Hello,
I am doing a configuration for connecting my server asterisk to a SIP provider. I ask if somebody can give me a basic code or a link to begin well;
Thanks !!!!
Le 5 avril 2016 à 23:23, Carlos Chavez <[email protected]> a écrit :
On 4/5/16 3:17 PM, Joshua Colp wrote:Carlos Chavez wrote:
>> 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 Asterisk and it is currently using
>> timerfd even though we have an E1 card installed. Is timerfd better than
>> dahdi? Any recommendations to test if timing may be a problem for voice
>> quality and DTMF?What is the scenario and the channels involved? Timing is only used
for things such as playback, music on hold, and ConfBridge. If it's
strictly a two party call then Asterisk forwards media as received.The problem appears on all calls, no matter the source or destination. There are desk phones, softphones and a couple SIP trunks
to another office. They all experience the problem. Calls between
extensions, from or to the E1, from or to trunks. The only scenario
left to try is connecting calls only via the E1 so we completely
eliminate the network side of things and se if we get the same
behaviour. During calls you can hear some background noice and
interruptions in the voice. DTMF fails when we try to dial to external
IVR.
I do not really believe that the fault is in the Asterisk server
but I have to eliminate all posibilities on my side before I can lay
blame on the network infrastructure. I was also just wondering if DAHDI
would not be a better timing source for Asterisk since it is hardware based?
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Carlos Chávez
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