Hello Again,
As you said It may be the problem with CODEC which i configured in my SIP.CONF.
I used followoing code for CODEC in SIP>CONF file :
disallow=all ; Disallow all codecs
allow=gsm
;allow=g723.1
;allow=ulaw ; Allow codecs in order of preference
;allow=alaw
;allow=gsm
;allow=ilbc
;allow=ilbc
Waiting for Positive Reply.
Thanks and Regards,
Nilesh
=====================Hi,
It may be the problem of the CODECS that you are using in your configuration.
Verify your codecs.
On Mon, 09 Aug 2004 Nilesh sonavani wrote :>Hello,
>
>I am New user on Asterisk.. I have some problems;;
>
>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.
>
>Also when i checked some logs i got some Warning as follows :
>
>WARNING : chan_sip.c:497 retrans_pkt: Maximum retries exceeded on call
>
>WARNING : chan_iax2.c:5689 set_config: Ignoring port for now
>
>And i want to ask you that what is mean by this error?
>
>Transmitting (no NAT):
>SIP/2.0 407 Proxy Authentication Required
>
>Waiting for Positive Reply.
>
>Thanks and Regards,
>Nilesh
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