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