Is anyone using something to log SIP results (connected/not, latency) that they really like?We do some logging using simple scripts writing the results of sip show peers to a text file if customers report issues, but it would be nice to have a tool t..
As system administrator, monitoring the continuity of services is vital. Today I would like to highlight some tools that could come in handy for VoIP monitoring.NagiosFor those of you who didnt know it, Nagios can be configured to monitor pretty m..
all, Ive done a basic install of 10.1.2 to have a play with the new ConfBridge application and have noticed high latency when in a conference. Its to the order of 900ms or so which is just too much for a conference to work well. I can account for ab..
All, How can I find out One way latency from my PBX to my SIP Trunk Provider. My SIP provider recommends a One way latency of 100ms for good Voice quality. Ping request to their IP Address gives me a response in approx. 260ms. Will that be good eno..
List, We are using F parameter in meetme Dialplan application to broadcast SIP INFO (1 and 0) as DTMF tone to all the participants. The DTMF configuration for all the connected SIP clients is SIP INFO. The problem we are seeing, asterisk is taking s..
On Mon, Jun 6, 2011 at 10:10 AM, Steve Edwards wrote: > On Mon, Jun 6, 2011 at 2:26 AM, Steve Edwards >> wrote: >> >> I strongly suggest using an existing library for the language of your >> choice. >> > > On Mon, 6 Jun 2011, A E [Gmail] wrote: > >C..
10% ________________________________ De: Matt Riddell Para: firstname.lastname@example.org Enviadas: Quarta-feira, 4 de Maio de 2011 0:32:28 Assunto: Re: [asterisk-users] Fading voice problem On 3/05/11 10:16 PM, Eduardo Leones wrote: > Guys, > >..
Guys, Im having problems in the fading voice calls, receptive and active, that in SIP accounts. While few people using the system, calls are perfect, but it beats the normal use of connections (average 30 concurrent), the voice begins to fade from peop..
On 11/29/2010 11:11 AM, Tony Mountifield wrote: > I have recently built a single-T1 Asterisk box using an HP DL120G6 > with a Digium TE122 card. > > I was finding that I was getting missed interrupts on the TE122, > causing the driver to report t..