Displaying 1 result from an estimated 1 matches for "thorsten_sip_for_test".
2013 Sep 26
0
Asterisk / SIP-Call / AGI-Script / SIGHUP after Answer
...22), peer -
audio=(ulaw|alaw|speex16|g722|speex32)/video=(nothing)/text=(nothing),
combined - (ulaw|alaw|g722)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1
(telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.2:10054
Looking for 3 in thorsten_sip_for_testing (domain myhost.org)
list_route: hop: <sip:03794281 at 192.168.1.2:51861>
<--- Transmitting (NAT) to 217.92.105.86:51861 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
192.168.1.2:51861;branch=z9hG4bKPj71edb9caa0e84a52b14777e7d949bc2a;received=217.92.105.86;rport=51861
From: "Thorste...