there is no special requiremnt to use g.729 but day to day my sip client incressing thats why some time i got breaking voice or voice quality not much better i think in LAN there is lots of brodcat on lan <br><br>if i purches g.729 transcoder license for asterisk to convert g.729 to g.711 then it will work or not<br><br>but why i need codec on trunk ????<br><br><b><i>Anselm Martin Hoffmeister <anselm@hoffmeister-online.de></i></b> wrote:<blockquote class="replbq" style="border-left: 2px solid rgb(16, 16, 255); margin-left: 5px; padding-left: 5px;"> Am Dienstag, den 23.10.2007, 02:56 -0700 schrieb satish patel:<br>> Dear all<br>> <br>> i have asterisk connected with avaya through E1 back-2-back<br>> now when i configure my sip client with g.729 codec then i m not able<br>> to put call from asterisk to avaya and when i user g.711 it is working<br>> fine so i dont know why i need G.729 on E1 Trunk it is TDM<br>> technologies then why
my call fail in g.729 case <br><br>Hi Satish,<br><br>Neither do I know why you _need_ G.729. Are there any specific reasons<br>why you do not want to use G711 in the sip client, which is "working<br>fine"? (Nota bene: there are some more codecs supported by asterisk,<br>some of which may be also supported by your sip phone)<br><br>Your E1 trunk obviously is G711-only - this is to be expected, because<br>the G711 wave samples are those which go over the wire (as time-division<br>multiplexed bitstream).<br><br>Together with the information from<br>http://www.voip-info.org/wiki-Asterisk+G.729+Licensing<br><br>namely<br>** G.729 requires a license per channel unless it is used<br>** in pass-thru mode.<br><br>which exactly matches your setup (by the way that was the first google<br>match for "g729 asterisk") we can guess that you did not buy the license<br>which would be necessary for asterisk to transcode G729/G711.<br><br>>
[sip_phone]------[asterisk]-----E1----[Avaya]----[analog_phone]<br>> <br>> Asterisk sip client configure with g.711 alaw/ulaw<br>> Avaya phone client configure g.711 alaw/ulaw<br>> <br>> suggest how do it implement g.729 on this case what change i have to<br>> done on both part<br><br>Avaya / E1 stays as is, sip client stays as is, your credit card data is<br>transferred to digium, and their license goes into the appropriate file<br>on your asterisk machine hard drive.<br><br>Others may have real world experience with those steps, but that is what<br>I read on this mailing list.<br><br>YMMV,<br>Anselm<br><br><br><br>_______________________________________________<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<br></blockquote><br><p> __________________________________________________<br>Do
You Yahoo!?<br>Tired of spam? Yahoo! Mail has the best spam protection around <br>http://mail.yahoo.com