<br><br><b><i>Benjamin Jacob <ben4asterisk@yahoo.com></i></b> wrote:<blockquote class="replbq" style="border-left: 2px solid rgb(16, 16, 255); margin-left: 5px; padding-left: 5px;"> <br><br><b><i>Tony Mountifield <tony@softins.clara.co.uk></i></b> wrote:<blockquote class="replbq" style="border-left: 2px solid rgb(16, 16, 255); margin-left: 5px; padding-left: 5px;"> In article <841715.98554.qm@web46405.mail.sp1.yahoo.com>,<br>Benjamin Jacob <ben4asterisk@yahoo.com> wrote:<br>> <br>> One on my clients' machine had Asterisk 1.4.4. installed. The complained of choppy Playback<br>> of gsm files.<br>> So scouring the internet gave me the solution of installing ztdummy and loading it as a module.<br>> Did it (using zaptel-1.4.1) , but to no effect. Re-compiled asterisk and re-installed. Sill<br>> no effect.<br>> <br>> Do I have to specify any parameter in the Asterisk compilation to look at ztdummy/rtc? As<br>> far as I remember (am
coming back to Asterisk after quite some time now), you don't really<br>> need to set anything over there for any zaptel specific compilation?<br>> <br>> And yes, all the files are gsm files and the codec used for the calls is ulaw.<br>> <br>> I even tried converting those gsm files to wav using sox and then playing them, but the<br>> behaviour is the same.<br>> <br>> Any ideas anyone.. something I am missing ??<br><br>Firstly, check whether Asterisk has chan_zap loaded and access to zaptel:<br><br>*CLI> zap show channels<br> Chan Extension Context Language MusicOnHold <br> pseudo default<br>*CLI><br><br>If you don't get pseudo shown, then you are not getting the benefit of<br>ztdummy.<br><br>However, the probably main cause of choppy sound is poor timing from the<br>SIP client (I'm assuming SIP), because Asterisk by default uses the incoming<br>stream to generate timing for the outbound stream.<br><br>There
are two main things to try:<br><br>1. Make sure that the SIP clients are NOT using silence suppression (may<br>be referred to as VAD, bandwidth saving, or something similar).<br><br>2. If ztdummy is running ok, edit /etc/asterisk/asterisk.conf and enable<br>the line "internal_timing=yes". That should make it play out based on<br>internal zaptel timing instead of timing off the incoming stream, I think.<br><br>Cheers<br>Tony<br>-- <br>Tony Mountifield<br>Work: tony@softins.co.uk - http://www.softins.co.uk<br>Play: tony@mountifield.org - http://tony.mountifield.org<br></ben4asterisk@yahoo.com></blockquote><br>Thanks Tony for the response.<br>zap show channels shows that things are fine, as you said :<br>*CLI> zap show channels<br> Chan Extension Context Language MOH Interpret <br> pseudo
default default<br><br>Tried setting internal_timing to yes as well. Still no difference.<br><br>Also, I don't think my SIP gateway uses Silence suppression, because the same SIP gateway connections work fine with another Asterisk server.<br><br>This is getting seriously irritating now!!! Have tried all the tricks and tips I've been finding on the net.<br><br>Yeah, btw, even Meetme playback is choppy. So, I think its somehow related to timing. But I am not the expert. <br><br>- Ben.<br><div> </div><hr size="1">Be a better friend, newshound, and know-it-all with Yahoo! Mobile. <a href="http://us.rd.yahoo.com/evt=51733/*http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ%20"> Try it now.</a>_______________________________________________<br>-- Bandwidth and Colocation Provided by http://www.api-digital.com --<br><br>asterisk-users mailing
list<br>To UNSUBSCRIBE or update options visit:<br> http://lists.digium.com/mailman/listinfo/asterisk-users</blockquote>Btw, I am on CentOS 5, with uname showing as:<br>Linux mserver.org 2.6.18-53.1.13.el5 #1 SMP Tue Feb 12 13:01:45 EST 2008 i686 i686 i386 GNU/Linux<br><br>And it is not a multiprocessor machine. Will the SMP option affect the working in any way?<br><br>- Ben.<br><p> 
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