<br><br><div class="gmail_quote">On Jan 30, 2008 10:35 PM, Dan Austin <<a href="mailto:Dan_Austin@phoenix.com">Dan_Austin@phoenix.com</a>> wrote:<br><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<div class="Ih2E3d">Franklin wrote:<br>> ztdummy can give you issues as a timing device.<br></div>Yes and no. See below<br><div class="Ih2E3d"><br>> Any way you could try using a Digium card just<br>> as a timing device to see if this helps?<br>
<br><br></div><div class="Ih2E3d">Tomasz wrote:<br>>> I am using Debian OS kernel 2.6.22-3-amd64<br>>> and zaptel driver 1.4 with ztdummy module for meetme<br>>> application. I use meetme with SIP channels.<br>
<br></div>Your kernel is new enough that you should be able to<br>leverage hi-res timers (you might need to patch ztdummy),<br>or at least a RTC set to 8192 ticks/sec. What does<br>dmesg show after ztdummy is loaded?</blockquote>
<div>it is 1024<br>Zapata Telephony Interface Registered on major 196<br>Zaptel Version: SVN-branch-1.4-r3748<br>Zaptel Echo Canceller: MG2<br>ztdummy: RTC rate is 1024<br><br>how can I increase it? <br></div><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
<br><div class="Ih2E3d"><br>>> I have such problem that when one connects to the<br>>> conference voice is "cut". Each voice sequence is<br>>> disturbed.<br></div>Do you have internal_timing=yes in asterisk.conf?<br>
This option allows Asterisk to time the RTP stream<br>based on zaptel/ztdummy clock and not on the received<br>RTP stream. In a MeetMe, where callers might mute<br>themselves, the received RTP stream is all but useless<br>
for timing.</blockquote><div>Yes I have it set. <br></div><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><br><font color="#888888"><br>Dan<br>
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