<div>Yes, a 'call' refers to two channels bridged. </div><div> </div><div>Jim, please help me to undertand the numbers. I have two g729 licenses, my SIP provider uses only g729 and my softphones support g729 too, asterisk.conf is set in its default value (sln).</div>
<div> </div><div>When a call (2 channels) is being made and succesfully recorded with MixMonitor (wav49 format), I see at CLI:</div><div> </div><div>testpbx*CLI> sip show channels<br>Peer User/ANR Call ID Format Hold Last Message Expiry Peer<br>
A.B.C.D 987654321 63ffff9237c5976 0x100 (g729) No Tx: ACK sip-provider1<br>W.X.Y.Z elder 4e4adc85-b2e21c 0x100 (g729) No Rx: ACK elder</div>
<div> </div><div>testpbx*CLI> g729 show licenses<br>0/2 encoders/decoders of 2 licensed channels are currently in use</div><div>Licenses Found:<br>File: G729-... -- Key: G729-...-- Host-ID: ... -- Channels: 1 (Expires: 20...) (OK)<br>
File: G729-... -- Key: G729-...-- Host-ID: ... -- Channels: 1 (Expires: 20...) (OK)<br></div><div class="gmail_quote">Thanks for your answers,</div><div class="gmail_quote"> </div><div class="gmail_quote">Elder</div><div class="gmail_quote">
</div><div class="gmail_quote"> </div><div class="gmail_quote">On Thu, Jan 12, 2012 at 6:05 PM, Jim Dickenson <span dir="ltr"><<a href="mailto:dickenson@cfmc.com">dickenson@cfmc.com</a>></span> wrote:<br><blockquote style="margin:0px 0px 0px 0.8ex;padding-left:1ex;border-left-color:rgb(204,204,204);border-left-width:1px;border-left-style:solid" class="gmail_quote">
Here is a matrix we put together about g729 license needs:<br>
<br>
======================== ====================== ========================= ====== ======= ======== ========<br>
Asterisk to SIP Provider SIP Client to Asterisk asterisk.conf sln defined record monitor encoders decoders<br>
======================== ====================== ========================= ====== ======= ======== ========<br>
ulaw ulaw yes yes yes 0 0<br>
ulaw ulaw yes yes no 0 0<br>
ulaw ulaw yes no no 0 0<br>
ulaw ulaw yes no yes 0 0<br>
<br>
ulaw ulaw no yes yes 0 0<br>
ulaw ulaw no yes no 0 0<br>
ulaw ulaw no no no 0 0<br>
ulaw ulaw no no yes 0 0<br>
<br>
ulaw g729 yes yes yes 3 3<br>
ulaw g729 yes yes no 2 3<br>
ulaw g729 yes no no 1 1<br>
ulaw g729 yes no yes 3 3<br>
<br>
ulaw g729 no yes yes 3 3<br>
ulaw g729 no yes no 2 3<br>
ulaw g729 no no no 1 1<br>
ulaw g729 no no yes 3 3<br>
<br>
g729 ulaw yes yes yes 2 5<br>
g729 ulaw yes yes no 2 5<br>
g729 ulaw yes no no 1 1<br>
g729 ulaw yes no yes 2 3<br>
<br>
g729 ulaw no yes yes 2 5<br>
g729 ulaw no yes no 2 5<br>
g729 ulaw no no no 1 1<br>
g729 ulaw no no yes 2 3<br>
<br>
g729 g729 yes yes yes 4 7<br>
g729 g729 yes yes no 3 7<br>
g729 g729 yes no no 1 1<br>
g729 g729 yes no yes 4 5<br>
<br>
g729 g729 no yes yes 4 7<br>
g729 g729 no yes no 3 7<br>
g729 g729 no no no 1 1<br>
g729 g729 no no yes 4 5<br>
<span class="HOEnZb"><font color="#888888"><br>
--<br>
Jim Dickenson<br>
mailto:<a href="mailto:dickenson@cfmc.com">dickenson@cfmc.com</a><br>
<br>
CfMC<br>
<a href="http://www.cfmc.com/" target="_blank">http://www.cfmc.com/</a><br>
</font></span><div class="HOEnZb"><div class="h5"><br>
<br>
<br>
On Jan 12, 2012, at 3:00 PM, Kevin P. Fleming wrote:<br>
<br>
> On 01/12/2012 11:57 AM, Daniel - Asterisk wrote:<br>
>> The simplest answer, I purchased one additional license and one<br>
>> simultaneous call is being recorded now. I do not understand why the<br>
>> ulaw codec (or format) is involved here (... No translator path from<br>
>> alaw to unknown ...)<br>
>><br>
>> Any entry will be very appreciated.<br>
><br>
> When you say 'call', do you mean a call between two phones (endpoints)? If so, and both endpoints are using G.729 for audio, then yes, recording that call in any format other than G.729 will require *two* G.729 decoders, one for each audio stream being received by Asterisk. Even in a case where you are only recording the combined audio from the two phones (MixMonitor), the audio must still be decoded in order to be mixed.<br>
><br>
> --<br>
> Kevin P. Fleming<br>
> Digium, Inc. | Director of Software Technologies<br>
> Jabber: <a href="mailto:kfleming@digium.com">kfleming@digium.com</a> | SIP: <a href="mailto:kpfleming@digium.com">kpfleming@digium.com</a> | Skype: kpfleming<br>
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA<br>
> Check us out at <a href="http://www.digium.com" target="_blank">www.digium.com</a> & <a href="http://www.asterisk.org" target="_blank">www.asterisk.org</a><br>
><br>
> --<br>
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</div></div></blockquote></div><br>