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. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130602/5a4997f1/attachment.htm>
On 2/6/13 2:01 pm, Muhammad Yousuf wrote:> 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 1.8.11.0Isn't g723.1 considered pretty poor quality these days? Can't you set voipswitch to use something apart from that? Kind regards, Chris -- This email is made from 100% recycled electrons