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 __________________________________ Yahoo! Mail Mobile Take Yahoo! Mail with you! Check email on your mobile phone. http://mobile.yahoo.com/learn/mail _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users