[Apr 19 08:39:13] VERBOSE[3400] chan_sip.c:
<--- SIP read from UDP:172.16.0.20:5060 --->
INVITE sip:100@172.16.0.10:5060;user=phone SIP/2.0
Allow:INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,REFER,SUBSCRIBE,NOTIFY,UPDATE
Via: SIP/2.0/UDP 172.16.0.20:5060;branch=z9hG4bKf73a3d86b3a377ab
From: "1000" <sip:1000@172.16.0.10>;tag=7ec3f92d-16226
To: <sip:100@172.16.0.10:5060;user=phone>
Call-ID: BD21-1143-4100062266E8A23356586-022@SipHost
CSeq:1141 INVITE
Contact: <sip:1000@172.16.0.20:5060>
Expires:90
Max-Forwards:70
Min-SE:90
Session-Expires: 1800
Supported: replaces,timer
User-Agent:dlink
Content-Type: application/sdp
Content-Length: 333
v=0
o=1000 2123421310 2123421310 IN IP4 172.16.0.20
s=Session SDP
c=IN IP4 172.16.0.20
t=0 0
m=audio 9000 RTP/AVP 8 4 18 2 0 101
a=rtpmap:8 PCMA/8000/1
a=rtpmap:4 G723/8000/1
a=rtpmap:18 G729/8000/1
a=fmtp:18 annexb=no
a=rtpmap:2 G726-32/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-16
<------------->
[Apr 19 08:39:13] VERBOSE[3400] chan_sip.c: --- (16 headers 14 lines) ---
[Apr 19 08:39:13] VERBOSE[3400] chan_sip.c: Sending to 172.16.0.20:5060 (no NAT)
[Apr 19 08:39:13] VERBOSE[3400][C-00000001] chan_sip.c: Sending to 172.16.0.20:5060 (no NAT)
[Apr 19 08:39:13] VERBOSE[3400][C-00000001] chan_sip.c: Using INVITE request as basis request - BD21-1143-4100062266E8A23356586-022@SipHost
[Apr 19 08:39:13] VERBOSE[3400][C-00000001] chan_sip.c: Found peer '1000' for '1000' from 172.16.0.20:5060
[Apr 19 08:39:13] VERBOSE[3400][C-00000001] chan_sip.c:
<--- Reliably Transmitting (no NAT) to 172.16.0.20:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 172.16.0.20:5060;branch=z9hG4bKf73a3d86b3a377ab;received=172.16.0.20
From: "1000" <sip:1000@172.16.0.10>;tag=7ec3f92d-16226
To: <sip:100@172.16.0.10:5060;user=phone>;tag=as174558a4
Call-ID: BD21-1143-4100062266E8A23356586-022@SipHost
CSeq: 1141 INVITE
Server: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1c734ae6"
Content-Length: 0
<------------>
[Apr 19 08:39:13] VERBOSE[3400][C-00000001] chan_sip.c: Scheduling destruction of SIP dialog 'BD21-1143-4100062266E8A23356586-022@SipHost' in 6400 ms (Method: INVITE)
[Apr 19 08:39:13] VERBOSE[3400] chan_sip.c:
<--- SIP read from UDP:172.16.0.20:5060 --->
ACK sip:100@172.16.0.10:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 172.16.0.20:5060;branch=z9hG4bKf73a3d86b3a377ab
From: "1000" <sip:1000@172.16.0.10>;tag=7ec3f92d-16226
To: <sip:100@172.16.0.10:5060;user=phone>;tag=as174558a4
Call-ID: BD21-1143-4100062266E8A23356586-022@SipHost
CSeq:1141 ACK
Max-Forwards:70
Content-Length: 0
<------------->
[Apr 19 08:39:13] VERBOSE[3400] chan_sip.c: --- (8 headers 0 lines) ---
[Apr 19 08:39:13] VERBOSE[3400] chan_sip.c:
<--- SIP read from UDP:172.16.0.20:5060 --->
INVITE sip:100@172.16.0.10:5060;user=phone SIP/2.0
Allow:INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,REFER,SUBSCRIBE,NOTIFY,UPDATE
Via: SIP/2.0/UDP 172.16.0.20:5060;branch=z9hG4bK2330025bbd617c5d
From: "1000" <sip:1000@172.16.0.10>;tag=7ec3f92d-16226
To: <sip:100@172.16.0.10:5060;user=phone>
Call-ID: BD21-1143-4100062266E8A23356586-022@SipHost
CSeq:1142 INVITE
Contact: <sip:1000@172.16.0.20:5060>
Expires:90
Max-Forwards:70
Min-SE:90
Authorization:Digest username="1000",realm="asterisk",nonce="1c734ae6",uri="sip:100@172.16.0.10:5060;user=phone",response="3b02d882988d13ba6558854920fffcf6",algorithm=MD5
Session-Expires: 1800
Supported: replaces,timer
User-Agent:dlink
Content-Type: application/sdp
Content-Length: 333
v=0
o=1000 2123421310 2123421310 IN IP4 172.16.0.20
s=Session SDP
c=IN IP4 172.16.0.20
t=0 0
m=audio 9000 RTP/AVP 8 4 18 2 0 101
a=rtpmap:8 PCMA/8000/1
a=rtpmap:4 G723/8000/1
a=rtpmap:18 G729/8000/1
a=fmtp:18 annexb=no
a=rtpmap:2 G726-32/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-16
<------------->
[Apr 19 08:39:13] VERBOSE[3400] chan_sip.c: --- (17 headers 14 lines) ---
[Apr 19 08:39:13] VERBOSE[3400][C-00000001] chan_sip.c: Sending to 172.16.0.20:5060 (no NAT)
[Apr 19 08:39:13] VERBOSE[3400][C-00000001] chan_sip.c: Using INVITE request as basis request - BD21-1143-4100062266E8A23356586-022@SipHost
[Apr 19 08:39:13] VERBOSE[3400][C-00000001] chan_sip.c: Found peer '1000' for '1000' from 172.16.0.20:5060
[Apr 19 08:39:13] VERBOSE[3400][C-00000001] netsock2.c: == Using SIP RTP CoS mark 5
[Apr 19 08:39:13] VERBOSE[3400][C-00000001] chan_sip.c: Found RTP audio format 8
[Apr 19 08:39:13] VERBOSE[3400][C-00000001] chan_sip.c: Found RTP audio format 4
[Apr 19 08:39:13] VERBOSE[3400][C-00000001] chan_sip.c: Found RTP audio format 18
[Apr 19 08:39:13] VERBOSE[3400][C-00000001] chan_sip.c: Found RTP audio format 2
[Apr 19 08:39:13] VERBOSE[3400][C-00000001] chan_sip.c: Found RTP audio format 0
[Apr 19 08:39:13] VERBOSE[3400][C-00000001] chan_sip.c: Found RTP audio format 101
[Apr 19 08:39:13] VERBOSE[3400][C-00000001] chan_sip.c: Found audio description format PCMA for ID 8
[Apr 19 08:39:13] VERBOSE[3400][C-00000001] chan_sip.c: Found audio description format G723 for ID 4
[Apr 19 08:39:13] VERBOSE[3400][C-00000001] chan_sip.c: Found audio description format G729 for ID 18
[Apr 19 08:39:13] VERBOSE[3400][C-00000001] chan_sip.c: Found audio description format G726-32 for ID 2
[Apr 19 08:39:13] VERBOSE[3400][C-00000001] chan_sip.c: Found audio description format PCMU for ID 0
[Apr 19 08:39:13] VERBOSE[3400][C-00000001] chan_sip.c: Found audio description format telephone-event for ID 101
[Apr 19 08:39:13] VERBOSE[3400][C-00000001] chan_sip.c: Capabilities: us - (gsm|ulaw|alaw), peer - audio=(g723|ulaw|alaw|g726|g729)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
[Apr 19 08:39:13] VERBOSE[3400][C-00000001] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[Apr 19 08:39:13] VERBOSE[3400][C-00000001] chan_sip.c: Peer audio RTP is at port 172.16.0.20:9000
[Apr 19 08:39:13] VERBOSE[3400][C-00000001] chan_sip.c: Looking for 100 in context (domain 172.16.0.10)
[Apr 19 08:39:13] VERBOSE[3400][C-00000001] chan_sip.c: list_route: hop: <sip:1000@172.16.0.20:5060>
[Apr 19 08:39:13] VERBOSE[3400][C-00000001] chan_sip.c:
<--- Transmitting (no NAT) to 172.16.0.20:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.16.0.20:5060;branch=z9hG4bK2330025bbd617c5d;received=172.16.0.20
From: "1000" <sip:1000@172.16.0.10>;tag=7ec3f92d-16226
To: <sip:100@172.16.0.10:5060;user=phone>
Call-ID: BD21-1143-4100062266E8A23356586-022@SipHost
CSeq: 1142 INVITE
Server: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:100@172.16.0.10:5060>
Content-Length: 0
<------------>
[Apr 19 08:39:13] VERBOSE[14401][C-00000001] pbx.c: -- Executing [100@context:1] Dial("SIP/1000-00000002", "SIP/2000/100,120,mXWKT") in new stack
[Apr 19 08:39:13] VERBOSE[14401][C-00000001] netsock2.c: == Using SIP RTP CoS mark 5
