Displaying 1 result from an estimated 1 matches for "as18eaba8f".
2005 Feb 20
0
SIP to SIP calls have no audio until put on hold and taken back off
...ring with capability 0x8 (alaw)
Answering with non-codec capability 0x1 (telephone-event)
12 headers, 12 lines
Reliably Transmitting:
INVITE sip:1010@192.168.201.110:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.200.100:5060;branch=z9hG4bK065f8ed2
From: "1019" <sip:1019@192.168.200.100>;tag=as18eaba8f
To: <sip:1010@192.168.201.110:5060>
Contact: <sip:1019@192.168.200.100>
Call-ID: 5414c15b4b8b646d388a196d27079fa1@192.168.200.100
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sun, 20 Feb 2005 02:01:59 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
C...