search for: manxpow

I've helped a lot of people on the mailing lists and on IRC #asterisk. and wanted to let people know that I will be in Europe between May 19 and June 21. Stockholm (VON 2005), Brussels (holiday/vacation), Amsterdam (holiday/vacation), and Madrid (Astricon). There are several weeks during my trip that I have no current plans for and may add other cities to my itinerary. I'm looking

I've moved 1 of our facilities over from 1.2 to 1.4 two weeks back. So far, the only issue that I've encounted is. I have a scheduled CRON job that runs at 3am every Sunday, that issues a: asterisk -rx 'restart when convenient' The first Sunday that it ran, Asterisk never restarted. The CRON logs show that it issued the command successfully. This Sunday, it ran but never

Is there a way to reduce the volume of the dial tone on a zap channel? I don't want to reduce the audio volume on calls so txgain in zapata.conf will not work. I am having problems with asterisk not recognizing the first dialed digit from an analog phone about 8-15% of the time. Once the dialtone goes away, the digits are always recognized. Any other thoughts on how to solve this are also

> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Eric "ManxPower" Wieling > Sent: Thursday, December 14, 2006 4:30 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Hardware TDM Switching > > asterisk@nicox.org wrote: > > > Do anybody know, if there is a way to connect 2 zap-channels...

...to the System Services menu. The Cepstral version of the ping is option 28, the Festival version of the ping is option 32. Please report problems and/or issues directly to me. I'm trying to get ping.agi stable enough to release the source code for it under GPL. Thanks in advance, Eric aka ManxPower -- Sample configs, scripts, more : http://www.fnords.org/~eric/asterisk/ BTEL Consulting 504-899-1387 or 850-484-4545 or 877-677-9643

Hi I want to submit a bug about auto-dial , but I am not sure on which project the auto-dial comes, how to know about which project , auto-dial comes Thanks Joseph ___________________________________________________________ To help you stay safe and secure online, we've developed the all new Yahoo! Security Centre.

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! I am facing some delay in conferencing. Using DIAX for Voip calls ,no hardware used yet I am using MeetMe to achieve conferencing and am having a lot of delays. Can anyone tell me how to reduce the delay Regards, Amna Saleem -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

...og stations that works well. The problem with the SC420 is that it won't let you set the interrupts yourself and you end up with interrupts being shared. =============================================================== Message: 26 Date: Mon, 25 Jul 2005 08:31:47 -0500 From: Eric Wieling aka ManxPower <eric@fnords.org> Subject: Re: [Asterisk-Users] Need Advice To: tarsi@binhost.com, Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <42E4E9C3.2080806@fnords.org> Content-Type: text/plain; charset=us-ascii; format=flowed Natha...

Hi List; What configuration needed to let my FXS send and receive FAX? Regards Bilal

...e will be sure that his outside call will be via > that specific line." > Just assign a key on your phone to dial that extension, and you will have > dial tone on selected line, > then as a traditional PBX you can send any digits to your provider. > > > On 10/1/07, Eric ManxPower Wieling <eric at fnords.org> wrote: >> ignorepat continues dialtone after a leading digit has been dialed on >> FXS ports. How does ignorepat help this guy? >> >> Al lists wrote: >>> ignorpat is your friend >>> >>> On 9/30/07, Tzafrir Coh...

Hi, does the hint extension work together with the Snom phones in stable? I don't get an error in the dialplan, but it does not work either. On SIP/26 I want to monitor SIP/22. This is what I do right now: extension.conf [incoming] exten => 955,hint,SIP/22 exten => 955,1,Dial(SIP/22) sip.conf [26] ... subscribecontext=incoming ... Running Asterisk 1.0.2-BRIstuffed-0.2.0-RC2b TIA

...ll. I guess I was wondering if other people have experienced the same problems, and - in the event of a possible solution - how they solved the problem. Is there something that can be done on our Asterisk box or at the provider's side? Bjorn > > > > > > > From: Eric \ManxPower\ Wieling [eric@fnords.org] > > Sent: 2006-03-29 22:52:34 CEST > > To: Asterisk Users Mailing List - Non-Commercial Discussion [asterisk-users@lists.digium.com] > > Subject: Re: SV: [Asterisk-Users] IAX - only one way traffic > > > > Bj?rn O wrote: > > > Thi...

Hi all, when a user dials any number, asterisk automatically generates ringing which caller can hear, and after 2 - 3 rings asterisk detects that the called user is busy, then caller hears busy tone. for example user hears--- tone--tone--tobeep beep beep ---Can i some how eliminate the false ringing at the start so that user hears only beep beep beep if the called user is busy. I have used the R

2008/9/29 Alex Balashov <abalashov at evaristesys.com> > Try this: > > exten => _XXXX,1,Set(THISTECH=${CUT(CHANNEL,/,1)}) > exten => _XXXX,n,NoOp(Technology is ${THISTECH}) > exten => _XXXX,n,Set(THISCHANNEL=${CUT(CHANNEL,/,2)}) > exten => _XXXX,n,NoOp(Channel is ${THISCHANNEL}) Hi, I don't have any spare zaptel enabled system I could try this on, but I

Some of you may know me as ManxPower from #Asterisk at irc.freenode.,net I've posted my demp weather report Asterisk AGI script at http://www.fnords.org/~eric/asterisk/downloads/ I have no affiliation with Cepstral. Below is the README: Contact: Eric Wieling <eric@fnords.org> If you want a demo of this AGI script yo...

...ould check with me before spending time as it may be a feature I have already played with locally and haven't gotten around to intergrating it into the mainline CVS code. I would like to thank miro_ for his patience and fnancial support, along with [Sim], klasstek, bkw_, PavelL, theo and ManxPower for willingly diving into nearly untested code and debuging. Lastly, we cannot forget Mark Spencer for this absolutely amazing piece of software! A quick sample config: skinny.conf: ; Typical config for a 7910 [jeremy] ; Device name device=SEP0007EB363201...

I've got this low-ping 100%-up dsl connection between two asterisk 1.2.7.1 servers. And oftentimes one of them would declare its opposite UNREACHABLE. Why can this happen? The host stanzas in iax.conf have raw IP's, so no DNS monkey business here.. An inquiring mind wants to know.

Is it possible to make Asterisk do inband DTMF over IAX?