rendered paste body<--- SIP read from 172.101.1.177:5060 --->
INVITE sip:6338@172.101.1.253 SIP/2.0
From: "6336"<sip:6336@172.101.1.253>;tag=3ec790-b10165ac-13c4-30fe-5e7f131c-30fe
To: <sip:6338@172.101.1.253>
Call-ID: 414c18-b10165ac-13c4-30fe-5963bb98-30fe@172.101.1.253
CSeq: 1 INVITE
Via: SIP/2.0/UDP 172.101.1.177:5060;rport;branch=z9hG4bK-30fe-bf6149-7878ea3e
Max-Forwards: 70
Supported: replaces
Allow: INVITE, ACK, BYE, REFER, NOTIFY, CANCEL
User-Agent: WIP330
Contact: <sip:6336@172.101.1.177:5060;transport=UDP>
Content-Type: application/sdp
Content-Length: 155
v=0
o=rtp/7 11855626630 11855626630 IN IP4 172.101.1.177
s=-
c=IN IP4 172.101.1.177
t=0 0
m=audio 2070 RTP/AVP 8 0 18
a=ptime:20
a=SilenceSupp:off
<------------->
--- (13 headers 8 lines) ---
Sending to 172.101.1.177 : 5060 (NAT)
Using INVITE request as basis request - 414c18-b10165ac-13c4-30fe-5963bb98-30fe@172.101.1.253
Found peer '6336'
voip*CLI>
<--- Reliably Transmitting (no NAT) to 172.101.1.177:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 172.101.1.177:5060;branch=z9hG4bK-30fe-bf6149-7878ea3e;received=172.101.1.177;rport=5060
From: "6336"<sip:6336@172.101.1.253>;tag=3ec790-b10165ac-13c4-30fe-5e7f131c-30fe
To: <sip:6338@172.101.1.253>;tag=as71f9bab9
Call-ID: 414c18-b10165ac-13c4-30fe-5963bb98-30fe@172.101.1.253
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="26a0c509"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '414c18-b10165ac-13c4-30fe-5963bb98-30fe@172.101.1.253' in 31040 ms (Method: INVITE)
voip*CLI>
<--- SIP read from 172.101.1.177:5060 --->
INVITE sip:6338@172.101.1.253 SIP/2.0
From: "6336"<sip:6336@172.101.1.253>;tag=3ec790-b10165ac-13c4-30fe-5e7f131c-30fe
To: <sip:6338@172.101.1.253>
Call-ID: 414c18-b10165ac-13c4-30fe-5963bb98-30fe@172.101.1.253
CSeq: 1 INVITE
Via: SIP/2.0/UDP 172.101.1.177:5060;rport;branch=z9hG4bK-30fe-bf6149-7878ea3e
Max-Forwards: 70
Supported: replaces
Allow: INVITE, ACK, BYE, REFER, NOTIFY, CANCEL
User-Agent: WIP330
Contact: <sip:6336@172.101.1.177:5060;transport=UDP>
Content-Type: application/sdp
Content-Length: 155
v=0
o=rtp/7 11855626630 11855626630 IN IP4 172.101.1.177
s=-
c=IN IP4 172.101.1.177
t=0 0
m=audio 2070 RTP/AVP 8 0 18
a=ptime:20
a=SilenceSupp:off
<------------->
--- (13 headers 8 lines) ---
Ignoring this INVITE request
voip*CLI>
<--- SIP read from 172.101.1.177:5060 --->
ACK sip:6338@172.101.1.253 SIP/2.0
From: "6336"<sip:6336@172.101.1.253>;tag=3ec790-b10165ac-13c4-30fe-5e7f131c-30fe
To: <sip:6338@172.101.1.253>;tag=as71f9bab9
Call-ID: 414c18-b10165ac-13c4-30fe-5963bb98-30fe@172.101.1.253
CSeq: 1 ACK
Via: SIP/2.0/UDP 172.101.1.177:5060;rport;branch=z9hG4bK-30fe-bf6149-7878ea3e
Max-Forwards: 70
User-Agent: WIP330
Contact: <sip:6336@172.101.1.177:5060;transport=UDP>
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
voip*CLI>
<--- SIP read from 172.101.1.177:5060 --->
INVITE sip:6338@172.101.1.253 SIP/2.0
From: "6336"<sip:6336@172.101.1.253>;tag=3ec790-b10165ac-13c4-30fe-5e7f131c-30fe
To: <sip:6338@172.101.1.253>
Call-ID: 414c18-b10165ac-13c4-30fe-5963bb98-30fe@172.101.1.253
CSeq: 2 INVITE
