Here is a matrix we put together about g729 license needs: ======================== ====================== ========================= ====== ======= ======== ======== Asterisk to SIP Provider SIP Client to Asterisk asterisk.conf sln defined record monitor encoders decoders ======================== ====================== ========================= ====== ======= ======== ======== ulaw ulaw yes yes yes 0 0 ulaw ulaw yes yes no 0 0 ulaw ulaw yes no no 0 0 ulaw ulaw yes no yes 0 0
ulaw ulaw no yes yes 0 0 ulaw ulaw no yes no 0 0 ulaw ulaw no no no 0 0 ulaw ulaw no no yes 0 0 ulaw g729 yes yes yes 3 3 ulaw g729 yes yes no 2 3 ulaw g729 yes no no 1 1 ulaw g729 yes no yes 3 3 ulaw g729 no yes yes 3 3 ulaw g729 no yes no 2 3 ulaw g729 no no no 1 1 ulaw g729 no no yes 3 3 g729 ulaw yes yes yes 2 5 g729 ulaw yes yes no 2 5 g729 ulaw yes no no 1 1 g729 ulaw yes no yes 2 3 g729 ulaw no yes yes 2 5 g729 ulaw no yes no 2 5 g729 ulaw no no no 1 1 g729 ulaw no no yes 2 3 g729 g729 yes yes yes 4 7 g729 g729 yes yes no 3 7 g729 g729 yes no no 1 1 g729 g729 yes no yes 4 5 g729 g729 no yes yes 4 7 g729 g729 no yes no 3 7 g729 g729 no no no 1 1 g729 g729 no no yes 4 5 -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Jan 12, 2012, at 3:00 PM, Kevin P. Fleming wrote: > On 01/12/2012 11:57 AM, Daniel - Asterisk wrote: >> The simplest answer, I purchased one additional license and one >> simultaneous call is being recorded now. I do not understand why the >> ulaw codec (or format) is involved here (... No translator path from >> alaw to unknown ...) >> >> Any entry will be very appreciated. > > 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. > > -- > Kevin P. Fleming > Digium, Inc. | Director of Software Technologies > Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming > 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA > Check us out at www.digium.com & www.asterisk.org > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users