search for: rockriver

...ke 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

...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

...ot 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

...07,102,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

I'm running SIP between my Lucent TNT acting as a gateway, and an asterisk server. We have a PRI coming into the Lucent. Basically the problem I'm having is mostly on inbound calls but some outbound calls as well. I hear echo and sometimes some weird artifacting on calls coming in from the lucent. Everything routed over IAX to VoIP Jet or Nufone sounds fine. It seems like every 3

...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

...c 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 copied a config that was posted to the list but it didn't seem to work. Any help would be appreciated. Jeremiah -- ______________________________________________________________ Rock River Internet Jeremiah Millay 202 W. State St, 8th Floor jeremiah@rockriver.net Rockford, IL 61101 815-968-9888 Ext. 2202 USA fax 968-6888

So, We want to be able to put a fax machine on the line port of the X100P in our asterisk server. We however also want to use this card for 911 calling. We need some sort of mechanisim to "disable" the line out port on the x100p by software to "interrupt a call" on the line. Anyone done anything like this?

Trying to use: Asterisk 1.4.2 Zaptel 1.4.0 chan_zap won't compile in asterisk 1.4.2 when used with zaptel 1.4.0. The changelog has this entry: * channels/chan_zap.c, configure, configure.ac: If we receive ZT_EVENT_REMOVED, destroy the specified channel. (issue #7256, tzafrir) Also, update the configure script to make sure that we don't try to build chan_zap

I'm looking for someone familiar with setting up some of the more advanced features of the Lucent TNT, preferably someone with knowledge of Trunk Groups and choosing outgoing PRI channels based on call type and perhaps NPA-NXX We currently have 8 PRI's. 7 of them are for our dialup pool, the 8th is for our voip. We currently run the dialup PRI's to a seperate TNT We want to

...dirintro = "dir-intro-fnln"; else dirintro = "dir-intro-fn"; } -- ______________________________________________________________ Rock River Internet Jeremiah Millay 202 W. State St, 8th Floor jeremiah@rockriver.net Rockford, IL 61101 815-968-9888 Ext. 2202 USA fax 968-6888

I currently have two Cisco phones, a 7960 and 7970. The 7960 has a SIP OS on it and the 7970 has a SCCP. When the 7960 powers up it loads OS79XX.TXT, SIPDefault.cnf, SIP000E3875266C.cnf, RINGLIST.DAT, and dialplan.xml. I have a Cisco SmartNet agreement with the phone so I have access to download the firmware. I recently purchased a Cisco 7970 phone and was in the process of configuring

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

...t 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

One of my friends is facing this problems and I could not find any solution to that. Hence this post. In her Asterisk PBX, she has programmed about 10 agents, and strategy is rrmemory. Everything works fine. When an agent has received an ACD call, another call is not presented to him as long as he is on the ACD call. However when an agent has made an outgoing call, he is still presented another

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

Has anyone else had problems with the Page() application not working under Asterisk 1.2.3? We use Cisco 7960 phones and set one of the lines to auto answer. When someone dials the paging extension it calls the page app and invites all the lines on the phones that are set to auto answer into a meetme conference where all the members are muted except the original caller. When I try to use the