Via: SIP/2.0/UDP 172.101.1.177:5060;rport;branch=z9hG4bK-30fe-bf6374-2dc8f67a
Max-Forwards: 70
Supported: replaces
Allow: INVITE, ACK, BYE, REFER, NOTIFY, CANCEL
User-Agent: WIP330
Contact: <sip:6336@172.101.1.177:5060;transport=UDP>
Proxy-Authorization: Digest username="6336",realm="asterisk",nonce="26a0c509",uri="sip:6338@172.101.1.253;transport=UDP",response="543f5a1ddb19fe7a9b903ef353ef538b",algorithm=MD5
Content-Type: application/sdp
Content-Length: 155
v=0
o=rtp/7 11855626630 11855626630 IN IP4 172.101.1.177
s=-
c=IN IP4 172.101.1.177
t=0 0
m=audio 2070 RTP/AVP 8 0 18
a=ptime:20
a=SilenceSupp:off
<------------->
--- (14 headers 8 lines) ---
Sending to 172.101.1.177 : 5060 (NAT)
Using INVITE request as basis request - 414c18-b10165ac-13c4-30fe-5963bb98-30fe@172.101.1.253
Found peer '6336'
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 18
Peer audio RTP is at port 172.101.1.177:2070
Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x10c (ulaw|alaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 172.101.1.177:2070
Looking for 6338 in default (domain 172.101.1.253)
list_route: hop: <sip:6336@172.101.1.177:5060;transport=UDP>
voip*CLI>
<--- Transmitting (no NAT) to 172.101.1.177:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.101.1.177:5060;branch=z9hG4bK-30fe-bf6374-2dc8f67a;received=172.101.1.177;rport=5060
From: "6336"<sip:6336@172.101.1.253>;tag=3ec790-b10165ac-13c4-30fe-5e7f131c-30fe
To: <sip:6338@172.101.1.253>
Call-ID: 414c18-b10165ac-13c4-30fe-5963bb98-30fe@172.101.1.253
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:6338@172.101.1.253>
Content-Length: 0
<------------>
Audio is at 172.101.1.253 port 13664
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x100 (g729) to SDP
<--- Reliably Transmitting (no NAT) to 172.101.1.177:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.101.1.177:5060;branch=z9hG4bK-30fe-bf6374-2dc8f67a;received=172.101.1.177;rport=5060
From: "6336"<sip:6336@172.101.1.253>;tag=3ec790-b10165ac-13c4-30fe-5e7f131c-30fe
To: <sip:6338@172.101.1.253>;tag=as4c70d5b0
Call-ID: 414c18-b10165ac-13c4-30fe-5963bb98-30fe@172.101.1.253
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:6338@172.101.1.253>
Content-Type: application/sdp
Content-Length: 257
v=0
o=root 12830 12830 IN IP4 172.101.1.253
s=session
c=IN IP4 172.101.1.253
t=0 0
m=audio 13664 RTP/AVP 8 0 18
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------>
voip*CLI>
<--- SIP read from 172.101.1.177:5060 --->
ACK sip:6338@172.101.1.253 SIP/2.0
From: "6336"<sip:6336@172.101.1.253>;tag=3ec790-b10165ac-13c4-30fe-5e7f131c-30fe
To: <sip:6338@172.101.1.253>;tag=as4c70d5b0
Call-ID: 414c18-b10165ac-13c4-30fe-5963bb98-30fe@172.101.1.253
CSeq: 2 ACK
Via: SIP/2.0/UDP 172.101.1.177:5060;rport;branch=z9hG4bK-30fe-bf63a5-76e95d44
Max-Forwards: 70
User-Agent: WIP330
Contact: <sip:6336@172.101.1.177:5060;transport=UDP>
Proxy-Authorization: Digest username="6336",realm="asterisk",nonce="26a0c509",uri="sip:6338@172.101.1.253;transport=UDP",response="543f5a1ddb19fe7a9b903ef353ef538b",algorithm=MD5
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
Reliably Transmitting (no NAT) to 172.101.1.177:5060:
