All pastes #554487 Raw Edit

trentster

public text v1 · immutable
·published 2007-06-10 04:56 UTC
rendered paste body
INVITE sip:0860121218@pabx.itws.co.za;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.1.1.3:5060;branch=z9hG4bK-nq8yy222qscg;rport
From: "6720" <sip:6720@pabx.itws.co.za>;tag=2ps68pkrvk
To: <sip:0860121218@pabx.itws.co.za;user=phone>
Call-ID: 3c2a60a8493e-7gnq2vbwskte@snom320-000413242316
CSeq: 1 INVITE
Max-Forwards: 70
Contact: <sip:6720@10.1.1.3:5060;line=gv8x1x75>;flow-id=1
P-Key-Flags: keys="3"
User-Agent: snom320/6.5.8
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer
Supported: timer, 100rel, replaces, callerid
Session-Expires: 3600;refresher=uas
Min-SE: 90
Content-Type: application/sdp

Jun 10 06:38:13 DEBUG[22367] chan_sip.c: Header 0: INVITE sip:0860121218@pabx.itws.co.za;user=phone SIP/2.0 (56)
Jun 10 06:38:13 DEBUG[22367] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.1.1.3:5060;branch=z9hG4bK-nq8yy222qscg;rport (64)
Jun 10 06:38:13 DEBUG[22367] chan_sip.c: Header 2: From: "6720" <sip:6720@pabx.itws.co.za>;tag=2ps68pkrvk (54)
Jun 10 06:38:13 DEBUG[22367] chan_sip.c: Header 3: To: <sip:0860121218@pabx.itws.co.za;user=phone> (47)
Jun 10 06:38:13 DEBUG[22367] chan_sip.c: Header 4: Call-ID: 3c2a60a8493e-7gnq2vbwskte@snom320-000413242316 (55)
Jun 10 06:38:13 DEBUG[22367] chan_sip.c: Header 5: CSeq: 1 INVITE (14)
Jun 10 06:38:13 DEBUG[22367] chan_sip.c: Header 6: Max-Forwards: 70 (16)
Jun 10 06:38:13 DEBUG[22367] chan_sip.c: Header 7: Contact: <sip:6720@10.1.1.3:5060;line=gv8x1x75>;flow-id=1 (57)
Jun 10 06:38:13 DEBUG[22367] chan_sip.c: Header 8: P-Key-Flags: keys="3" (21)
Jun 10 06:38:13 DEBUG[22367] chan_sip.c: Header 9: User-Agent: snom320/6.5.8 (25)
Jun 10 06:38:13 DEBUG[22367] chan_sip.c: Header 10: Accept: application/sdp (23)
Jun 10 06:38:13 DEBUG[22367] chan_sip.c: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO (88)
Jun 10 06:38:13 DEBUG[22367] chan_sip.c: Header 12: Allow-Events: talk, hold, refer (31)
Jun 10 06:38:13 DEBUG[22367] chan_sip.c: Header 13: Supported: timer, 100rel, replaces, callerid (44)
Jun 10 06:38:13 DEBUG[22367] chan_sip.c: Header 14: Session-Expires: 3600;refresher=uas (35)
Jun 10 06:38:13 DEBUG[22367] chan_sip.c: Header 15: Min-SE: 90 (10)
Jun 10 06:38:13 DEBUG[22367] chan_sip.c: Header 16: Content-Type: application/sdp (29)
Jun 10 06:38:13 DEBUG[22367] chan_sip.c: Header 17: Content-Length: 419 (19)
Jun 10 06:38:13 DEBUG[22367] chan_sip.c: Header 18:  (0)
Jun 10 06:38:13 DEBUG[22367] chan_sip.c: Line: v=0 (3)
Jun 10 06:38:13 DEBUG[22367] chan_sip.c: Line: o=root 1713850934 1713850934 IN IP4 10.1.1.3 (44)
Jun 10 06:38:13 DEBUG[22367] chan_sip.c: Line: s=call (6)
Jun 10 06:38:13 DEBUG[22367] chan_sip.c: Line: c=IN IP4 10.1.1.3 (17)
