[Asterisk-Users] CDR for PSTN

[Kamran Ahmad]

hi list

i am using Quintum gw for pstn. sip---->PSTN call
when i iniating call quintum is replying me 
"183 Session Progress"

asterisk starts calculating CDR

actually it should start from "when both side starts
RTP after 200 ok and ACK.

if callee (PSTN) receives call after 10 seconds these
10 sec are included in CDR

Sip read: from softphone INVITE sip
Transmitting (NAT): to softphone SIP/2.0 100 Trying
Reliably Transmitting: to quintum INVITE 
Sip read: from quintum SIP/2.0 183 Session Progress
Transmitting (NAT): to softphone SIP/2.0 183 Session
Progress
Sip read: from quintum SIP/2.0 180 Ringing
Transmitting (NAT): to softphone SIP/2.0 180 Ringing
Sip read: from quintum SIP/2.0 200 OK
Transmitting: to quintum ACK 
Reliably Transmitting (NAT): to softphone SIP/2.0 200
OK
Sip read: from softphone ACK

12 headers, 0 lines
RFC3389: 4 bytes, level 8...
RFC3389: 4 bytes, level 8...
RFC3389: 4 bytes, level 8...
RFC3389: 4 bytes, level 8...
RFC3389: 4 bytes, level 8...
RFC3389: 4 bytes, level 8...
                                                      
                                       
Sip read: from quintum BYE 

Transmitting (NAT): to quintum SIP/2.0 200 OK

Reliably Transmitting: to softphoen BYE 

Sip read: from softphone SIP/2.0 200 OK




		
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