search for: loligo

...ibes some of the features that are possible to set in the ATA and specifically what needs to be done to get it working with Asterisk. It's not pretty, it's not HTML, but it's a lot of hints that I've collected from the list and other sources over the last year or so: http://www.loligo.com/asterisk/Cisco/ATA-186-guide.v20030628.txt JT

I download app_valetparking.c from http://www.loligo.com/asterisk/misc/apps/app_valetparking.c and followed the directions on http://www.loligo.com/asterisk/misc/apps/app_valetparking.README I am using asterisk-1.0.0 any suggestions [root@localhost asterisk]# astxs -install apps/app_valetparking.c gcc -pipe -Wall -Wstrict-prototypes -Wmissing-pro...

...out to Allison Smith. Please submit reasonably small changes to me by tomorrow 10:00 AM Eastern time, and I'll add them. As usual, donations to what will be a ~$110 USD expense would be appreciated, as I am paying for this round out of my pocket. Please send to paypal address "jtodd@loligo.com". I did not include all possible symbols on a North American keyboard, as it was getting exhausting and possibly silly. calls abandons staffing average Speed of Answer Sorry, but the user's mailbox can't accept more messages. Please enter the conference call number for the conf...

...EXTENSIONS+++++ EXTENSIONS+++++ EXTENSIONS+++++ EXTENSIONS+++++ EXTENSIONS ; Asterisk Management Portal (AMP) ; Copyright (C) 2004 Coalescent Systems Inc ; dialparties.agi (?www.sprackett.com/asterisk/) ; Asterisk::AGI (?asterisk.gnuinter.net/) ; gsm (?www.ibiblio.org/pub/Linux/utils/compre..) ; loligo sounds (?www.loligo.com/asterisk/sounds/) ; mpg123 (?voip-info.org/wiki-Asterisk+config+mus..) ; include extension contexts generated from AMP #include extensions_additional.conf ; Customizations to this dialplan should be made in extensions_custom.conf ; See extensions_custom.conf.sample for an...

Hi, I've had a search through the archives and didn't find much. Is anyone using the Monitor application? I have it working but there is a really big drawback. The files are always called the same thing, which means if I make 2 calls one after the other the first recording is lost. I half expected Monitor to use something like ZAP-2-1-<yyyymmddhhmmss>-in/out.wav for it's

...he past some interest in the Asterisk example files that I have made available from my experimental platform (home office) so I've put my newest versions up on my site. Feel free to dig through these and cut pieces up for your own use, or just marvel at the bad logic flows. :) http://www.loligo.com/asterisk/current/ JT

...rst SIP 802.11 phone. I got my hands on the "first" one sold, so that perhaps makes me the first person to have a real 802.11 SIP phone commercially in the US interworking with Asterisk. Whee! Can someone point me to other commercially shipping phones to prove me wrong? http://www.loligo.com/asterisk/misc/WiSIP/WiSIP-review.html JT

...k (Trixbox 2.2) and my ATA-186 with both ports. I need to be able to call one port from the other-- the idea is to have two phones in two different locations that _can_ call each other. So, in reading the Asterisk Wiki and other sites, the best documentation I found was this: http://www.loligo.com/asterisk/Cisco/ATA-186-guide.v20030628.txt <http://www.loligo.com/asterisk/Cisco/ATA-186-guide.v20030628.txt> "Note that each line must have it's own distinct and complete configuration, and if you use both lines on the ATA-186, it will REGISTER twice. Further note that you c...

