Code:
<--- SIP read from UDP:192.168.10.1:6666 --->
INVITE sip:915141111234@192.168.10.130:15060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.17:6666;rport;branch=z9hG4bK31625
From: <sip:1001@192.168.10.130:15060>;tag=30407
To: <sip:915141111234@192.168.10.130:15060>
Call-ID: 5259
CSeq: 20 INVITE
Contact: <sip:1001@192.168.10.1:6666>
Content-Type: application/sdp
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Max-Forwards: 70
User-Agent: Linphone/3.3.2 (eXosip2/3.3.0)
Subject: Phone call
Content-Length: 404
v=0
o=1001 123456 654321 IN IP4 192.168.10.1
s=A conversation
c=IN IP4 192.168.10.1
t=0 0
m=audio 7078 RTP/AVP 112 111 110 3 0 8 101
a=rtpmap:112 speex/32000/1
a=fmtp:112 vbr=on
a=rtpmap:111 speex/16000/1
a=fmtp:111 vbr=on
a=rtpmap:110 speex/8000/1
a=fmtp:110 vbr=on
a=rtpmap:3 GSM/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-11
<------------->
--- (13 headers 17 lines) ---
== Using UDPTL CoS mark 5
Sending to 192.168.10.1:6666 (no NAT)
Using INVITE request as basis request - 5259
Found peer '1001' for '1001' from 192.168.10.1:6666
<--- Reliably Transmitting (NAT) to 192.168.10.1:6666 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.17:6666;branch=z9hG4bK31625;received=192.168.10.1;rport=6666
From: <sip:1001@192.168.10.130:15060>;tag=30407
To: <sip:915141111234@192.168.10.130:15060>;tag=as23a2df50
Call-ID: 5259
CSeq: 20 INVITE
Server: Asterisk PBX SVN-branch-1.8-r293530
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1924f809"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '5259' in 6400 ms (Method: INVITE)
<--- SIP read from UDP:192.168.10.1:6666 --->
ACK sip:915141111234@192.168.10.130:15060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.17:6666;rport;branch=z9hG4bK31625
From: <sip:1001@192.168.10.130:15060>;tag=30407
To: <sip:915141111234@192.168.10.130:15060>;tag=as23a2df50
Call-ID: 5259
CSeq: 20 ACK
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
<--- SIP read from UDP:192.168.10.1:6666 --->
INVITE sip:915141111234@192.168.10.130:15060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.17:6666;rport;branch=z9hG4bK23771
From: <sip:1001@192.168.10.130:15060>;tag=30407
To: <sip:915141111234@192.168.10.130:15060>
Call-ID: 5259
CSeq: 21 INVITE
Contact: <sip:1001@192.168.10.1:6666>
Authorization: Digest username="1001", realm="asterisk", nonce="1924f809", uri="sip:915141111234@192.168.10.130:15060", response="0ae561d027e685717da46511cfbcbf13", algorithm=MD5
Content-Type: application/sdp
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Max-Forwards: 70
User-Agent: Linphone/3.3.2 (eXosip2/3.3.0)
Subject: Phone call
Content-Length: 404
v=0
o=1001 123456 654321 IN IP4 192.168.10.1
s=A conversation
c=IN IP4 192.168.10.1
t=0 0
m=audio 7078 RTP/AVP 112 111 110 3 0 8 101
a=rtpmap:112 speex/32000/1
a=fmtp:112 vbr=on
a=rtpmap:111 speex/16000/1
a=fmtp:111 vbr=on
a=rtpmap:110 speex/8000/1
a=fmtp:110 vbr=on
a=rtpmap:3 GSM/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-11
<------------->
--- (14 headers 17 lines) ---
Sending to 192.168.10.1:6666 (NAT)
Using INVITE request as basis request - 5259
Found peer '1001' for '1001' from 192.168.10.1:6666
== Using SIP RTP CoS mark 5
Found RTP audio format 112
Found RTP audio format 111
Found RTP audio format 110
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format speex for ID 112
Found audio description format speex for ID 111
Found audio description format speex for ID 110
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x20000021e (gsm|ulaw|alaw|speex|speex16|g726aal2)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.10.1:7078
Looking for 915141111234 in Interne (domain 192.168.10.130:15060)
list_route: hop: <sip:1001@192.168.10.1:6666>
<--- Transmitting (NAT) to 192.168.10.1:6666 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.17:6666;branch=z9hG4bK23771;received=192.168.10.1;rport=6666
From: <sip:1001@192.168.10.130:15060>;tag=30407
To: <sip:915141111234@192.168.10.130:15060>
Call-ID: 5259
CSeq: 21 INVITE
Server: Asterisk PBX SVN-branch-1.8-r293530
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:915141111234@192.168.10.130:15060>
Content-Length: 0
<------------>
-- Executing [915141111234@Interne:1] Log("SIP/1001-00000019", "NOTICE, Dialing out from "SoftPhone" <1001> to 15141111234 through Babytel Provider") in new stack
[Nov 3 20:09:58] NOTICE[2799]: Ext. 915141111234:1 @ Interne: Dialing out from "SoftPhone" <1001> to 15141111234 through Babytel Provider
-- Executing [915141111234@Interne:2] Dial("SIP/1001-00000019", "SIP/BabyTEL/15141111234,50") in new stack
== Using UDPTL CoS mark 5
== Using SIP RTP CoS mark 5
Audio is at 15060
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 216.18.125.12:5065:
INVITE sip:15141111234@216.18.125.12:5065 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.130:15060;branch=z9hG4bK190b332f;rport
Max-Forwards: 70
From: "SoftPhone" <sip:1001@216.18.125.12>;tag=as63526dd5
To: <sip:15141111234@216.18.125.12:5065>
Contact: <sip:1001@192.168.10.130:15060>
Call-ID: 64c18d840ecdc7f124098b426c7cfd08@216.18.125.12
CSeq: 102 INVITE
User-Agent: Asterisk PBX SVN-branch-1.8-r293530
Date: Thu, 04 Nov 2010 03:09:58 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 281
v=0
o=root 1215646287 1215646287 IN IP4 192.168.10.130
s=Asterisk PBX SVN-branch-1.8-r293530
c=IN IP4 192.168.10.130
t=0 0
m=audio 7018 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv