<div dir="ltr"><div>Search jitter in sample sip.conf. Everything is well documented there.<br><br><br></div>Regards,<br>Qasim<br></div><div class="gmail_extra"><br><br><div class="gmail_quote">On Tue, Apr 23, 2013 at 3:03 PM, Muhammad Yousuf <span dir="ltr"><<a href="mailto:muyousif5@gmail.com" target="_blank">muyousif5@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">I am using asterisk as SIP/GSM gateway. I have 2 gsm cards installed in server. I am having some issue in audio quality. I want to enable jitter buffer on asterisk but don't know, how to do. Any one can help me.<br>
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