[Apr 19 08:39:13] VERBOSE[14401][C-00000001] chan_sip.c: Audio is at 13716
[Apr 19 08:39:13] VERBOSE[14401][C-00000001] chan_sip.c: Adding codec 100004 (alaw) to SDP
[Apr 19 08:39:13] VERBOSE[14401][C-00000001] chan_sip.c: Adding codec 100003 (ulaw) to SDP
[Apr 19 08:39:13] VERBOSE[14401][C-00000001] chan_sip.c: Adding codec 100002 (gsm) to SDP
[Apr 19 08:39:13] VERBOSE[14401][C-00000001] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Apr 19 08:39:13] VERBOSE[14401][C-00000001] chan_sip.c: Reliably Transmitting (no NAT) to 172.16.0.20:5060:
INVITE sip:100@172.16.0.20:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.0.10:5060;branch=z9hG4bK16e17a80
Max-Forwards: 70
From: "PSTN Phone" <sip:1000@172.16.0.10:5060>;tag=as145b939c
To: <sip:100@172.16.0.20:5060>
Contact: <sip:1000@172.16.0.10:5060>
Call-ID:
20853266711a687119364bfb485b4a96@172.16.0.10:5060CSeq: 102 INVITE
User-Agent: Asterisk
Date: Sat, 19 Apr 2014 04:39:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 529
v=0
o=root 387132854 387132854 IN IP4 172.16.0.10
s=Asterisk PBX 11.8.1
c=IN IP4 172.16.0.10
t=0 0
m=audio 13716 RTP/AVP 8 0 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=ice-ufrag:2a32b5623206171467a6506743ae0475
a=ice-pwd:1a5be7d976a9c3a84e7dd3943a1b5150
a=candidate:Hac100062 1 UDP 2130706431 172.16.0.10 13716 typ host
a=candidate:Hac100062 2 UDP 2130706430 172.16.0.10 13717 typ host
a=sendrecv
---
[Apr 19 08:39:13] VERBOSE[14401][C-00000001] app_dial.c: -- Called SIP/2000/100
[Apr 19 08:39:13] VERBOSE[14401][C-00000001] res_musiconhold.c: -- Started music on hold, class 'default', on SIP/1000-00000002
[Apr 19 08:39:13] VERBOSE[14401][C-00000001] chan_sip.c: Audio is at 10282
[Apr 19 08:39:13] VERBOSE[14401][C-00000001] chan_sip.c: Adding codec 100004 (alaw) to SDP
[Apr 19 08:39:13] VERBOSE[14401][C-00000001] chan_sip.c: Adding codec 100003 (ulaw) to SDP
[Apr 19 08:39:13] VERBOSE[14401][C-00000001] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Apr 19 08:39:13] VERBOSE[14401][C-00000001] chan_sip.c:
<--- Transmitting (no NAT) to 172.16.0.20:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.16.0.20:5060;branch=z9hG4bK2330025bbd617c5d;received=172.16.0.20
From: "1000" <sip:1000@172.16.0.10>;tag=7ec3f92d-16226
To: <sip:100@172.16.0.10:5060;user=phone>;tag=as3b3d3d2c
Call-ID: BD21-1143-4100062266E8A23356586-022@SipHost
CSeq: 1142 INVITE
Server: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:100@172.16.0.10:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 506
v=0
o=root 887912993 887912993 IN IP4 172.16.0.10
s=Asterisk PBX 11.8.1
c=IN IP4 172.16.0.10
t=0 0
m=audio 10282 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=ice-ufrag:53afa21f5bc7d61b2ceaee3132cfb05f
a=ice-pwd:25f4c76f163a074a6fa7421231f75cd8
a=candidate:Hac100062 1 UDP 2130706431 172.16.0.10 10282 typ host
a=candidate:Hac100062 2 UDP 2130706430 172.16.0.10 10283 typ host
a=sendrecv
<------------>
[Apr 19 08:39:13] WARNING[14401][C-00000001] mp3/interface.c: Junk at the beginning of frame 49443303
[Apr 19 08:39:13] VERBOSE[3400] chan_sip.c:
<--- SIP read from UDP:172.16.0.20:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.16.0.10:5060;branch=z9hG4bK16e17a80
From: "PSTN Phone" <sip:1000@172.16.0.10:5060>;tag=as145b939c
To: <sip:100@172.16.0.20:5060>
Call-ID:
20853266711a687119364bfb485b4a96@172.16.0.10:5060CSeq:102 INVITE
