I am trying to use asterisk as transcoder between voipswitch 2.0 and gsm gateway. Voipswitch supports g723.1 but gsm gateway does not. Now I have g723.1 codec in my asterisk. call leg from voipswitch is using codec g723.1 and call leg from gsm gateway is using codec gsm. I am having one way audio and getting below mentioned warning. Asterisk version is 1.8.11.0
[Jun 2 17:08:28] WARNING[21652]: translate.c:162 framein: no samples for g723tolin [Jun 2 17:08:28] WARNING[21652]: translate.c:162 framein: no samples for g723tolin [Jun 2 17:08:28] WARNING[21652]: translate.c:162 framein: no samples for g723tolin [Jun 2 17:08:28] WARNING[21652]: translate.c:162 framein: no samples for plz help what could be the issue.
-- _____________________________________________________________________ -- 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