Jun 10 06:38:13 DEBUG[22367] chan_sip.c: Line: t=0 0 (5)
Jun 10 06:38:13 DEBUG[22367] chan_sip.c: Line: m=audio 62792 RTP/AVP 18 3 9 2 4 101 (36)
Jun 10 06:38:13 DEBUG[22367] chan_sip.c: Line: a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:wS0kjUaq/tZhYtnDL8SbHDqs2RlqpR626PyL6SJT (82)
Jun 10 06:38:13 DEBUG[22367] chan_sip.c: Line: a=rtpmap:18 g729/8000 (21)
Jun 10 06:38:13 DEBUG[22367] chan_sip.c: Line: a=rtpmap:3 gsm/8000 (19)
Jun 10 06:38:13 DEBUG[22367] chan_sip.c: Line: a=rtpmap:2 g726-32/8000 (23)
Jun 10 06:38:13 DEBUG[22367] chan_sip.c: Line: a=rtpmap:4 g723/8000 (20)
Jun 10 06:38:13 DEBUG[22367] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33)
Jun 10 06:38:13 DEBUG[22367] chan_sip.c: Line: a=fmtp:101 0-16 (15)
Jun 10 06:38:13 DEBUG[22367] chan_sip.c: Line: a=ptime:20 (10)
Jun 10 06:38:13 DEBUG[22367] chan_sip.c: Line: a=encryption:optional (21)
Jun 10 06:38:13 DEBUG[22367] chan_sip.c: Line: a=sendrecv (10)
Jun 10 06:38:13 VERBOSE[22367] logger.c: --- (18 headers 17 lines) ---
Jun 10 06:38:13 DEBUG[22367] chan_sip.c: Allocating new SIP dialog for 3c2a60a8493e-7gnq2vbwskte@snom320-000413242316 - INVITE (With RTP)
Jun 10 06:38:13 DEBUG[22367] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE
Jun 10 06:38:13 DEBUG[22367] chan_sip.c: Begin: parsing SIP "Supported: timer, 100rel, replaces, callerid"
Jun 10 06:38:13 DEBUG[22367] chan_sip.c: Found SIP option: -timer-
Jun 10 06:38:13 DEBUG[22367] chan_sip.c: Matched SIP option: timer
Jun 10 06:38:13 DEBUG[22367] chan_sip.c: Found SIP option: -100rel-
Jun 10 06:38:13 DEBUG[22367] chan_sip.c: Matched SIP option: 100rel
Jun 10 06:38:13 DEBUG[22367] chan_sip.c: Found SIP option: -replaces-
Jun 10 06:38:13 DEBUG[22367] chan_sip.c: Matched SIP option: replaces
Jun 10 06:38:13 DEBUG[22367] chan_sip.c: Found SIP option: -callerid-
Jun 10 06:38:13 DEBUG[22367] chan_sip.c: Found no match for SIP option: callerid (Please file bug report!)
Jun 10 06:38:13 DEBUG[22367] chan_sip.c: * SIP extension value: 7 for call 3c2a60a8493e-7gnq2vbwskte@snom320-000413242316
Jun 10 06:38:13 VERBOSE[22367] logger.c: Using INVITE request as basis request - 3c2a60a8493e-7gnq2vbwskte@snom320-000413242316
Jun 10 06:38:13 VERBOSE[22367] logger.c: Sending to 10.1.1.3 : 5060 (NAT)
Jun 10 06:38:13 DEBUG[22367] chan_sip.c: Setting NAT on RTP to 524288
Jun 10 06:38:13 VERBOSE[22367] logger.c: Reliably Transmitting (NAT) to 220.238.243.11:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.1.1.3:5060;branch=z9hG4bK-nq8yy222qscg;received=220.238.243.11;rport=5060
From: "6720" <sip:6720@pabx.itws.co.za>;tag=2ps68pkrvk
To: <sip:0860121218@pabx.itws.co.za;user=phone>;tag=as2a09ac5e
Call-ID: 3c2a60a8493e-7gnq2vbwskte@snom320-000413242316
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6a3b79bb"
Content-Length: 0


