I will try to answer it this way:<br><br>G.711 is toll quality voice, if everything is functioning properly should be almost identical to a regular phone call. <br><br>You will need to do trouble shooting to (in the words drilled into me by an old boss): isolate, identify and quantify the issue. I would start by setting up a record/playback extension, call it from PSTN and call it from the SIP phones, see where the noise is being introduced, from there could be hundreds of different things (LAN congestion, interrupt sharing on the PSTN card, bad wiring, faulty switch, etc).
<br><br>So to answer your question there isn't a parameter that says "Noise=Yes/No", you need to: isolate, identify and quantify the noise. <br><br><div><span class="gmail_quote">On 9/14/07, <b class="gmail_sendername">
satish patel</b> <<a href="mailto:satish_patel_2000_2000@yahoo.com">satish_patel_2000_2000@yahoo.com</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
I have both type of call sip-2-pstn and pstn-2 -sip but quality is not good so how to check asterisk voice quality and codec quality i am useing G.711 alaw and ulaw and it is my LAN network so is there any specific perameter or option to improve quality of voice ???
<div><span class="e" id="q_1150422605df556b_1"><br><br><b><i>Adrian Marsh <<a href="mailto:Adrian.Marsh@ubiquisys.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">Adrian.Marsh@ubiquisys.com</a>
></i></b> wrote:<blockquote style="border-left: 2px solid rgb(16, 16, 255); margin-left: 5px; padding-left: 5px;"> Satish,<br><br>Whats your network setup? Do you get bad quality on internal-asterisk calls, or only on external calls? Are you making pure IP calls (sip2sip), or are there E1/T1 cards involved? What codecs are you currently using? What devices are you using?
<br><br>Adrian Marsh<br> <br>________________________________________<br>From: <a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">asterisk-users-bounces@lists.digium.com
</a> [mailto:<a href="mailto:asterisk-users-bounces@lists.digium.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">asterisk-users-bounces@lists.digium.com</a>] On Behalf Of satish patel<br>Sent: 14 September 2007 06:48
<br>To:
<a href="mailto:asterisk-users@lists.digium.com" target="_blank" onclick="return top.js.OpenExtLink(window,event,this)">asterisk-users@lists.digium.com</a><br>Subject: [asterisk-users] Asterisk voice quality tuning<br><br>
Dear all<br><br> I have asterisk 1.4.11 on CentOS. I have SIP IP phone arround 100 but i got Noice on voice call so what would be the resone and how to fine tune my voice quality on asterisk ?? what codec would be best for my asterisk
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