I am currently having a voice quality problem with one of our Asterisk servers.We have checked the network and we have found no problems that could cause the voice to sound cracked and with small interruptions.I am looking at the timing source for Aster..
Hello.Voice quality when calling – this is one of the most important in the PBX. You need to record the quality parameters for each call to improve.Because the overall quality of a call can only be determined upon completion, I did it in the HangUp hand..
Hello;What is the best method to let the voice quality through Dahdi channels to be clear and no echo? Is it the wanpipe or it is working only with sangoma?Reg..
Any one used Digium IP Phones D40?
I need to know if they are stable with good voice quality? Comparing to Polycom 330, which is better? Let us talk frankly although I know that we have to support Digium.
On 04/15/2012 07:26 PM, Patrick Lists wrote: > On 04/15/2012 01:15 PM, Gustavo Garcia Bernardo wrote: >> Is it a good idea to use asterisk transcoding from G711 to iLBC or >> should I find out any other solution not involving transcoding (f.e. >> us..
On 04/15/2012 01:15 PM, Gustavo Garcia Bernardo wrote: > Is it a good idea to use asterisk transcoding from G711 to iLBC or > should I find out any other solution not involving transcoding (f.e. > using G.729 that is supported in both sides). Im worr..
Is it a good idea to use asterisk transcoding from G711 to iLBC or should I find out any other solution not involving transcoding (f.e. using G.729 that is supported in both sides).Im worried about voice quality and trying to avoid paying for G.729 licensi..
A client of ours get lots of problem with there voice quality when the do a lot SIP calls. In a application I log the rtpqos audio jitter an lost packets.(see Below) Does anybody know what the numbers mean? If I look at a sample of the channel variabl..
We noticed a very sharp drop in voice quality when using digium g729a codec. The problem seems to happen if the A channel (callers channel) is a landline/mobile number contacted using the same outgoing provider (as a local channel). It sounds like listen..
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..