<DIV>Hello Again,</DIV>
<DIV> </DIV>
<DIV>As you said It may be the problem with my CODEC which i configured in my SIP.CONF, so followoing is the code which i used for defining CODEC in SIP.CONF :</DIV>
<DIV> </DIV>
<DIV>disallow=all ; Disallow all codecs<BR>allow=gsm<BR>;allow=g723.1<BR>;allow=ulaw ; Allow codecs in order of preference<BR>;allow=alaw<BR>;allow=gsm<BR>;allow=ilbc </DIV>
<DIV>;allow=ilbc </DIV>
<DIV> </DIV>
<DIV>Waiting for positive Reply.</DIV>
<DIV> </DIV>
<DIV>Thanks and Regards,</DIV>
<DIV>Nilesh.</DIV>
<DIV> </DIV>
<DIV>My Previous mail was as follows :</DIV>
<DIV> </DIV>
<DIV>Hello,<BR>><BR>>I am New user on Asterisk.. I have some problems;;<BR>><BR>>When I called to another user from my user on soft phone, the call is correctly going, but when the other man receives the call and say "Hello", i can hear only first word and after that voice is not coming though call is going on.<BR>><BR>>Also when i checked some logs i got some Warning as follows :<BR>><BR>>WARNING : chan_sip.c:497 retrans_pkt: Maximum retries exceeded on call<BR>><BR>>WARNING : chan_iax2.c:5689 set_config: Ignoring port for now<BR>><BR>>And i want to ask you that what is mean by this error?<BR>><BR>>Transmitting (no NAT):<BR>>SIP/2.0 407 Proxy Authentication Required<BR>><BR>>Waiting for Positive Reply.<BR>><BR>>Thanks and Regards,<BR>>Nilesh<BR></DIV><p>
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