similar to: Brief silences during calls

Hi, how can I convert .wav files to .WAV: # file greet.* greet.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz greet.WAV: RIFF (little-endian) data, WAVE audio, GSM 6.10, mono 8000 Hz using 'sox'? Thanks -- Domenico Viggiani

Hello - I have 8 polycom 501s all setup great using ulaw. We have put them through a pretty rigorous torture over the last 4 months, and they've performed famously. No dropped calls ever. We invested in some g729 licenses. changed my ipmid.cfg so that g729 is priority 1 and ulaw is priority 2. I added allow=g729 to my extension's sip.conf entry, where existed before disallow=all

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

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

i have ordered 500s from tritechcoa.com several 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

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

Hi, I'm using Asterisk@home and am having trouble using the conference bridge that comes built in. We're using Polycom phones. When we transfer the first person into the conference room (e.g. 8101) , they get into the room fine. When we try to transfer a second person into the conference room, they get dropped as soon as we finish the transfer. This is using Polycom SoundPoint 301

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

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

I want to have something for the kids to play with which just records until silence is detected, plays back what was recorded, then repeats. They are having fun with Echo() at the moment :) I have mocked something up with: exten => *93,1,Answer exten => *93,n,Record(/tmp/echo:alaw|1) exten => *93,n,Playback(/tmp/echo) exten => *93,n,Goto(2) But it has the shortcomings that a beep is

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

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?

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

I'm wondering whether there's a problem with the blindxfer and atxfer commands. I was using Asterisk STABLE and pressing the # key to transfer calls worked fine, except of course when you called up FedEx and they asked "Enter the number of packages, followed by the Pound key". I found on the wiki (http://www.voip-info.org/tiki-index.php?page=Asterisk+config+features.conf) that

[This email is either empty or too large to be displayed at this time]

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

Hi, This may be a stupid/easy question for many of you. Q. how do I know what codec is being used for a particular call or call leg? Thanks. AK -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051006/5c225e0b/attachment.htm

I'm having slight problems with outgoing audio quality on Zap channels. People hear an interrupted voice. Can anyone help..? Regards, Fabrizio Mazzoni Macron SPA