similar to: Patch 5779 on 1.0.9?

I'd like to be able to use my Snom 360 LEDs to view the status of parking slots, so I'm trying to install the "metermaid" patch (http://bugs.digium.com/view.php?id=5779). Can someone help an svn newbie figure out how to install this patch? I've done the following: svn checkout http://svn.digium.com/svn/asterisk/team/oej/metermaids cd metermaids make clean make but I get

Used the Directory application instead of the Directory AGI. -----Original Message----- From: Colin Anderson [mailto:ColinA@landmarkmasterbuilder.com] Sent: Tuesday, January 23, 2007 11:29 AM To: 'asterisk-users@lists.digium.com' Subject: [asterisk-users] * 1.0.9 Voicemail record name does not playback in Directory() On * 1.0.9 User logs into voicemail, dials Option Zero, then Option

Using 1.0.9: If I have: exten => s,1,Dial(SIP/5555&SIP/12345@192.168.1.1) How can I return the DIALSTATUS variable for the second SIP channel ONLY if the second SIP channel is busy, regardless of the dialstatus of the first SIP channel? What I want is, if the second SIP channel is busy go to n+1 or n+101 regardless of the status of the first SIP channel. tia

Hi all, I'm using 1.2.9.1, with the "metermaid" patches to show parking spot status on Snom BLF lights. I see from http://www.asterisk.org/node/97 that the metermaid code has changed substantially since 1.2.9.1. Is anyone successfully using the "new" metermaid functionality in 1.4.x? I'd like to hear any good/bad experiences before I attempt even a test

Disclaimer: Not a troll I'm curious as to this obsession with uptime is. All of the posts of this type are along the lines of "After X days, Y thing does not work but if I reload or reboot, it's OK" - so why not cron a reboot? Is it considered bad form or something like that? I reboot every night whether it is needed or not, not afraid to admit it, and everything works fine for

We are running our * server as a virtual PBX for 6 companies. I am having all of the Allison prompts plus our own custom IVR prompts being re-recorded for each company, in a different voice (marketing thing) with a different personality (perky, corporate, earthy) . I'm curious if someone could point out a dirty trick to get the voice to play right, for internal and external callers,

>I can not even get IP anymore from my DHCP Hate to ask the obvious, but is the DHCP server on the same VLAN?

I know this is not strictly an asterisk issue but it is related I guess. Just to let you know that after many calls to Mitel the consensus is that they will be releasing a new version of the 5220 that is dual boot (minet and SIP) next week or the week after. This firmware will only appear on NEW phones manufactured after the release date (no one could confirm but the 23rd of sept was

I'm using a Snom 320 in a CAP position and the receptionist wants to do blind transfers. OK, no problem so far. Now she has asked me how to UN-transfer a call, as in, she transfers a call and wants to hook the call back before it connects (she wanted to tell the caller additional information for example) I don't think that this is possible as once my dialplan starts using Dial()

AHA! # 1 is the case! Seems the user was fooling around with the phone after the firmware upgrade. Shame that that setting couldn't be locked out. Thanks to Mr Tahir and Mr Stredicke for their spot on responses. -----Original Message----- From: Usman Tahir [mailto:Usman.Tahir@snom.de] Sent: Friday, September 23, 2005 12:34 AM To: ColinA@landmarkmasterbuilder.com Cc:

Colin, Thanks for your assistance. Reading over your advice I seem to still be a bit confused. My agents are not on the Asterisk server; it appears in your advice that my the call will travel this path: WWW interface --> agent enters their DID, platform to use, and termination DID --> AST calls agent --> Agent calls termination DID If my agents are not on the Asterisk server

Has anyone ever encountered an echo on the IP phone side of a call? It is an echo of the user's own voice. I believe that no one else in the office is experiencing this problem. The phone itself is a Snom 320. I've asked Snom for assistance since my source no longer carries Snom, but unlike previous times they've been slow to respond. ----- Mike Hammett Intelligent Computing

the user you are connecting as should have full rights to /var/run/asterisk: http://www.voip-info.org/wiki-Asterisk+non-root hth -----Original Message----- From: Erick Perez [mailto:eaperezh@gmail.com] Sent: Monday, April 03, 2006 9:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Unable to connect to remote asterisk (does /var/run/asterisk.ctl

I was always puzzled by posts to the list about people having problems getting hints to work on a Snom, since I always seem to have no problem making it work. That is, until today when I tried to get a sidecar to work. All I could do was get a monitored extension light to light up continuously, regardless of state. Frustrating! Going back to my working dialplans where I got 1 or 2 lights working

In the example I posted previous, there is an obvious gaping security hole, it would be trivial for someone to read the querystring and exploit it to make free phone calls, spoof caller ID (if you allow the CallerID to be set with a QueryString value), etc. You want to make damn sure that the URL is not publicly accessible or somehow obsfucate the querystring, or use POST. In my case, I

Dear Users, Recently I have started using HINT option in Asterisk 1.2.4 with my Polycom 500 phone. What I have notice that for a day or two everything is working great but then HINTs stop working on my Polycom phone. It also happens when I reload asterisk from console. I do sip debug and I do not see anymore asterisk sending NOTIFY messages about my watched extension. To make it work again I

http://www.pcmag.com/article2/0,1759,1812887,00.asp Specifically, his assertion that ISP's would sniff traffic and block, say, the SIP port. You could play wack-a-mole with port numbers, no? Also a community based, Freenet style of encryption implementation for "free" VoIP traffic would address this issue. I raise this to the list because I'm sure there's a grain of

Hi Andrew - > On Monday 13 March 2006 10:20, Noah Miller wrote: >>> The transer button on the polycom phone does not seem to transfer/park >>> the call properly. I have to use the # -> 700 to park the call. >> >> If I recall, using the Polycom transfer, you have to make sure it is done >> as a blind transfer. The Polycom attended transfer

Friends, The developer team for Asterisk not only consists of coders - a very important part are the testers, those that test new code and give feedback. For a few weeks, I've been maintaining a large number of branches with various stuff in them and have gotten very little feedback, not enough to judge whether or not to move forward with these patches. Some, but not all, code is

> They instantly got us to look at the output of zttest and we found that this was (in their words) 'extremely low', with 'best' and > 'worst' readings of 99.975586% and 99.963379% respectively. Might want to give PCI latency setting a try, it helped for me. My ZTTEST would drop occasionally to 99.95% until I set: setpci -v -s 01:01.0 latency_timer=ff