Dial started: -- Executing GotoIf("Local/1@ps_out-6dfa,2", "2?ps_out_pstn|1|1") in new stack -- Goto (ps_out_pstn,1,1) -- Executing Dial("Local/1@ps_out-6dfa,2", "SIP/0079139257974@vf-out|60000") in new stack We're at 195.224.39.103 port 11622 Video is at 195.224.39.103 port 15468 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP 14 headers, 11 lines Reliably Transmitting (no NAT) to 195.189.173.11:5060: INVITE sip:0079139257974@voipfone.co.uk SIP/2.0 ... -- Called 0079139257974@vf-out --- (11 headers 0 lines)--- Transmitting (no NAT) to 195.189.173.11:5060: ACK sip:0079139257974@voipfone.co.uk SIP/2.0 ... --- We're at 195.224.39.103 port 11622 Video is at 195.224.39.103 port 15468 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to 195.189.173.11:5060: INVITE sip:0079139257974@voipfone.co.uk SIP/2.0 CSeq: 103 INVITE ... call2ru*CLI> <-- SIP read from 195.189.173.11:5060: SIP/2.0 100 Trying now got cellphone ringing, after a bit: call2ru*CLI> <-- SIP read from 195.189.173.11:5060: SIP/2.0 183 Session Progress ... i'm hangup call: nothing changed. after some time: Scheduling destruction of call '4d7257b6778be9e725977af276324f8d@195.224.39.103' in 32000 ms Reliably Transmitting (no NAT) to 195.189.173.11:5060: CANCEL sip:0079139257974@voipfone.co.uk SIP/2.0 .... call2ru*CLI> <-- SIP read from 195.189.173.11:5060: SIP/2.0 487 Request Terminated --- (10 headers 0 lines)--- Transmitting (no NAT) to 195.189.173.11:5060: ACK sip:0079139257974@voipfone.co.uk SIP/2.0 --- call2ru*CLI> <-- SIP read from 195.189.173.11:5060: SIP/2.0 200 OK