Content-Type: application/sdp
Content-Length: 0
<------------->
[Apr 19 08:39:13] VERBOSE[3400] chan_sip.c: --- (8 headers 0 lines) ---
[Apr 19 08:39:13] VERBOSE[14401][C-00000001] res_rtp_asterisk.c: > 0x964c468 -- Probation passed - setting RTP source address to 172.16.0.20:9000
[Apr 19 08:39:15] VERBOSE[3400] chan_sip.c:
<--- SIP read from UDP:172.16.0.20:5060 --->
SIP/2.0 183 Session in progress
Via: SIP/2.0/UDP 172.16.0.10:5060;branch=z9hG4bK16e17a80
From: "PSTN Phone" <sip:1000@172.16.0.10:5060>;tag=as145b939c
To: <sip:100@172.16.0.20:5060>;tag=33e5ff91-16228
Call-ID:
20853266711a687119364bfb485b4a96@172.16.0.10:5060CSeq:102 INVITE
Contact: <sip:172.16.0.20:5060>
User-Agent:dlink
Content-Type: application/sdp
Content-Length: 217
v=0
o=2000 2123423630 2123423630 IN IP4 172.16.0.20
s=Session SDP
c=IN IP4 172.16.0.20
t=0 0
m=audio 9002 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-16
a=sendonly
<------------->
[Apr 19 08:39:15] VERBOSE[3400] chan_sip.c: --- (10 headers 10 lines) ---
[Apr 19 08:39:15] VERBOSE[3400][C-00000001] chan_sip.c: list_route: hop: <sip:172.16.0.20:5060>
[Apr 19 08:39:15] VERBOSE[3400][C-00000001] chan_sip.c: Found RTP audio format 8
[Apr 19 08:39:15] VERBOSE[3400][C-00000001] chan_sip.c: Found RTP audio format 101
[Apr 19 08:39:15] VERBOSE[3400][C-00000001] chan_sip.c: Found audio description format PCMA for ID 8
[Apr 19 08:39:15] VERBOSE[3400][C-00000001] chan_sip.c: Found audio description format telephone-event for ID 101
[Apr 19 08:39:15] VERBOSE[3400][C-00000001] chan_sip.c: Capabilities: us - (gsm|ulaw|alaw), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
[Apr 19 08:39:15] VERBOSE[3400][C-00000001] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[Apr 19 08:39:15] VERBOSE[3400][C-00000001] chan_sip.c: Peer audio RTP is at port 172.16.0.20:9002
[Apr 19 08:39:15] VERBOSE[14401][C-00000001] app_dial.c: -- Call on SIP/2000-00000003 placed on hold
[Apr 19 08:39:15] VERBOSE[14401][C-00000001] res_musiconhold.c: -- Stopped music on hold on SIP/1000-00000002
[Apr 19 08:39:15] VERBOSE[14401][C-00000001] res_musiconhold.c: -- Started music on hold, class 'default', on SIP/1000-00000002
[Apr 19 08:39:15] VERBOSE[14401][C-00000001] app_dial.c: -- SIP/2000-00000003 is making progress passing it to SIP/1000-00000002
[Apr 19 08:39:15] WARNING[14401][C-00000001] mp3/interface.c: Junk at the beginning of frame 49443303
[Apr 19 08:39:15] VERBOSE[14401][C-00000001] res_rtp_asterisk.c: > 0x94fe578 -- Probation passed - setting RTP source address to 172.16.0.20:9002
[Apr 19 08:39:23] VERBOSE[3400] chan_sip.c:
<--- SIP read from UDP:172.16.0.20:5060 --->
SIP/2.0 200 OK
Allow:INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,REFER,SUBSCRIBE,NOTIFY,UPDATE
Via: SIP/2.0/UDP 172.16.0.10:5060;branch=z9hG4bK16e17a80
From: "PSTN Phone" <sip:1000@172.16.0.10:5060>;tag=as145b939c
To: <sip:100@172.16.0.20:5060>;tag=33e5ff91-16228
Call-ID:
20853266711a687119364bfb485b4a96@172.16.0.10:5060CSeq:102 INVITE
Contact: <sip:172.16.0.20:5060>
User-Agent:dlink
Content-Type: application/sdp
Content-Length: 205
v=0
o=2000 2123431110 2123431110 IN IP4 172.16.0.20
s=Session SDP
c=IN IP4 172.16.0.20
t=0 0
m=audio 9002 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-16
<------------->
[Apr 19 08:39:23] VERBOSE[3400] chan_sip.c: --- (11 headers 9 lines) ---
