search for: horanappraisals

Hiya, hope I don't bore anybody with this. There are certainly a lot of monitor-y things out there and they just didn't fit my need, so maybe this will fit someone's besides mine. http://horanappraisals.com/asterisk/pbxmonitor/ contains two files. one is a php script called pbxmonitor, and one is a flat file of extensions to extension name mappings of internal users. It contains example data that needs editing to fit your scenario. so the pbxmonitor.db might have (separated by tabs): SIP/200...

I am looking for a way to allow a user to record a call simply by pressing a button or some combination of buttons on their phone. Is this possible? I have read the stuff about the monitor command; however, this doesn't seem to be very interactive for the user. Thanks, Andy -- H. Andy Goss Network Engineer Network Advocates Inc. Main: 502.412.1050 DID: 502.992.5933 Mobile: 502.387.8216

...al times over the past 4 months. great service and delivery, and the prices are lowish, only problem is, they add a $20 handling fee per phone, on top of phone price, and shipping, making the lower price not as good -----Original Message----- From: Mojo with Horan & Company, LLC [mailto:mojo@horanappraisals.com] Sent: Friday, July 15, 2005 12:01 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] seems-to-be-inexpensive source of polycom 301 and501 Hello, sorry if this kind of a price really isn't news to you all, but it seemed really good to me. www.tritec...

...ello? hello?. Neither of these things happen when the phones runs in ulaw. Does anyone have any idea where to look? I'll post whatever logs anyone thinks might help. I'm using 1.2.0b1, but this occurred with my CVS HEAD of around 7/20/2005 as well. Thanks! Mojo -- Mojo <mojo@horanappraisals.com> Office Manger, Horan & Company, LLC (907) 747-6666 x112

Im not looking for anything super detailed, just something to run through the master.csv file and give total time per account code. . . .does anyone out there have a script like this I could work from?

...hese steps to reset it, but it worked for me. YMMV... After provisioning, the Iaxy works great; I'd recommend it to anyone looking for an inexpensive, simple FXS. --Mike >Message: 9 >Date: Fri, 31 Mar 2006 09:05:45 -0900 >From: "Mojo with Horan & Company, LLC" <mojo@horanappraisals.com> >Subject: Re: [Asterisk-Users] IAXY codec support and questions.. >To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> >Message-ID: <442D6F79.7010905@horanappraisals.com> >Content-Type: text/plain; charset=ISO-8859-1; fo...

The only advantage is when you travel. Last year I took my wifi sip phone to Astricon in Madrid and everything worked as expected. I am just packing it and heading for Paris... Wojtek -----Original Message----- From: Mojo with Horan & Company, LLC [mailto:mojo@horanappraisals.com] Sent: Tuesday, June 20, 2006 1:16 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Anyone using VoIP WiFi phones? I have been pretty happy with my cisco 7920, but it has been by the wayside for six months or more now due to the wimpy battery life....

On this page http://www.voip-info.org/wiki-Asterisk+record+calls there is "Example by Mojo". I have done everything he said and I have sox package installed. [root@pbx recordings]# sox -help sox: Version 12.17.7 ... When I open this web page http://10.0.0.26/recordings/index.php I get this: No Recordings Found And there are recordings in /var/spool/asterisk/monitor Do I have to do

Hello all, I need some help with my Cisco 7960 enabling TFTP. Does anyone know what numbers to press in the menu? Or can I enable this through telnet? Many thanks, Christian

I am running 1.2.5 with a simple queue and have announce-holdtime = yes in queues.conf for that queue. The person is being told their posistion in the queue and the CLI says the estimated hold time, but it never plays it for the caller. It worked previously, i am not sure when it stopped, i think after 1.2.1. Is this a known bug? I dont want to report it to the bug tracker if its already been

I am getting read to roll out close to 100 polycom phones and wondered if any one knows of a program to take a list of MAC addresses, extensions, and names and generate the configuration files? -- Bruce Nortex Networks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060504/d3bb612a/attachment.htm

Hi, I'm having trouble setting the ringtone on my Polycom 501. The relevant entry in extensions.conf is: exten => 801,hint,SIP/creative1 exten => 801,1,SetVar(ALERT_INFO="Test") exten => 801,2,Dial(SIP/creative1,20,Ttr) In the sip.cfg: <alertInfo voIpProt.SIP.alertInfo.1.value="Test" voIpProt.SIP.alertInfo.1.class="13"/> and <TEST

Ok asked this earlier with no response so I will phrase it a different way. I am sure someone had to deal with this and there is a "best way." I want to let Asterisk make the decision on best path based on local exchange - xxx-yyy - where xxx is one of my local area codes and xxx is exchange designator. The problem is that the list is rather large. Maybe 50-100. The idea is that I can

...com-- > >asterisk-users mailing list >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users ------------------------------ Message: 4 Date: Fri, 20 Jul 2007 08:55:33 -0800 From: "Mojo with Horan & Company, LLC" <mojo at horanappraisals.com> Subject: Re: [asterisk-users] G729 copy protection To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Message-ID: <46A0E905.20209 at horanappraisals.com> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Not until the...

Hi all, Configuration: Analog phone connected to TDM400p. I'd like the phone to give a half-ring (chirp) periodically when there is a message waiting. Can this be done? How is it configured? The visible "Message waiting" indicator and the stutter dial tone are working fine, but are not sufficient for me. Thanks!

I have a gateway, which I call from my mobile phone (free of charge, since it is the same phone company). This gateway gives me a dial tone. I can than dial to any extension number or even other gateways, .... It is getting more a trouble to remember all the numbers, or to key in all the long phone numbers when you got the dialtone. I was thinking of using for this Sphinx2. How can I

> -----Urspr?ngliche Nachricht----- > Von: Mojo with Horan & Company, LLC [mailto:mojo at horanappraisals.com] > Gesendet: Dienstag, 25. M?rz 2008 23:23 > An: Asterisk Users Mailing List - Non-Commercial Discussion > Betreff: Re: [asterisk-users] Asterisk parking hold and > transferdigittimeout > > It seems that the dialplan comes into play. If your parking > lot is 700, >...

How do I combined these in and out wav files on the fly through asterisk to where I hear the whole conversation and only have one wav-file (i.e. : agent-1001-asterisk-478-1127389080-17-in_out.wav) agent-1001-asterisk-478-1127389080-17-in.wav agent-1001-asterisk-478-1127389080-17-out.wav __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best

Hello, I have VPC (Voice Pulse Connect) and NuFone for providers and I have setup modules.conf with the registered (Digium) G.729 Codec such as: load => codec_g729a.so load => res_crypto.so With both sip/iax2 configuration disallow=all is first and then allow=g729 is next (allow=ulaw,allow=alaw,allow=gsm are next after allow=g729) and it always dials via ulaw. Why is this happening? Josh

Dear Group, I have my Asterisk box working with a Mediatrix 1204. I have 2 questions; 1) I do not seem to get a Call ID on the call coming via the Mediatrix 1204. I was wondering if anyone had this configured and if they could share this with me? 2) How do you route a call based on caller ID on Asterisk. At the moment I'm routing calls via DNIS. Thanks and Regards Shad Mortazavi