similar to: Asterisk Sends 180-RINGING to UAeven withprogressinband=yes

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

Andrew, I don't think it's a Polycom issue. We took Asterisk out of the picture and had our Polycom phones communicate directly with an Audiocodes PSTN gateway. Unlike Asterisk, the audiocodes do not send 180 Ringing before sending 183 Session Progress, and the polycom's play the correct tones in this case. We WANT Asterisk to send progress tones in band. In our case it IS needed.

We must have had the magic version of 1.6.x then, because we increased our buddy watch limit from 8 to 48 in that version. -----Original Message----- From: Jessee J Holmes [mailto:jholmes@atacomm.com] Sent: Wednesday, October 11, 2006 8:18 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] RE: Welcome to the "asterisk-users" mailing list Limit

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. > >

I have progressinband=yes in sip.conf, but Asterisk sends a 180-Ringing to my polycom phones and then it also sends 183-Session Progress. That doesn't seem to make sense. Shouldn't Asterisk NOT send 180-Ringing if progressinband=yes ? Doug.

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

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:

I think that limit was increased in 1.6.6 or 1.6.7. -----Original Message----- From: C F [mailto:shmaltz@gmail.com] Sent: Tue 10/10/2006 6:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Subject: Re: [asterisk-users] RE: Welcome to the "asterisk-users" mailing list On 10/10/06, Eric ManxPower Wieling <eric@fnords.org> wrote: > What I

Is there a way to get asterisk to send you a email when it looses or an extension doesn?t re-register Roger Workman Business Development Upperclassman/Universal Holdings LLC Voice: 304.324.3800 Fax: 304.324.3801 ICQ: 4447584 Website: http://www.upperclassman.net Billing Questions: billing at upperclassman.net Rental Questions: rentals at upperclassman.net Maintenance: help at

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> >

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

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

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

Sangoma support did something and the driver is there. Rebooting now with the new card. Hold me, I'm scared. -----Original Message----- From: Colin Anderson [mailto:ColinA@landmarkmasterbuilder.com] Sent: Friday, December 22, 2006 10:09 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Sangoma Wanpipe 2.3.4-3 compilation fails un der FC2

I have a ALPHA version of my new ping AGI demo available. Access via: IAXTel 1-700-923-3645 or Dial(IAX2/guest@ext.fnords.org) When asked for an extension, enter 2101. This will bring you 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

Mitel ICP 3300 & Asterisk, Is possible that integration? (C?sar) -----Mensaje original----- De: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] En nombre de asterisk-users-request@lists.digium.com Enviado el: Jueves, 25 de Mayo de 2006 03:00 p.m. Para: asterisk-users@lists.digium.com Asunto: Asterisk-Users Digest, Vol 22, Issue 147 Send

-----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of asterisk-users-request at lists.digium.com Sent: Wednesday, August 22, 2007 10:51 PM To: asterisk-users at lists.digium.com Subject: asterisk-users Digest, Vol 37, Issue 88 Send asterisk-users mailing list submissions to asterisk-users at lists.digium.com