[Apr 19 08:39:23] VERBOSE[3400][C-00000001] chan_sip.c: Found RTP audio format 8
[Apr 19 08:39:23] VERBOSE[3400][C-00000001] chan_sip.c: Found RTP audio format 101
[Apr 19 08:39:23] VERBOSE[3400][C-00000001] chan_sip.c: Found audio description format PCMA for ID 8
[Apr 19 08:39:23] VERBOSE[3400][C-00000001] chan_sip.c: Found audio description format telephone-event for ID 101
[Apr 19 08:39:23] VERBOSE[3400][C-00000001] chan_sip.c: Capabilities: us - (gsm|ulaw|alaw), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
[Apr 19 08:39:23] VERBOSE[3400][C-00000001] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[Apr 19 08:39:23] VERBOSE[3400][C-00000001] chan_sip.c: Peer audio RTP is at port 172.16.0.20:9002
[Apr 19 08:39:23] VERBOSE[3400][C-00000001] chan_sip.c: list_route: hop: <sip:172.16.0.20:5060>
[Apr 19 08:39:23] VERBOSE[3400][C-00000001] chan_sip.c: set_destination: Parsing <sip:172.16.0.20:5060> for address/port to send to
[Apr 19 08:39:23] VERBOSE[3400][C-00000001] chan_sip.c: set_destination: set destination to 172.16.0.20:5060
[Apr 19 08:39:23] VERBOSE[3400][C-00000001] chan_sip.c: Transmitting (no NAT) to 172.16.0.20:5060:
ACK sip:172.16.0.20:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.0.10:5060;branch=z9hG4bK6a8cdf26
Max-Forwards: 70
From: "PSTN Phone" <sip:1000@172.16.0.10:5060>;tag=as145b939c
To: <sip:100@172.16.0.20:5060>;tag=33e5ff91-16228
Contact: <sip:1000@172.16.0.10:5060>
Call-ID:
20853266711a687119364bfb485b4a96@172.16.0.10:5060CSeq: 102 ACK
User-Agent: Asterisk
Content-Length: 0
---
[Apr 19 08:39:23] VERBOSE[14401][C-00000001] app_dial.c: -- Call on SIP/2000-00000003 left from hold
[Apr 19 08:39:23] VERBOSE[14401][C-00000001] res_musiconhold.c: -- Stopped music on hold on SIP/1000-00000002
[Apr 19 08:39:23] VERBOSE[14401][C-00000001] app_dial.c: -- SIP/2000-00000003 answered SIP/1000-00000002
[Apr 19 08:39:23] VERBOSE[14401][C-00000001] chan_sip.c: Audio is at 10282
[Apr 19 08:39:23] VERBOSE[14401][C-00000001] chan_sip.c: Adding codec 100004 (alaw) to SDP
[Apr 19 08:39:23] VERBOSE[14401][C-00000001] chan_sip.c: Adding codec 100003 (ulaw) to SDP
[Apr 19 08:39:23] VERBOSE[14401][C-00000001] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Apr 19 08:39:23] VERBOSE[14401][C-00000001] chan_sip.c:
<--- Reliably Transmitting (no NAT) to 172.16.0.20:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.0.20:5060;branch=z9hG4bK2330025bbd617c5d;received=172.16.0.20
From: "1000" <sip:1000@172.16.0.10>;tag=7ec3f92d-16226
To: <sip:100@172.16.0.10:5060;user=phone>;tag=as3b3d3d2c
Call-ID: BD21-1143-4100062266E8A23356586-022@SipHost
CSeq: 1142 INVITE
Server: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:100@172.16.0.10:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 506
v=0
o=root 887912993 887912993 IN IP4 172.16.0.10
s=Asterisk PBX 11.8.1
c=IN IP4 172.16.0.10
t=0 0
m=audio 10282 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=ice-ufrag:53afa21f5bc7d61b2ceaee3132cfb05f
a=ice-pwd:25f4c76f163a074a6fa7421231f75cd8
a=candidate:Hac100062 1 UDP 2130706431 172.16.0.10 10282 typ host
a=candidate:Hac100062 2 UDP 2130706430 172.16.0.10 10283 typ host
a=sendrecv
<------------>
[Apr 19 08:39:23] VERBOSE[3400] chan_sip.c: Retransmitting #1 (no NAT) to 172.16.0.20:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.0.20:5060;branch=z9hG4bK2330025bbd617c5d;received=172.16.0.20
