I face one different issue in asterisk server. The issue is, once i establish a session between 2 user with asterisk using sipcommunicator the audio is not working in one side. That is the receiver is not able to hear audio. Actually the GSM to PCMU transcoding is happen here. In sip config file i gave allow=all. Is asterisk has any issue with GSM to PCMU transcoding ?. Then i put one fix in sip.conf file. The fix as follows
Now the issue is solved. Here the transcoding is not happened. Is anyone facing this issue before? Please help me. Is this asterisk issue ?
Thanks in advance