Hi, I have a customer running a 120 second long WAV file on their MoH. The problem is that it's always starting from the beginning, so people being put on hold, talked to, put on hold again, etc always hear the first 10-15 seconds. Is there a way to have Asterisk MoH remember where it left off? Or at the very least just play the same stream to all people using the same MoH class, so that it just plays like a CD and the person hears wherever the stream is at at a given moment? Regards, Mike -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090309/3bde4753/attachment.htm
Danny Nicholas
2009-Mar-09 19:16 UTC
[asterisk-users] MoH - always starting from the beginning
According to the documentation I've read, MOH will always start at the start of the file. You could possibly put all of the on-hold folks into a conference room, but a more transparent option would be to create "staggered" versions of the wav file and add random=yes to moh.conf 1.wav = original wav 2.wav = wav + 10 seconds 3.wav = wav + 20 seconds _____ From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Mike Sent: Monday, March 09, 2009 2:02 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] MoH - always starting from the beginning Hi, I have a customer running a 120 second long WAV file on their MoH. The problem is that it's always starting from the beginning, so people being put on hold, talked to, put on hold again, etc always hear the first 10-15 seconds. Is there a way to have Asterisk MoH remember where it left off? Or at the very least just play the same stream to all people using the same MoH class, so that it just plays like a CD and the person hears wherever the stream is at at a given moment? Regards, Mike -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090309/475cf7f7/attachment.htm
Running an earlier version of Asterisk (1.2), we were using Hints to show busy extensions on other (SNOM) phones. When we went to version 1.4 they stopped working, using the same syntax. (Copied and pasted) Does anyone have any tips or clues? Is the exact location in the file critical? Maybe we put the code in a "back alley"? Cary -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090309/3f84d349/attachment.htm
Mark Michelson
2009-Mar-09 19:54 UTC
[asterisk-users] MoH - always starting from the beginning
Mike wrote:> Hi, > > > > I have a customer running a 120 second long WAV file on their MoH. The > problem is that it's always starting from the beginning, so people being > put on hold, talked to, put on hold again, etc always hear the first > 10-15 seconds. > > > > Is there a way to have Asterisk MoH remember where it left off? Or at > the very least just play the same stream to all people using the same > MoH class, so that it just plays like a CD and the person hears wherever > the stream is at at a given moment? > > > > > > Regards, > > > > Mike >What version of Asterisk are you using? There was a recent bug introduced in 1.4.23. The fix for the issue is here: http://svn.digium.com/svn-view/asterisk?view=rev&rev=174218 Mark Michelson
To get busy state for a sip channel in 1.4 it appears the peer/friend must have a call-limit. Steve On 3/9/09, Cary Fitch <caryf at usawide.net> wrote:> Running an earlier version of Asterisk (1.2), we were using Hints to show > busy extensions on other (SNOM) phones. > > > > When we went to version 1.4 they stopped working, using the same syntax. > (Copied and pasted) > > > > Does anyone have any tips or clues? > > > > Is the exact location in the file critical? Maybe we put the code in a "back > alley"? > > > > Cary > > > >-- Sent from my mobile device
According to voip-info.org, the call-limit is mandatory to make hints work as of 1.4.X. -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Stephen Davies Sent: Monday, March 09, 2009 4:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Hints To get busy state for a sip channel in 1.4 it appears the peer/friend must have a call-limit. Steve On 3/9/09, Cary Fitch <caryf at usawide.net> wrote:> Running an earlier version of Asterisk (1.2), we were using Hints to show > busy extensions on other (SNOM) phones. > > > > When we went to version 1.4 they stopped working, using the same syntax. > (Copied and pasted) > > > > Does anyone have any tips or clues? > > > > Is the exact location in the file critical? Maybe we put the code in a"back> alley"? > > > > Cary > > > >-- Sent from my mobile device _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Andrew Thomas
2009-Mar-10 07:08 UTC
[asterisk-users] MoH - always starting from the beginning
You could always run a shoutcast server and stream from that. ? ? -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Mike Sent: 09 March 2009 19:02 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] MoH - always starting from the beginning Hi, I have a customer running a 120 second long WAV file on their MoH.? The problem is that it's always starting from the beginning, so people being put on hold, talked to, put on hold again, etc always hear the first 10-15 seconds. Is there a way to have Asterisk MoH remember where it left off? Or at the very least just play the same stream to all people using the same MoH class, so that it just plays like a CD and the person hears wherever the stream is at at a given moment? Regards, Mike