From: "1000" <sip:1000@172.16.0.10>;tag=7ec3f92d-16226
To: <sip:100@172.16.0.10:5060;user=phone>;tag=as3b3d3d2c
Call-ID: BD21-1143-4100062266E8A23356586-022@SipHost
CSeq: 1142 INVITE
Server: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:100@172.16.0.10:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 506
v=0
o=root 887912993 887912993 IN IP4 172.16.0.10
s=Asterisk PBX 11.8.1
c=IN IP4 172.16.0.10
t=0 0
m=audio 10282 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=ice-ufrag:53afa21f5bc7d61b2ceaee3132cfb05f
a=ice-pwd:25f4c76f163a074a6fa7421231f75cd8
a=candidate:Hac100062 1 UDP 2130706431 172.16.0.10 10282 typ host
a=candidate:Hac100062 2 UDP 2130706430 172.16.0.10 10283 typ host
a=sendrecv
---
[Apr 19 08:39:23] VERBOSE[14401][C-00000001] res_rtp_asterisk.c: > 0x94fe578 -- Probation passed - setting RTP source address to 172.16.0.20:9002
[Apr 19 08:39:23] VERBOSE[3400] chan_sip.c:
<--- SIP read from UDP:172.16.0.20:5060 --->
ACK sip:100@172.16.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.0.20:5060;branch=z9hG4bKea4e119ec5abf0ed
From: "1000" <sip:1000@172.16.0.10>;tag=7ec3f92d-16226
To: <sip:100@172.16.0.10:5060;user=phone>;tag=as3b3d3d2c
Call-ID: BD21-1143-4100062266E8A23356586-022@SipHost
CSeq:1142 ACK
Max-Forwards:70
Authorization:Digest username="1000",realm="asterisk",nonce="1c734ae6",uri="sip:100@172.16.0.10:5060;user=phone",response="3b02d882988d13ba6558854920fffcf6",algorithm=MD5
User-Agent:dlink
Content-Length: 0
<------------->
[Apr 19 08:39:23] VERBOSE[3400] chan_sip.c: --- (10 headers 0 lines) ---
[Apr 19 08:39:23] VERBOSE[3400] chan_sip.c:
<--- SIP read from UDP:172.16.0.20:5060 --->
ACK sip:100@172.16.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.0.20:5060;branch=z9hG4bKea4e119ec5abf0ed
From: "1000" <sip:1000@172.16.0.10>;tag=7ec3f92d-16226
To: <sip:100@172.16.0.10:5060;user=phone>;tag=as3b3d3d2c
Call-ID: BD21-1143-4100062266E8A23356586-022@SipHost
CSeq:1142 ACK
Max-Forwards:70
Authorization:Digest username="1000",realm="asterisk",nonce="1c734ae6",uri="sip:100@172.16.0.10:5060;user=phone",response="3b02d882988d13ba6558854920fffcf6",algorithm=MD5
User-Agent:dlink
Content-Length: 0
<------------->
[Apr 19 08:39:23] VERBOSE[3400] chan_sip.c: --- (10 headers 0 lines) ---
[Apr 19 08:39:26] VERBOSE[3400] chan_sip.c:
<--- SIP read from UDP:172.16.0.20:5060 --->
BYE sip:100@172.16.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.0.20:5060;branch=z9hG4bK45eff9d052d63fb4
From: "1000" <sip:1000@172.16.0.10>;tag=7ec3f92d-16226
To: <sip:100@172.16.0.10:5060;user=phone>;tag=as3b3d3d2c
Call-ID: BD21-1143-4100062266E8A23356586-022@SipHost
CSeq:1143 BYE
Max-Forwards:70
User-Agent:dlink
Content-Length: 0
<------------->
[Apr 19 08:39:26] VERBOSE[3400] chan_sip.c: --- (9 headers 0 lines) ---
[Apr 19 08:39:26] VERBOSE[3400][C-00000001] chan_sip.c: Sending to 172.16.0.20:5060 (no NAT)
[Apr 19 08:39:26] VERBOSE[3400][C-00000001] chan_sip.c: Scheduling destruction of SIP dialog 'BD21-1143-4100062266E8A23356586-022@SipHost' in 6400 ms (Method: BYE)
[Apr 19 08:39:26] VERBOSE[3400][C-00000001] chan_sip.c:
<--- Transmitting (no NAT) to 172.16.0.20:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.0.20:5060;branch=z9hG4bK45eff9d052d63fb4;received=172.16.0.20
From: "1000" <sip:1000@172.16.0.10>;tag=7ec3f92d-16226
To: <sip:100@172.16.0.10:5060;user=phone>;tag=as3b3d3d2c