---
Jun 10 06:38:13 DEBUG[22367] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id  #1356665
Jun 10 06:38:13 VERBOSE[22367] logger.c: Scheduling destruction of call '3c2a60a8493e-7gnq2vbwskte@snom320-000413242316' in 15000 ms
Jun 10 06:38:13 VERBOSE[22367] logger.c: Found user '6720'
Jun 10 06:38:14 VERBOSE[22367] logger.c:

Jun 10 06:38:13 VERBOSE[22367] logger.c: Sending to 10.1.1.3 : 5060 (NAT)
Jun 10 06:38:13 DEBUG[22367] chan_sip.c: Setting NAT on RTP to 524288
Jun 10 06:38:13 VERBOSE[22367] logger.c: Reliably Transmitting (NAT) to 220.238.243.11:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.1.1.3:5060;branch=z9hG4bK-nq8yy222qscg;received=220.238.243.11;rport=5060
From: "6720" <sip:6720@pabx.itws.co.za>;tag=2ps68pkrvk
To: <sip:0860121218@pabx.itws.co.za;user=phone>;tag=as2a09ac5e
Call-ID: 3c2a60a8493e-7gnq2vbwskte@snom320-000413242316
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6a3b79bb"
Content-Length: 0


---
Jun 10 06:38:13 DEBUG[22367] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id  #1356665
Jun 10 06:38:13 VERBOSE[22367] logger.c: Scheduling destruction of call '3c2a60a8493e-7gnq2vbwskte@snom320-000413242316' in 15000 ms
Jun 10 06:38:13 VERBOSE[22367] logger.c: Found user '6720'
Jun 10 06:38:14 VERBOSE[22367] logger.c:
<-- SIP read from 220.238.243.11:5060:
INVITE sip:0860121218@pabx.itws.co.za;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.1.1.3:5060;branch=z9hG4bK-nq8yy222qscg;rport
From: "6720" <sip:6720@pabx.itws.co.za>;tag=2ps68pkrvk
To: <sip:0860121218@pabx.itws.co.za;user=phone>
Call-ID: 3c2a60a8493e-7gnq2vbwskte@snom320-000413242316
CSeq: 1 INVITE
Max-Forwards: 70
Contact: <sip:6720@10.1.1.3:5060;line=gv8x1x75>;flow-id=1
P-Key-Flags: keys="3"
User-Agent: snom320/6.5.8
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer
Supported: timer, 100rel, replaces, callerid
Session-Expires: 3600;refresher=uas
Min-SE: 90
Content-Type: application/sdp
Content-Length: 419
v=0
o=root 1713850934 1713850934 IN IP4 10.1.1.3
s=call
c=IN IP4 10.1.1.3
t=0 0
m=audio 62792 RTP/AVP 18 3 9 2 4 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:wS0kjUaq/tZhYtnDL8SbHDqs2RlqpR626PyL6SJT
a=rtpmap:18 g729/8000
a=rtpmap:3 gsm/8000
a=rtpmap:9 g722/8000
a=rtpmap:2 g726-32/8000
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=encryption:optional
a=sendrecv

Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Header 0: INVITE sip:0860121218@pabx.itws.co.za;user=phone SIP/2.0 (56)
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.1.1.3:5060;branch=z9hG4bK-nq8yy222qscg;rport (64)
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Header 2: From: "6720" <sip:6720@pabx.itws.co.za>;tag=2ps68pkrvk (54)
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Header 3: To: <sip:0860121218@pabx.itws.co.za;user=phone> (47)

s=call
c=IN IP4 10.1.1.3
t=0 0
m=audio 62792 RTP/AVP 18 3 9 2 4 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:wS0kjUaq/tZhYtnDL8SbHDqs2RlqpR626PyL6SJT
a=rtpmap:18 g729/8000
a=rtpmap:3 gsm/8000
a=rtpmap:9 g722/8000
a=rtpmap:2 g726-32/8000
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=encryption:optional
a=sendrecv

Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Header 0: INVITE sip:0860121218@pabx.itws.co.za;user=phone SIP/2.0 (56)
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.1.1.3:5060;branch=z9hG4bK-nq8yy222qscg;rport (64)
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Header 2: From: "6720" <sip:6720@pabx.itws.co.za>;tag=2ps68pkrvk (54)
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Header 3: To: <sip:0860121218@pabx.itws.co.za;user=phone> (47)
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Header 4: Call-ID: 3c2a60a8493e-7gnq2vbwskte@snom320-000413242316 (55)
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Header 5: CSeq: 1 INVITE (14)
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Header 6: Max-Forwards: 70 (16)
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Header 7: Contact: <sip:6720@10.1.1.3:5060;line=gv8x1x75>;flow-id=1 (57)
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Header 8: P-Key-Flags: keys="3" (21)
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Header 9: User-Agent: snom320/6.5.8 (25)
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Header 10: Accept: application/sdp (23)
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO (88)
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Header 12: Allow-Events: talk, hold, refer (31)
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Header 13: Supported: timer, 100rel, replaces, callerid (44)
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Header 14: Session-Expires: 3600;refresher=uas (35)
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Header 15: Min-SE: 90 (10)
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Header 16: Content-Type: application/sdp (29)
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Header 17: Content-Length: 419 (19)
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Header 18:  (0)
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Line: v=0 (3)
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Line: o=root 1713850934 1713850934 IN IP4 10.1.1.3 (44)
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Line: s=call (6)
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Line: c=IN IP4 10.1.1.3 (17)
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Line: t=0 0 (5)
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Line: m=audio 62792 RTP/AVP 18 3 9 2 4 101 (36)

Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Line: t=0 0 (5)
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Line: m=audio 62792 RTP/AVP 18 3 9 2 4 101 (36)
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Line: a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:wS0kjUaq/tZhYtnDL8SbHDqs2RlqpR626PyL6SJT (82)
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Line: a=rtpmap:18 g729/8000 (21)
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Line: a=rtpmap:3 gsm/8000 (19)
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Line: a=rtpmap:9 g722/8000 (20)
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Line: a=rtpmap:2 g726-32/8000 (23)
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Line: a=rtpmap:4 g723/8000 (20)
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33)
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Line: a=fmtp:101 0-16 (15)
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Line: a=ptime:20 (10)
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Line: a=encryption:optional (21)
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Line: a=sendrecv (10)
Jun 10 06:38:14 VERBOSE[22367] logger.c: --- (18 headers 17 lines) ---
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Ignoring SIP message because of retransmit (INVITE Seqno 1, ours 1)
Jun 10 06:38:14 VERBOSE[22367] logger.c: Ignoring this INVITE request
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Got a SIP re-invite for call 3c2a60a8493e-7gnq2vbwskte@snom320-000413242316
Jun 10 06:38:14 VERBOSE[22367] logger.c:
<-- SIP read from 220.238.243.11:5060:
ACK sip:0860121218@pabx.itws.co.za;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.1.1.3:5060;branch=z9hG4bK-nq8yy222qscg;rport
From: "6720" <sip:6720@pabx.itws.co.za>;tag=2ps68pkrvk
To: <sip:0860121218@pabx.itws.co.za;user=phone>;tag=as2a09ac5e
Call-ID: 3c2a60a8493e-7gnq2vbwskte@snom320-000413242316
CSeq: 1 ACK
Max-Forwards: 70
Contact: <sip:6720@10.1.1.3:5060;line=gv8x1x75>;flow-id=1
Content-Length: 0


Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Header 0: ACK sip:0860121218@pabx.itws.co.za;user=phone SIP/2.0 (53)
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.1.1.3:5060;branch=z9hG4bK-nq8yy222qscg;rport (64)
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Header 2: From: "6720" <sip:6720@pabx.itws.co.za>;tag=2ps68pkrvk (54)
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Header 3: To: <sip:0860121218@pabx.itws.co.za;user=phone>;tag=as2a09ac5e (62)
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Header 4: Call-ID: 3c2a60a8493e-7gnq2vbwskte@snom320-000413242316 (55)
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Header 5: CSeq: 1 ACK (11)
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Header 6: Max-Forwards: 70 (16)
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Header 7: Contact: <sip:6720@10.1.1.3:5060;line=gv8x1x75>;flow-id=1 (57)
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Header 8: Content-Length: 0 (17)
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Header 9:  (0)
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Header 8: Content-Length: 0 (17)
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Header 9:  (0)
Jun 10 06:38:14 VERBOSE[22367] logger.c: --- (9 headers 0 lines) ---
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: **** Received ACK (6) - Command in SIP ACK
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #1356665
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Stopping retransmission on '3c2a60a8493e-7gnq2vbwskte@snom320-000413242316' of Response 1: Match Found
Jun 10 06:38:14 VERBOSE[22367] logger.c:
<-- SIP read from 220.238.243.11:5060:
INVITE sip:0860121218@pabx.itws.co.za;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.1.1.3:5060;branch=z9hG4bK-90ifcmbqip4r;rport
From: "6720" <sip:6720@pabx.itws.co.za>;tag=2ps68pkrvk
To: <sip:0860121218@pabx.itws.co.za;user=phone>
Call-ID: 3c2a60a8493e-7gnq2vbwskte@snom320-000413242316
CSeq: 2 INVITE
Max-Forwards: 70
Contact: <sip:6720@10.1.1.3:5060;line=gv8x1x75>;flow-id=1
P-Key-Flags: keys="3"
User-Agent: snom320/6.5.8
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer
Supported: timer, 100rel, replaces, callerid
Session-Expires: 3600;refresher=uas
Min-SE: 90
Proxy-Authorization: Digest username="6720",realm="asterisk",nonce="6a3b79bb",uri="sip:0860121218@pabx.itws.co.za;user=phone",response="a3f15ff5371956d279c656252c3acace",algori$
Content-Type: application/sdp
Content-Length: 419

v=0
o=root 1713850934 1713850934 IN IP4 10.1.1.3
s=call
c=IN IP4 10.1.1.3
t=0 0
m=audio 62792 RTP/AVP 18 3 9 2 4 101
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:wS0kjUaq/tZhYtnDL8SbHDqs2RlqpR626PyL6SJT
a=rtpmap:18 g729/8000
a=rtpmap:3 gsm/8000
a=rtpmap:9 g722/8000
a=rtpmap:2 g726-32/8000
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=encryption:optional
a=sendrecv

Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Header 0: INVITE sip:0860121218@pabx.itws.co.za;user=phone SIP/2.0 (56)
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Header 1: Via: SIP/2.0/UDP 10.1.1.3:5060;branch=z9hG4bK-90ifcmbqip4r;rport (64)
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Header 2: From: "6720" <sip:6720@pabx.itws.co.za>;tag=2ps68pkrvk (54)
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Header 3: To: <sip:0860121218@pabx.itws.co.za;user=phone> (47)
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Header 4: Call-ID: 3c2a60a8493e-7gnq2vbwskte@snom320-000413242316 (55)
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Header 5: CSeq: 2 INVITE (14)
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Header 6: Max-Forwards: 70 (16)
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Header 7: Contact: <sip:6720@10.1.1.3:5060;line=gv8x1x75>;flow-id=1 (57)
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Header 8: P-Key-Flags: keys="3" (21)
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Header 9: User-Agent: snom320/6.5.8 (25)
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Header 10: Accept: application/sdp (23)
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Header 11: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO (88)
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Header 12: Allow-Events: talk, hold, refer (31)
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Header 13: Supported: timer, 100rel, replaces, callerid (44)
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Header 14: Session-Expires: 3600;refresher=uas (35)
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Header 15: Min-SE: 90 (10)
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Header 16: Proxy-Authorization: Digest username="6720",realm="asterisk",nonce="6a3b79bb",uri="sip:0860121218@pabx.itws.co.za;user=phone$
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Header 17: Content-Type: application/sdp (29)
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Header 18: Content-Length: 419 (19)
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Header 19:  (0)
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Line: v=0 (3)
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Line: o=root 1713850934 1713850934 IN IP4 10.1.1.3 (44)
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Line: s=call (6)
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Line: c=IN IP4 10.1.1.3 (17)
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Line: t=0 0 (5)
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Line: m=audio 62792 RTP/AVP 18 3 9 2 4 101 (36)
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Line: a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:wS0kjUaq/tZhYtnDL8SbHDqs2RlqpR626PyL6SJT (82)
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Line: a=rtpmap:18 g729/8000 (21)
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Line: a=rtpmap:3 gsm/8000 (19)
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Line: a=rtpmap:9 g722/8000 (20)
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Line: a=rtpmap:2 g726-32/8000 (23)
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Line: a=rtpmap:4 g723/8000 (20)
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33)
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Line: a=fmtp:101 0-16 (15)
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Line: a=rtpmap:101 telephone-event/8000 (33)
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Line: a=fmtp:101 0-16 (15)
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Line: a=ptime:20 (10)
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Line: a=encryption:optional (21)
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Line: a=sendrecv (10)
Jun 10 06:38:14 VERBOSE[22367] logger.c: --- (19 headers 17 lines) ---
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: **** Received INVITE (5) - Command in SIP INVITE
Jun 10 06:38:14 VERBOSE[22367] logger.c: Using INVITE request as basis request - 3c2a60a8493e-7gnq2vbwskte@snom320-000413242316
Jun 10 06:38:14 VERBOSE[22367] logger.c: Sending to 10.1.1.3 : 5060 (NAT)
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Setting NAT on RTP to 524288
Jun 10 06:38:14 VERBOSE[22367] logger.c: Found user '6720'
Jun 10 06:38:14 VERBOSE[22367] logger.c: Found RTP audio format 18
Jun 10 06:38:14 VERBOSE[22367] logger.c: Found RTP audio format 3
Jun 10 06:38:14 VERBOSE[22367] logger.c: Found RTP audio format 9
Jun 10 06:38:14 VERBOSE[22367] logger.c: Found RTP audio format 2
Jun 10 06:38:14 VERBOSE[22367] logger.c: Found RTP audio format 4
Jun 10 06:38:14 VERBOSE[22367] logger.c: Found RTP audio format 101
Jun 10 06:38:14 VERBOSE[22367] logger.c: Peer audio RTP is at port 10.1.1.3:62792
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Peer audio RTP is at port 10.1.1.3:62792
Jun 10 06:38:14 VERBOSE[22367] logger.c: Found description format g729
Jun 10 06:38:14 VERBOSE[22367] logger.c: Found description format gsm
Jun 10 06:38:14 VERBOSE[22367] logger.c: Found description format g722
Jun 10 06:38:14 VERBOSE[22367] logger.c: Found description format g726-32
Jun 10 06:38:14 VERBOSE[22367] logger.c: Found description format g723
Jun 10 06:38:14 VERBOSE[22367] logger.c: Found description format telephone-event
Jun 10 06:38:14 VERBOSE[22367] logger.c: Capabilities: us - 0x10c (ulaw|alaw|g729), peer - audio=0x113 (g723|gsm|g726|g729)/video=0x0 (nothing), combined - 0x100 (g729)
Jun 10 06:38:14 VERBOSE[22367] logger.c: Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Checking SIP call limits for device 6720
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: Updating call counter for incoming call
Jun 10 06:38:14 VERBOSE[22367] logger.c: Looking for 0860121218 in from-internal (domain pabx.itws.co.za;user=phone)
Jun 10 06:38:14 DEBUG[22367] chan_sip.c: build_route: Contact hop: <sip:6720@10.1.1.3:5060;line=gv8x1x75>;flow-id=1
Jun 10 06:38:14 VERBOSE[22367] logger.c: list_route: hop: <sip:6720@10.1.1.3:5060;line=gv8x1x75>
Jun 10 06:38:14 VERBOSE[22367] logger.c: Transmitting (NAT) to 220.238.243.11:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.1.1.3:5060;branch=z9hG4bK-90ifcmbqip4r;received=220.238.243.11;rport=5060
From: "6720" <sip:6720@pabx.itws.co.za>;tag=2ps68pkrvk
To: <sip:0860121218@pabx.itws.co.za;user=phone>
Call-ID: 3c2a60a8493e-7gnq2vbwskte@snom320-000413242316
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:0860121218@209.212.101.56>
Content-Length: 0


