How would you tell the PBX try an extension once and 3 digits have been pressed. The exception being 9 as that gives a outside line. -- Rock River Internet Roger Grunkemeyer 202 W. State St, 8th Floor grunky@rockriver.net Rockford, IL 61101 815-968-9888 x101

...#39;m not finding a value that I can enter that would shorten this time. I'd like to have a pattern match in say 5 seconds as opposed to 10. Any ideas on how I can accomplish this? -- Rock River Internet Roger Grunkemeyer 202 W. State St, 8th Floor grunky@rockriver.net Rockford, IL 61101 815-968-9888 x102

...=> 107,102,Voicemail(b102@pstn) exten => 107,103,Hangup I'm just wondering if I could get all this in one line. Would dialing via IAX2 help rather then through the zaptel lines? -- Rock River Internet Roger Grunkemeyer 202 W. State St, 8th Floor grunky@rockriver.net Rockford, IL 61101 815-968-9888 x102

...an> This standard dialplan and files for the ftp server was grabbed off this page http://www.freedomphones.net/polycom/files/ Hopefully this message will help someone down the road. -- Rock River Internet Roger Grunkemeyer 202 W. State St, 8th Floor grunky@rockriver.net Rockford, IL 61101 815-968-9888 x102

...\; -exec echo Deleted {} \;'); system('find '.$dir.'/'.$context.'/*/Old -name msg????.??? -mtime +'.$ageold.' -exec rm {} \; -exec echo Deleted {} \;'); -- Rock River Internet Roger Grunkemeyer 202 W. State St, 8th Floor grunky@rockriver.net Rockford, IL 61101 815-968-9888 x102

I feel I'm on the verge of setting up a pbx for handling internal calls only... The last problem - I think - I've run into is w/ the phone registration running asterisk -vvvc I get a bunch of messages looking like so Apr 6 14:46:05 NOTICE[1116957488]: chan_sip.c:5623 handle_request: Registration from 'sip:2001@192.168.22.254' failed for '192.168.22.1' Apr 6

Hi there, Here at my company we are willing to use the asterisk IVR system. The problem we are having rigth now is that all our GWs use G729. I've read that in order to asterisk be able to make transcoding from the GSM audio files to G.729, it is necesary to purchase a license from digium. Is this correct? I've seen that licenses are purchased on a per-channel basis. Could

...when it comes to registering this puppy. I used the web interface to specify the username/password but still nothing. Any ideas or docs I could look at to get this Polycom phone setup? -- Rock River Internet Roger Grunkemeyer 202 W. State St, 8th Floor grunky@rockriver.net Rockford, IL 61101 815-968-9888 x101

Outside of Asterisk - is their anything a linux admin can do to optimize or speed network traffic to/from the pbx to sip phones? I'm looking for some options in /proc Since Asterisks is a network bound/sensetive app I'd start their. I'm running RH9. Optimizations for other linux and unix distro welcome aswell.

Does anyone know how to send a message to a Cisco 7940/7960 phone running SIP images 6.3 telling it to light up one of its LED's when new voice mail arrives? I found alot of web based solutions http://www.voip-info.org/wiki-Asterisk+GUI and easy ways of getting email or getting paged of a new voice mail - but nothing where you can just look at the phone and see a blinking light or

I have a few SIP phones, Cisco 7960s, and was looking into implementing some compression, ala G.729. I'm looking into purchasing a g729 licenses just to get an idea of performance and voice quality, over lans, wireless and single channel isdn. Does anyone have positive/negative experience w/ getting licenses/support from Digium? Hows the sound quality compared w/ g.711? Is 729 better

I'm starting out w/ a Cisco 7940, running the Sip image version 6.3. I've downloaded/installed asterisk via cvs. I've set the phone up to get its info via dhcp - the dhcp, tftp, astericks box & phone are on the same network. I've gone through and setup a test account per the instructions @ http://voip-info.org/wiki-Asterisk+phone+cisco+79xx but time I do a sip show

I have a setup of 3 Cisco 7940 running Sip image 6.3. All these phone are registered by the below information *CLI> sip show peers Name/username Host Mask Port Status 2002/2002 192.168.22.199 (D) 255.255.255.255 5060 Unmonitored 2001/2001 192.168.22.200 (D) 255.255.255.255 5060 Unmonitored 2000/2000 192.168.22.198 (D)