...of files, each with 1-10 seconds of silence, and put them in my sounds directory for public consumption. Not a big deal for most of you to create these files yourselves, but perhaps a minor pain that hopefully I've removed for some people who don't have sound tools handy. http://www.loligo.com/asterisk/sounds/silence/ JT

...s based on current published prices on Amazon's system. Feel free to tell me that I'm wrong with these calculations - but be specific if you find any problems, as I suspect others may glom onto these figures as gospel and I'd hate to have the wrong data in there. http://www.loligo.com/asterisk/misc/amazon-ec2.xls The net of my calculations is that a small instance of 20 users in a standard office environment would cost about $75 per month, which when compared to running a server in-house works out to be (raw cost, not including admin time and not discounting out-of-of...

http://www.loligo.com/asterisk/misc/Presentations/Asterisk-features-20040106.xls I had been asked a while ago to put together a short Excel spreadsheet listing many of the "common" features of Asterisk as compared to a typical PBX. Many PBX vendors supply an exhaustive list of their features, and I fi...

Does anyone know of a AGI script that takes advantage of the weather sound files that's included with the extra sound files available from www.loligo.com/asterisk/sounds/ <http://www.loligo.com/asterisk/sounds/> ? Thank, Jeramie -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050215/263c4d50/attachment.htm

How many times do you ask for help here before getting a respone? Every single thing I do No matter what I get busy extensions. I am willing to pay someone to help here. Anybody got a clue? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050802/d0d1326c/attachment.htm

...waiting) exten => *71,5,Playback(de-activated) exten => *71,6,Macro(hangupcall) [app-callforward] ; dialed call forward app - forwards calling extension exten => _*72.,1,DBput(CF/${CALLERIDNUM}=${EXTEN:3}) exten => _*72.,2,Answer exten => _*72.,3,Wait(1) exten => _*72.,4,Playback(loligo/call-fwd-unconditional) exten => _*72.,5,Playback(loligo/for) exten => _*72.,6,Playback(loligo/extension) exten => _*72.,7,SayDigits(${CALLERIDNUM}) exten => _*72.,8,Playback(loligo/is-set-to) exten => _*72.,9,SayDigits(${EXTEN:3}) exten => _*72.,10,Macro(hangupcall) ; prompting c...

Hi, There is any simple way to have external access to the Asterisk database? I want to be able to edit a specific family only, which is a phonebook. Thanks, Dan

How do you suggest doing that? How can I convert wav files to gsm files? thanks Shoval Tomer, MCSE IT Manager Softov Advanced System Ltd. Email: shoval@softov.co.il Mobile: 972-55-229220 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031102/f84d7805/attachment.htm

was wondering if anyone could give us a run through an explanation of the wiki and other examples of connecting to iptel's sip express router using asterisk pbx so i can understand better the call processing .. given the example i work from on john todd's www.loligo.com site ; exten => _3.,1,SetCallerID(${IPTELUSERID}) exten => _3.,2,SetCIDname(${IPTELUSERNAME}) exten => _3.,3,Dial(SIP/${EXTEN:1}@iptel) exten => _3.,4,Playback(invalid) exten => _3.,5,Hangup are the first two statements mandatory to connection to iptel.org ?? i understand th...

Is there any way to define call parking parameters for different contexts? For example, if I have a client in context 100 and another client in context 200, can they both define parking positions, say, from 701-710, where 701 in context 100 is different from 701 in context 200? Or even better, can context 100 define parking positions 701-710 and context 200 define parking positions

I was at VON in Boston today and saw on the Digium stand a Cisco 7960 with a picture of Tux and the Asterisk log on its display. I WANT IT!!!!! Anyone know where I can download this file please? -- Mark, G7LTT/KC2ENI Randolph, NJ http://www.g7ltt.com

...o 827-4v and *, as I am using between * and the 7910 phones? I read through the pdf docs and looked around but in this situation I assume all traffic and etc will go through *? Or will it just talk to * in the loop and use the Skinny/SCCP to talk to the router. Now I have come across http://www.loligo.com/asterisk/cisco/827-4v/cisco-827-4v-with-asterisk-version1 Which seems to use SIP between the router and *. Not sure if they were using SIP to communicate to the phones or not. Please excuse my ignorance in this matter I am a newbie to PBX, VOIP, and *. There is allot to take in at once and I...