Displaying 1 result from an estimated 1 matches for "3e68de2444adcd12482755b9adb9b72a".
2005 Feb 20
0
SIP to SIP calls have no audio until put on hold and taken back off
...ea24231eb6f@192.168.201.111
CSeq: 22568 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:1010@192.168.200.100>
Content-Length: 0
to 192.168.201.111:5060
asterisk*CLI>
Sip read:
SIP/2.0 100 Trying
To: <sip:1010@192.168.201.110:5060>;tag=3e68de2444adcd12482755b9adb9b72a
Via: SIP/2.0/UDP 192.168.200.100:5060;branch=z9hG4bK065f8ed2
From: "1019" <sip:1019@192.168.200.100>;tag=as18eaba8f
Call-ID: 5414c15b4b8b646d388a196d27079fa1@192.168.200.100
CSeq: 102 INVITE
Content-Length: 0
Supported: sip-cc, sip-cc-01, replaces, timer
Allow: INVITE, ACK, CANCEL,...