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? -- Telecomunicaciones Abiertas de M?xico S.A. de C.V. Carlos Ch?vez +52 (55)9116-91161
<!DOCTYPE html>
<html><head>
<meta charset="UTF-8">
</head><body><p>Hello,</p><p>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;</p><p>Thanks !!!!</p><blockquote
type="cite"><p>Le 5 avril 2016 à 23:23, Carlos Chavez
<cursor@telecomabmex.com> a écrit :<br><br><br>On 4/5/16
3:17 PM, Joshua Colp wrote:</p><blockquote
type="cite"><p>Carlos Chavez
wrote:</p></blockquote><p>>> 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<br>>> could cause the
voice to sound cracked and with small interruptions. I<br>>> am
looking at the timing source for Asterisk and it is currently
using<br>>> timerfd even though we have an E1 card installed. Is
timerfd better than<br>>> dahdi? Any recommendations to test if
timing may be a problem for voice<br>>> quality and
DTMF?</p><blockquote type="cite"><p>What is the
scenario and the channels involved? Timing is only used <br>for things
such as playback, music on hold, and ConfBridge. If it's <br>strictly
a two party call then Asterisk forwards media as
received.</p></blockquote><p>The problem appears on all calls,
no matter the source or destination. There are desk phones, softphones and a
couple SIP trunks <br>to another office. They all experience the problem.
Calls between <br>extensions, from or to the E1, from or to trunks. The
only scenario <br>left to try is connecting calls only via the E1 so we
completely <br>eliminate the network side of things and se if we get the
same <br>behaviour. During calls you can hear some background noice and
<br>interruptions in the voice. DTMF fails when we try to dial to external
<br>IVR.<br><br> I do not really believe that the fault is in
the Asterisk server <br>but I have to eliminate all posibilities on my
side before I can lay <br>blame on the network infrastructure. I was also
just wondering if DAHDI <br>would not be a better timing source for
Asterisk since it is hardware based?<br><br>--
<br>Telecomunicaciones Abiertas de México S.A. de C.V.<br>Carlos
Chávez<br>+52 (55)9116-91161<br><br><br>--
<br>_____________________________________________________________________<br>--
Bandwidth and Colocation Provided by http://www.api-digital.com --<br>New
to Asterisk? Join us for a live introductory webinar every Thurs:<br>
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http://lists.digium.com/mailman/listinfo/asterisk-users</p></blockquote><p><br></p><div
class="io-ox-signature"><p>Mamadou
NGOM</p><p>Ingénieur Télécommunications &
Réseaux</p><p>Mobile: 06 72 45 23 03</p><p>Skype:
Mamadou Numericap</p><p>NumeriCap – SAS au capital de 30.000,00€ -
RCS de Toulon N° 530188432 – TVA FR 485301188432 – APE6110Z - ARCEP N°13/0015.
<br>siège social : « le Galaxie C » 526 avenue Maréchal de Lattre de
Tassigny 83000 Toulon. <a
href="mailto:mail%3Afinance@numericap.com">mail:
finance@numericap.com</a><br>Centre d’exploitation : « Résidence les
Coquières » 11 avenue Joseph Fallen - 13400 Aubagne – Tel
:<a>04.42.73.88.52</a> <br></p></div></body></html>
Carlos Chavez wrote:> 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?Either timerfd or dahdi should perform the same. I wouldn't expect dahdi to improve things unless something was really wrong on the system itself. -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org
On Tue, 5 Apr 2016, Mamadou NGOM wrote:> 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;1) You should start a fresh thread rather than hijacking an existing unrelated thread. 2) You should show us what you have done so far. Most SIP providers have sample snippets for your sip.conf and extensions.conf files. -- Thanks in advance, ------------------------------------------------------------------------- Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281
On Tue, 5 Apr 2016, Mamadou NGOM wrote:> 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;0) Don't top-post. -- Thanks in advance, ------------------------------------------------------------------------- Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST https://www.linkedin.com/in/steve-edwards-4244281
On Tuesday 05 Apr 2016, Mamadou NGOM wrote:> 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 !!!!Rule One: Start your own topics -- don't jump in on someone else's, unless it's actually relevant. Rule Two: Type your reply *after* the thing you are replying to. Your SIP trunk provider will know how to connect an Asterisk box to them. That, after all, is what most people are going to be connecting. -- AJS Note: Originating address only accepts e-mail from list! If replying off- list, change address to asterisk1list at earthshod dot co dot uk .