Call-ID: BD21-1143-4100062266E8A23356586-022@SipHost
CSeq: 1143 BYE
Server: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
<------------>
[Apr 19 08:39:26] VERBOSE[14401][C-00000001] pbx.c: -- Executing [h@context:9996] NoOp("SIP/1000-00000002", "") in new stack
[Apr 19 08:39:26] WARNING[14401][C-00000001] res_odbc.c: SetConnectAttr (Txn isolation) returned an error: HY000: [MySQL][ODBC 5.1 Driver]MySQL server has gone away
[Apr 19 08:39:33] WARNING[3400] chan_sip.c: Autodestruct on dialog 'BD21-1143-4100062266E8A23356586-022@SipHost' with owner SIP/1000-00000002 in place (Method: BYE). Rescheduling destruction for 10000 ms
[Apr 19 08:39:33] VERBOSE[3400] chan_sip.c:
<--- SIP read from UDP:172.16.0.20:5060 --->
BYE sip:1000@172.16.0.10:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.0.20:5060;branch=z9hG4bK9438a22e89e0c201
From: <sip:100@172.16.0.20:5060>;tag=33e5ff91-16228
To: "PSTN Phone" <sip:1000@172.16.0.10:5060>;tag=as145b939c
Call-ID:
20853266711a687119364bfb485b4a96@172.16.0.10:5060CSeq:2 BYE
Contact: <sip:172.16.0.20:5060>
Max-Forwards:70
User-Agent:dlink
Content-Length: 0
<------------->
[Apr 19 08:39:33] VERBOSE[3400] chan_sip.c: --- (10 headers 0 lines) ---
[Apr 19 08:39:33] VERBOSE[3400][C-00000001] chan_sip.c: Sending to 172.16.0.20:5060 (no NAT)
[Apr 19 08:39:33] VERBOSE[3400][C-00000001] chan_sip.c: Scheduling destruction of SIP dialog '20853266711a687119364bfb485b4a96@172.16.0.10:5060' in 6400 ms (Method: BYE)
[Apr 19 08:39:33] VERBOSE[3400][C-00000001] chan_sip.c:
<--- Transmitting (no NAT) to 172.16.0.20:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.0.20:5060;branch=z9hG4bK9438a22e89e0c201;received=172.16.0.20
From: <sip:100@172.16.0.20:5060>;tag=33e5ff91-16228
To: "PSTN Phone" <sip:1000@172.16.0.10:5060>;tag=as145b939c
Call-ID:
20853266711a687119364bfb485b4a96@172.16.0.10:5060CSeq: 2 BYE
Server: Asterisk
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
<------------>
[Apr 19 08:39:39] WARNING[3400] chan_sip.c: Autodestruct on dialog 'BD21-1143-4100062266E8A23356586-022@SipHost' with owner SIP/1000-00000002 in place (Method: BYE). Rescheduling destruction for 10000 ms
[Apr 19 08:39:39] WARNING[3400] chan_sip.c: Autodestruct on dialog '20853266711a687119364bfb485b4a96@172.16.0.10:5060' with owner SIP/2000-00000003 in place (Method: BYE). Rescheduling destruction for 10000 ms
[Apr 19 08:39:43] VERBOSE[3400] chan_sip.c:
<--- SIP read from UDP:172.16.0.21:5060 --->
<------------->
[Apr 19 08:39:45] VERBOSE[3400] chan_sip.c: Reliably Transmitting (no NAT) to 172.16.0.20:5060:
OPTIONS sip:1000@172.16.0.20:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.0.10:5060;branch=z9hG4bK1d4d5f19
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.16.0.10:5060>;tag=as05bf74dd
To: <sip:1000@172.16.0.20:5060>
Contact: <sip:asterisk@172.16.0.10:5060>
Call-ID:
1dc734840c3aa32a689796ea59af2ce0@172.16.0.10:5060CSeq: 102 OPTIONS
User-Agent: Asterisk
Date: Sat, 19 Apr 2014 04:39:45 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
---
[Apr 19 08:39:45] VERBOSE[3400] chan_sip.c:
<--- SIP read from UDP:172.16.0.20:5060 --->
SIP/2.0 200 OK
Allow:INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,REFER,SUBSCRIBE,NOTIFY,UPDATE
Via: SIP/2.0/UDP 172.16.0.10:5060;branch=z9hG4bK1d4d5f19
From: "asterisk" <sip:asterisk@172.16.0.10:5060>;tag=as05bf74dd
