Hi
i am new to ser,
iam going through the list,
i found some of the similar queries
I have used the config file
changed according to my setup
and iam able to pick up messages from Asterisk also
But i have some problem here is
1. i have registered user 8888 in my SER, when i dial 8888
its going to voice mail, how can i set the caller==calee Busy
even i have added this
if (!uri==myself) {
t_relay();
break;
};
But still no use.
2. when i dial to PSTN number, when the PSTN also go to voice mail after certain rings
I suppose to get the voice Menu of other Side Number, but the voice go Blank ( instead of going to VoiceMenu)
when checked in my Log, i get this error in my syslog
Aug 28 18:33:55 router ser[757]: no location or no user
Aug 28 18:34:01 router ser[751]: 10 digit exp match w/leading 1
Aug 28 18:34:01 router ser[751]: route[2]:SIP-to-PSTN call routed
Aug 28 18:34:10 router ser[757]: no location or no user
Aug 28 18:34:25 router ser[758]: no location or no user
Aug 28 18:34:28 router ser[791]: ACC: call missed: from=88888 <
sip:88888@mydomain.com>;tag=1674409670, i-uri=
sip:1xxxxx80369@mydomain.com, method=INVITE, o-uri=
sip:1xxxxx80369@myprovider:5060, code=408 Request Timeout
Aug 28 18:34:28 router ser[791]: failure_route[5]
Aug 28 18:34:28 router ser[791]: route[5]: re-relay to PSTN
Aug 28 18:34:28 router ser[751]: ERROR: t_should_relay: status rewrite by UAS: stored: 408, received: 487
Aug 28 18:34:37 router ser[758]: 10 digit exp match w/leading 1
Aug 28 18:34:37 router ser[758]: route[2]:SIP-to-PSTN call routed
Aug 28 18:34:37 router ser[758]: 10 digit exp match w/leading 1
Aug 28 18:34:37 router ser[758]: route[2]:SIP-to-PSTN call routed
Aug 28 18:34:40 router ser[751]: no location or no user
any suggestions
Ram