<div dir="ltr"><br><div class="gmail_extra"><br><br><div class="gmail_quote">On Mon, May 6, 2013 at 5:51 AM, thomas peterseil <span dir="ltr"><<a href="mailto:thomas.peterseil@gmail.com" target="_blank">thomas.peterseil@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"><div>thank you very much for your help, but they are enabled in the vars.xml<br><br><X-PRE-PROCESS cmd="set" data="global_codec_prefs=OPUS,SILK@24000h,SILK@16000h,SILK@8000h,G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM"/><br>
<X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=OPUS,SILK@24000h,SILK@16000h,SILK@8000h,G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM"/><br></div></div></blockquote><div><br></div><div>Do you want OPUS, SILK, and G7221 to be preferred over G722 and PCMU? Just curious because you have all of those listed before the two codecs you actually meant to use. In any case, please get the debug log for the call between the phones and put it on <a href="http://pastebin.freeswitch.org">pastebin.freeswitch.org</a> so that we can see what's happening.<br>
<br></div><div>-MC<br><br></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><div><br></div><div>any other suggestions?<br><br>thanks,<br>
<br>thomas<br></div></div><div class="HOEnZb"><div class="h5"><div class="gmail_extra"><br><br><div class="gmail_quote">2013/5/6 Blake Priddy <span dir="ltr"><<a href="mailto:bpriddy@bryantschools.org" target="_blank">bpriddy@bryantschools.org</a>></span><br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><p dir="ltr">Maybe you need to list those in your vars.xml </p>
<div class="gmail_quote"><div><div>On May 6, 2013 7:12 AM, "thomas peterseil" <<a href="mailto:thomas.peterseil@gmail.com" target="_blank">thomas.peterseil@gmail.com</a>> wrote:<br type="attribution">
</div></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div>
<div dir="ltr"><div><div><div>hello,<br></div>i am running a new FS and i have two sip clients registered, one which supports G.722 and the other one supports G.711. i am still running mostly the default config, both codecs are listed when i do a show codecs. when i call now from 1000 to 1001 with different codecs it doesnt work. but i can call from both clients with different codecs to a 3300 conference.<br>
</div>is there something i have to enable in the default config to activate transcoding?<br><br></div>thanks for your help!<br><br>thomas<br></div>
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