similar to: Hardware TDM Switching (Out Of Office - on vacation)

> -----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: >

The cheap ones on EBay won't work with the SC420 server. I have one and can't make any of the clones work. I do have one TDM40B card for analog 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:

That's the main reason I opened this thread as it surprised me a bit ... > > > Any 2-wire analog leg will be a source of echo. Many, many, many calls > do not have a 2-wire leg. Even in handset audio circuit ? I was thinking that any handset is a potential echo source due to this audio circuit ... Do you agree ? > Think cell/mobile or endpoints with PRI or T-1. > >

Has anyone figured out a way to instantaneously swing over PRIs bearing calls in progress to another media gateway without dropping them? Obviously, this would require a DACS of some sort. But I am thinking that it is possible to swing T1s over in a DACS without actually causing the endpoint to reframe as long as the other endpoint is kept in sync. So, it'd be nice, for example, to bring

Yes, I am aware of that as well. 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

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 you may call via IAXtel 1-700-923-3645 extension 2101. Option 23 is

Hi all, Suppose I have a vector Fruits <- c('Apple','Orange','Pear','Banana','Mango') I want to print the statement "The fruits in the basket are: Apple, Orange, Pear, Banana, Mango" If I do: print(paste('The fruits in the basket are:',Fruits)) The output is: [1] "The fruits in the basket are: Apple" "The fruits in

Folks, Eric is spot-on--both phones are happy with the modified ${CALLERIDNAME} value since I removed the quotes in the call to SetCIDName(). I've replace my calls to SetCallerID() since I think using SetCIDName() is cleaner. Cheers, Rob > -----Original Message----- > Date: Sat, 29 Jan 2005 16:44:16 -0600 > From: Eric Wieling aka ManxPower <eric@fnords.org> >

The agenda for Astricon Europe in Madrid June 15-17 is now coming together. It will be an international conference, with speakers from both USA and Europe. Last year, we had over 25 nationalities participating in the first Astricon - the Astricon where Mark released Asterisk 1.0 on the conference floor, during his keynote! Many active members of the Asterisk community talks at the conference, one

I don't see that anywhere. Here is my zapata.conf This is only happing on my 7970 all other phone are working without trouble. [channels] context=pri signalling=pri_cpe switchtype=dms100 group=1 usecallerid=yes hidecallerid=no restrictcid=no usecallingpres=no useincomingcalleridonzaptransfer=yes callerid=asreceived faxdetect=incoming musiconhold=default echocancel=yes

On Fri, 2006-10-13 at 10:50 -0600, Joseph wrote: On Fri, 2006-10-13 at 07:27 +0200, Remco Barendse wrote: > > On Thu, 12 Oct 2006, Eric "ManxPower" Wieling wrote: > > > > > Matt Florell wrote: > > > > If you downgrade, let us know if it fixes things for you. > > > > > > > > It's strange that there were so many changes in the

Hi Again, Ok so I have been able to load the driver now I need help configuring it for the PRI. The provider of the PRI is AT&T. I think I have set up zaptel.conf correctly now what is the next step? Thanks Julian From: jvaranini@hotmail.comTo: asterisk-users@lists.digium.comSubject: RE: [asterisk-users] Modprobe ZaptelDate: Mon, 13 Nov 2006 19:17:50 +0000 Sorry meant

> -----Original Message----- > From: Eric "ManxPower" Wieling [mailto:eric@fnords.org] > Sent: Monday, December 11, 2006 5:11 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Asterisk Sends 180-RINGING to UAeven > withprogressinband=yes > > > Douglas Garstang wrote: > > > > > > Andrew, >

Hmmm. Ok, that's true. At the very least it will create confusing CDR's I think... maybe. We're not billing our OnNet traffic at all. Only the traffic that goes OffNet, to our switch is billed (if it leaves our switch that is...). I was thinking earlier too that we only need progressinband on traffic that goes to the PSTN, via our switch. OnNet traffic will never generate reorder

Thanks for replying. It's there, /usr/src/zaptel and it's valid. The script checks for its existence before compiling. The only thing that's unusual about this zaptel is that it is patched to support KB2 echo can. Gavin: Thanks, already cc'd Sangoma support. Hope they are working today! -----Original Message----- From: Eric "ManxPower" Wieling

like this: print STDOUT "EXEC background /var/lib/asterisk/sounds/wait-moment \n" system("program2.agi &") exit; program 2 would run while the sound played. -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Eric "ManxPower" Wieling Sent: Tuesday, December 02, 2008 3:20 PM

like this: print STDOUT "EXEC background /var/lib/asterisk/sounds/wait-moment \n" system("program2.agi &") exit; program 2 would run while the sound played. -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Eric "ManxPower" Wieling Sent: Tuesday, December 02, 2008 3:20 PM

Dear List; Thanks alot for the help. But how can I let the second dial tone (after pressing the extension to select that FXO port) to be difference than normal dial tone? Regards Bilal Ghayad -------------------------- Correction, on FXO port not FXS, second, read his email first: "Also, how it will be possible to assign an dedicated line (connected to FXO) to an button on the Polycom IP

Eric "ManxPower" Wieling wrote: >> >>> span=1,1,0,d4,ami >>> e&m=1-24 >>> > > Looks like you have told Asterisk to get it's timing from the Mitel. > I'll bet the Mitel is trying to get it's timing from Asterisk. > > Try span=1,0,0,d4,ami and run ztcfg -vvv > I just set this back. It was originally set to your

Open the console with verbose turned up. Make a test call and see where it is hanging. That will isolate the problem. > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Giorgio Incantalupo > Sent: Thursday, April 27, 2006 11:16 AM > To: Asterisk Users Mailing List - Non-Commercial