Hi all, I was trying to use G.723.1 codec for my terminator as Pass through. But when the second party pickup phone the call is going dropted automatically with the following error: No path to translate from SIP/123456-fca7(1) to SIP/myterminator.com-ff11(4) Dec 1 10:54:39 WARNING[7480]: app_dial.c:1024 dial_exec: Had to drop call because I couldn't make SIP/123456-fca7 compatible with SIP/myterminator.com-ff11 Please advice me how i can make it work? -- Best Regards, Code Lover Nepal
> Please advice me how i can make it work?It looks like your Phone is not compatible to G.723.1 or this codec is disabled within sip.conf Elmar
Hi, My IP Phone is using well G.723.1 because when i am testing it with another SIP GK, working well with G.723.1. But the problem is only accuring in Asterisk, my sip.conf is already having the configuration of this codec. [123456] -------- -------- disallow=all allow=g723 -- Thank You, Code Lover
Hi, Do you know from where i can buy g723 codec. for g729 i can buy it from digium.com. But Please let me know from where i can get g723 codec. And the codecs purchasing can solved my problem? -- Thank You, Code Lover
On Fri, 2005-12-02 at 05:59, Code Lover wrote:> Hi, > > Do you know from where i can buy g723 codec. for g729 i can buy it > from digium.com. But Please let me know from where i can get g723 > codec. > > And the codecs purchasing can solved my problem? > > > -- > Thank You, > Code LoverBecome the fisherman... http://www.voip-info.org/wiki/index.php?page=Asterisk+G.723.1+Licensing
Code Lover wrote:> Hi all, > > I was trying to use G.723.1 codec for my terminator as Pass through. > But when the second party pickup phone the call is going dropted > automatically with the following error: > > No path to translate from SIP/123456-fca7(1) to SIP/myterminator.com-ff11(4) > Dec 1 10:54:39 WARNING[7480]: app_dial.c:1024 dial_exec: Had to drop > call because I couldn't make SIP/123456-fca7 compatible with > SIP/myterminator.com-ff11 > > > Please advice me how i can make it work?If you are getting that error then either Asterisk has to stay in the audio path (t or T and other options to Dial will make this happen), OR both legs of the call are not G723.1. In either situation Asterisk cannot operate in "pass thru" mode.
I have the license for G729, however I need to use a different codec for the prepaid service, but when the call is started I have this error Asked to transmit frame type 256, while native formats is 4 (read/write 4/4) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060402/0b36a62a/attachment.htm