Displaying 1 result from an estimated 1 matches for "z9hg4bk7f41a807".
2007 Jun 25
1
Asterisk 1.4.5, Cisco 7960, call dropped when sip client put on hold/transfer
...nation: set destination to 192.168.96.16, port 5060
Audio is at 192.168.96.5 port 16816
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.96.16:5060:
INVITE sip:302 at 192.168.96.16:5060;user=phone;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.96.5:5060;branch=z9hG4bK7f41a807;rport
From: <sip:304 at 192.168.96.5;user=phone>;tag=as2c9302cc
To: "User Name 1" <sip:302 at 192.168.96.5>;tag=001200347d27001a7e20b127-2129053d
Contact: <sip:304 at 192.168.96.5>
Call-ID: 00120034-7d27000f-5f89ed49-5278b958 at 192.168.96.16
CSeq: 103 INVITE
User-Agent:...