To: <sip:1000@172.16.0.20:5060>;tag=c8bcb663-16258
Call-ID:
1dc734840c3aa32a689796ea59af2ce0@172.16.0.10:5060CSeq:102 OPTIONS
Contact: <sip:1000@172.16.0.20:5060>
User-Agent:dlink
Content-Length: 0
<------------->
[Apr 19 08:39:45] VERBOSE[3400] chan_sip.c: --- (10 headers 0 lines) ---
[Apr 19 08:39:45] VERBOSE[3400] chan_sip.c: Really destroying SIP dialog '1dc734840c3aa32a689796ea59af2ce0@172.16.0.10:5060' Method: OPTIONS
[Apr 19 08:39:45] WARNING[3400] chan_sip.c: Autodestruct on dialog 'BD21-1143-4100062266E8A23356586-022@SipHost' with owner SIP/1000-00000002 in place (Method: BYE). Rescheduling destruction for 10000 ms
[Apr 19 08:39:46] VERBOSE[3400] chan_sip.c: Reliably Transmitting (no NAT) to 172.16.0.20:5060:
OPTIONS sip:2000@172.16.0.20:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.0.10:5060;branch=z9hG4bK2ccbad6b
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.16.0.10:5060>;tag=as0d4a0493
To: <sip:2000@172.16.0.20:5060>
Contact: <sip:asterisk@172.16.0.10:5060>
Call-ID:
49f2eada73ed0bf4420190e276e6b815@172.16.0.10:5060CSeq: 102 OPTIONS
User-Agent: Asterisk
Date: Sat, 19 Apr 2014 04:39:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
---
[Apr 19 08:39:46] WARNING[3400] chan_sip.c: Autodestruct on dialog '20853266711a687119364bfb485b4a96@172.16.0.10:5060' with owner SIP/2000-00000003 in place (Method: BYE). Rescheduling destruction for 10000 ms
[Apr 19 08:39:46] VERBOSE[3400] chan_sip.c:
<--- SIP read from UDP:172.16.0.20:5060 --->
SIP/2.0 200 OK
Allow:INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,REFER,SUBSCRIBE,NOTIFY,UPDATE
Via: SIP/2.0/UDP 172.16.0.10:5060;branch=z9hG4bK2ccbad6b
From: "asterisk" <sip:asterisk@172.16.0.10:5060>;tag=as0d4a0493
To: <sip:2000@172.16.0.20:5060>;tag=ebeebdae-16259
Call-ID:
49f2eada73ed0bf4420190e276e6b815@172.16.0.10:5060CSeq:102 OPTIONS
Contact: <sip:2000@172.16.0.20:5060>
User-Agent:dlink
Content-Length: 0
<------------->
[Apr 19 08:39:46] VERBOSE[3400] chan_sip.c: --- (10 headers 0 lines) ---
[Apr 19 08:39:46] VERBOSE[3400] chan_sip.c: Really destroying SIP dialog '49f2eada73ed0bf4420190e276e6b815@172.16.0.10:5060' Method: OPTIONS
[Apr 19 08:39:52] WARNING[3400] chan_sip.c: Autodestruct on dialog 'BD21-1143-4100062266E8A23356586-022@SipHost' with owner SIP/1000-00000002 in place (Method: BYE). Rescheduling destruction for 10000 ms
[Apr 19 08:39:52] WARNING[3400] chan_sip.c: Autodestruct on dialog '20853266711a687119364bfb485b4a96@172.16.0.10:5060' with owner SIP/2000-00000003 in place (Method: BYE). Rescheduling destruction for 10000 ms
[Apr 19 08:39:58] WARNING[3400] chan_sip.c: Autodestruct on dialog 'BD21-1143-4100062266E8A23356586-022@SipHost' with owner SIP/1000-00000002 in place (Method: BYE). Rescheduling destruction for 10000 ms
[Apr 19 08:39:58] WARNING[3400] chan_sip.c: Autodestruct on dialog '20853266711a687119364bfb485b4a96@172.16.0.10:5060' with owner SIP/2000-00000003 in place (Method: BYE). Rescheduling destruction for 10000 ms
[Apr 19 08:40:05] WARNING[3400] chan_sip.c: Autodestruct on dialog 'BD21-1143-4100062266E8A23356586-022@SipHost' with owner SIP/1000-00000002 in place (Method: BYE). Rescheduling destruction for 10000 ms
[Apr 19 08:40:05] WARNING[3400] chan_sip.c: Autodestruct on dialog '20853266711a687119364bfb485b4a96@172.16.0.10:5060' with owner SIP/2000-00000003 in place (Method: BYE). Rescheduling destruction for 10000 ms