OPTIONS sip:6336@172.101.1.177:5060 SIP/2.0
Via: SIP/2.0/UDP 172.101.1.253:5060;branch=z9hG4bK2455ef69;rport
From: "asterisk" <sip:asterisk@172.101.1.253>;tag=as0770f643
To: <sip:6336@172.101.1.177:5060>
Contact: <sip:asterisk@172.101.1.253>
Call-ID: 23fd55f550f07b881e0052e16a3af9a8@172.101.1.253
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 27 Jul 2007 18:57:45 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
---
voip*CLI>
<--- SIP read from 172.101.1.177:5060 --->
SIP/2.0 501 Not Implemented
From: "asterisk"<sip:asterisk@172.101.1.253>;tag=as0770f643
To: <sip:6336@172.101.1.177>;tag=3ecc58-b10165ac-13c4-3101-51b0b1b8-3101
Call-ID: 23fd55f550f07b881e0052e16a3af9a8@172.101.1.253
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 172.101.1.253:5060;rport=5060;branch=z9hG4bK2455ef69
Supported: replaces
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '23fd55f550f07b881e0052e16a3af9a8@172.101.1.253' Method: OPTIONS
[Jul 27 13:57:45] NOTICE[12933]: chan_sip.c:15352 sip_poke_noanswer: Peer '6337' is now UNREACHABLE! Last qualify: 2266
Scheduling destruction of SIP dialog '414c18-b10165ac-13c4-30fe-5963bb98-30fe@172.101.1.253' in 31040 ms (Method: ACK)
set_destination: Parsing <sip:6336@172.101.1.177:5060;transport=UDP> for address/port to send to
set_destination: set destination to 172.101.1.177, port 5060
Reliably Transmitting (no NAT) to 172.101.1.177:5060:
BYE sip:6336@172.101.1.177:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 172.101.1.253:5060;branch=z9hG4bK73f4c462;rport
From: <sip:6338@172.101.1.253>;tag=as4c70d5b0
To: "6336"<sip:6336@172.101.1.253>;tag=3ec790-b10165ac-13c4-30fe-5e7f131c-30fe
Call-ID: 414c18-b10165ac-13c4-30fe-5963bb98-30fe@172.101.1.253
CSeq: 102 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
Retransmitting #1 (no NAT) to 172.101.1.177:5060:
BYE sip:6336@172.101.1.177:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 172.101.1.253:5060;branch=z9hG4bK73f4c462;rport
From: <sip:6338@172.101.1.253>;tag=as4c70d5b0
To: "6336"<sip:6336@172.101.1.253>;tag=3ec790-b10165ac-13c4-30fe-5e7f131c-30fe
Call-ID: 414c18-b10165ac-13c4-30fe-5963bb98-30fe@172.101.1.253
CSeq: 102 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
voip*CLI>
<--- SIP read from 172.101.1.177:5060 --->
SIP/2.0 200 OK
From: <sip:6338@172.101.1.253>;tag=as4c70d5b0
To: "6336"<sip:6336@172.101.1.253>;tag=3ec790-b10165ac-13c4-30fe-5e7f131c-30fe
Call-ID: 414c18-b10165ac-13c4-30fe-5963bb98-30fe@172.101.1.253
CSeq: 102 BYE
Via: SIP/2.0/UDP 172.101.1.253:5060;rport=5060;branch=z9hG4bK73f4c462
Supported: replaces
User-Agent: WIP330
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '414c18-b10165ac-13c4-30fe-5963bb98-30fe@172.101.1.253' Method: ACK
voip*CLI>
<--- SIP read from 172.101.1.177:5060 --->
SIP/2.0 200 OK
From: <sip:6338@172.101.1.253>;tag=as4c70d5b0
To: "6336"<sip:6336@172.101.1.253>;tag=3ec790-b10165ac-13c4-30fe-5e7f131c-30fe
Call-ID: 414c18-b10165ac-13c4-30fe-5963bb98-30fe@172.101.1.253
CSeq: 102 BYE
Via: SIP/2.0/UDP 172.101.1.253:5060;rport=5060;branch=z9hG4bK73f4c462
Supported: replaces
User-Agent: WIP330
Content-Length: 0