search for: moj

...es rampant in this thing, YMMV, details about what works and what doesn't are on the page linked to above. This is more of a proof-of-concept release than even v0.1. Thanks for reading. I hope this provides a viable alternative to the great but seeming-to-be-not-updated gastman software. Moj

?? Hello.? We are very successfully using asterisk (in the form of trixbox 2.2/asterisk 1.2).? We run a few conference lines for customer teleconferences which mostly work well but they seem to operate at half duplex.? If a person starts talking they will cut off others on the call.? Is this normal behavior?? Are there any options I can change to change this? ?? Thanks! James -------------- next

...ple checking their voicemail, people in echo rooms and meetme conferences, and people in IVR things. Not sure what else I'll have in it eventually, we'll see. It's only tested with SIP, IAX should work but dunno. I'll post back when I improve it. Comments, suggestions welcome! Moj -- Mojo <mojo@horanappraisals.com> Office Manger, Horan & Company, LLC (907) 747-6666 x112

Mojo with Horan & Company, LLC wrote: > And it makes *clear* calls assuming you're within allowable range. > Speakerphone seems to work well too. I meant to mention that the DTMF tones and dialtone sound like they're played at such a high volume that they clip through the handset...

Hi List; I read the following sentence: "The CLI prompt is set with the ASTERISK_PROMPT UNIX environment variable" In the following link: http://www.voip-info.org/wiki/index.php page=Asterisk+CLI+prompt The question is: what is the ASTERISK_PROMPT UNIX environment variable and where I can access it to change it? Also where I can find information about it? Regards Bilal Ghayad

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

Hi, Is it possible to get the remote channelname that will be bridged when the call is answered, only having the channel that is in the Ring(ing) state? As far as I can see no variable seems to fit when doing the show channel command. I want to be able to redirect/manipulate an incoming call before it gets answered/bridged, but to do that I have to now which channel to use. Is there a way?

Can I do this? I have a x100p card on my PSTN line and I have an incoming context for these calls which uses the "s" extension. I'm wanting to set up a simple IVR and would like to be able to test the dialplan as I go. But having to dial-in on my PSTN line each time is going to cost me money. Can I connect to my zap_incoming context from my locally connected SIP phone? I'm

Hello Newbie question here: I have a box running Ubuntu Linux 7.10 "gutsy gibbon", and have a single Digium TDM410E card, with 1 FXO module fitted and connected to my landline. I have it answering the landline, directing to SIP phones, diverting to voicemail etc - and it works great. What I can't work out is how to dial Out from this single card. It is possible? if so, is

This is really strange. Every message to the (VGA) console is written twice to the screen, but not on the SSH connection. Any clues how to stop this behavior? -- Executing BackGround("Zap/216-1", "custom/3566/91_0000|m|") in new stack -- Executing BackGround("Zap/216-1", "custom/3566/91_0000|m|") in new stack Bart

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!

> -----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 dia...

I've had slew of problems with my Bell Canada Single Number Reach (SNR) dropping in the past couple of months. Another outage Monday for several hours has me wondering if there's a way to 1. Make a call out of my system via a PSTN back to my SNR line, say every 30 minutes (this I'm sure is easy enough via the call file...however...) 2. Track the outgoing call and match to an

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

Hi, I'd like to try using a good quality microphone and a set of PC speakers (in the first instance) to create a powerful speakerphone; if I get that working, I'll probably try more elaborate audio equipment. For this to work, I'll need software acoustic echo cancellation, or the caller at the other end will constantly hear his/her voice echoing back. I gather Asterisk can do

Hello, I have some problems to use G722, when my client sent an invite request to asterisk using G722/16000 codec asterisk respond with G722/8000 codec. I dont know exactly if Asterisk supports G722/16000 codec?? If yes how can I activate It?? Thanks. Rachid. Below wireshak trace:

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

....c term.c translate.c ulaw.c utils.c --------------------------------------------------------------------------- and after around 4-5 seconds, this same bit spits out again. is it possible that the code from 07-20-2005 just won't compile? or should I not make clean before making from cvs? Moj

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