-- Called 0211234567@broadtalk <--- SIP read from SIPPROVIDER:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP MYIPADDRESS;branch=z9hG4bK6d99f0a7;rport=z9hG4bK6d99f0a7 From: "Me Test" ;tag=as29d296ad To: Call-ID: 626bf04e6689f3804c88769f74e2d084@MYIPADDRESS CSeq: 102 INVITE Content-Length: 0 <-------------> --- (7 headers 0 lines) --- <--- SIP read from SIPPROVIDER:5060 ---> SIP/2.0 407 Proxy authentication required Via: SIP/2.0/UDP MYIPADDRESS;branch=z9hG4bK6d99f0a7;rport=z9hG4bK6d99f0a7 From: "Me Test" ;tag=as29d296ad To: Call-ID: 626bf04e6689f3804c88769f74e2d084@MYIPADDRESS CSeq: 102 INVITE Proxy-Authenticate: Digest realm="SIPPROVIDER.DOM", nonce="64acd77224db44305c390ff79cde7e6d", stale=FALSE, algorithm=MD5 Content-Length: 0 <-------------> --- (8 headers 0 lines) --- Transmitting (no NAT) to SIPPROVIDER:5060: ACK sip:0211234567@SIPPROVIDER SIP/2.0 Via: SIP/2.0/UDP MYIPADDRESS:5060;branch=z9hG4bK6d99f0a7;rport From: "Me Test" ;tag=as29d296ad To: Contact: Call-ID: 626bf04e6689f3804c88769f74e2d084@MYIPADDRESS CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 --- Audio is at MYIPADDRESS port 10486 Adding codec 0x2 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP Reliably Transmitting (no NAT) to SIPPROVIDER:5060: INVITE sip:0211234567@SIPPROVIDER SIP/2.0 Via: SIP/2.0/UDP MYIPADDRESS:5060;branch=z9hG4bK3d5c4721;rport From: "Me Test" ;tag=as29d296ad To: Contact: Call-ID: 626bf04e6689f3804c88769f74e2d084@MYIPADDRESS CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Proxy-Authorization: Digest username="myusername", realm="SIPPROVIDER.DOM", algorithm=MD5, uri="sip:0211234567@SIPPROVIDER", nonce="64acd77224db44305c390ff79cde7e6d", response="959044ebd35ada3fac586edf65cf9c87", opaque="" Date: Thu, 03 May 2007 22:25:38 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 239 v=0 o=root 51323 51324 IN IP4 MYIPADDRESS s=session c=IN IP4 MYIPADDRESS t=0 0 m=audio 10486 RTP/AVP 3 101 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- <--- SIP read from SIPPROVIDER:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP MYIPADDRESS;branch=z9hG4bK3d5c4721;rport=z9hG4bK3d5c4721 From: "Me Test" ;tag=as29d296ad To: Call-ID: 626bf04e6689f3804c88769f74e2d084@MYIPADDRESS CSeq: 103 INVITE Content-Length: 0 <-------------> --- (7 headers 0 lines) --- <--- SIP read from SIPPROVIDER:5060 ---> SIP/2.0 406 Not acceptable Via: SIP/2.0/UDP MYIPADDRESS;branch=z9hG4bK3d5c4721;rport=z9hG4bK3d5c4721 From: "Me Test" ;tag=as29d296ad To: Call-ID: 626bf04e6689f3804c88769f74e2d084@MYIPADDRESS CSeq: 103 INVITE Contact: Content-Length: 0 <-------------> --- (8 headers 0 lines) --- -- Got SIP response 406 "Not acceptable" back from SIPPROVIDER Transmitting (no NAT) to SIPPROVIDER:5060: ACK sip:0211234567@SIPPROVIDER SIP/2.0 Via: SIP/2.0/UDP MYIPADDRESS:5060;branch=z9hG4bK3d5c4721;rport From: "Me Test" ;tag=as29d296ad To: Contact: Call-ID: 626bf04e6689f3804c88769f74e2d084@MYIPADDRESS CSeq: 103 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0