rendered paste body
-- 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" <sip:530@MYIPADDRESS>;tag=as29d296ad
To: <sip:0211234567@SIPPROVIDER>
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" <sip:530@MYIPADDRESS>;tag=as29d296ad
To: <sip:0211234567@SIPPROVIDER>
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" <sip:530@MYIPADDRESS>;tag=as29d296ad
To: <sip:0211234567@SIPPROVIDER>
Contact: <sip:530@MYIPADDRESS>
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" <sip:530@MYIPADDRESS>;tag=as29d296ad
To: <sip:0211234567@SIPPROVIDER>
Contact: <sip:530@MYIPADDRESS>
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" <sip:530@MYIPADDRESS>;tag=as29d296ad
To: <sip:0211234567@SIPPROVIDER>
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" <sip:530@MYIPADDRESS>;tag=as29d296ad
To: <sip:0211234567@SIPPROVIDER>
Call-ID: 626bf04e6689f3804c88769f74e2d084@MYIPADDRESS
CSeq: 103 INVITE
Contact: <sip:0211234567@SIPPROVIDER>
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" <sip:530@MYIPADDRESS>;tag=as29d296ad
To: <sip:0211234567@SIPPROVIDER>
Contact: <sip:530@MYIPADDRESS>
Call-ID: 626bf04e6689f3804c88769f74e2d084@MYIPADDRESS
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0