---
Jun 10 06:38:14 DEBUG[22353] chan_sip.c: Checking device state for peer 6720
Jun 10 06:38:14 DEBUG[22353] devicestate.c: Changing state for SIP/6720 - state 2 (In use)
Jun 10 06:38:14 DEBUG[22353] chan_sip.c: Checking device state for peer 6720
Jun 10 06:38:14 DEBUG[15969] pbx.c: Launching 'Macro'
Jun 10 06:38:14 VERBOSE[15969] logger.c:     -- Executing Macro("SIP/6720-0892aa10", "pinsets|1|1") in new stack
Jun 10 06:38:14 DEBUG[15969] pbx.c: Launching 'GotoIf'
Jun 10 06:38:14 VERBOSE[15969] logger.c:     -- Executing GotoIf("SIP/6720-0892aa10", "1 = 1?cdr|1") in new stack
Jun 10 06:38:14 VERBOSE[15969] logger.c:     -- Goto (macro-pinsets,cdr,1)
Jun 10 06:38:14 DEBUG[15969] pbx.c: Launching 'Authenticate'
Jun 10 06:38:14 VERBOSE[15969] logger.c:     -- Executing Authenticate("SIP/6720-0892aa10", "/etc/asterisk/pinset_1|a") in new stack
Jun 10 06:38:14 DEBUG[15969] chan_sip.c: sip_answer(SIP/6720-0892aa10)
Jun 10 06:38:14 VERBOSE[15969] logger.c: We're at 209.212.101.56 port 14464
Jun 10 06:38:14 VERBOSE[15969] logger.c: Adding codec 0x100 (g729) to SDP
Jun 10 06:38:14 VERBOSE[15969] logger.c: Adding non-codec 0x1 (telephone-event) to SDP
Jun 10 06:38:14 VERBOSE[15969] logger.c: Reliably Transmitting (NAT) to 220.238.243.11:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.1.1.3:5060;branch=z9hG4bK-90ifcmbqip4r;received=220.238.243.11;rport=5060
From: "6720" <sip:6720@pabx.itws.co.za>;tag=2ps68pkrvk
To: <sip:0860121218@pabx.itws.co.za;user=phone>;tag=as12f04922
Call-ID: 3c2a60a8493e-7gnq2vbwskte@snom320-000413242316
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:0860121218@209.212.101.56>
Content-Type: application/sdp
Content-Length: 243

v=0
o=root 22348 22348 IN IP4 209.212.101.56
s=session
c=IN IP4 209.212.101.56
t=0 0
m=audio 14464 RTP/AVP 18 101
a=rtpmap:18 G729/8000
m=audio 14464 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
Jun 10 06:38:14 DEBUG[15969] chan_sip.c: *** SIP TIMER: Initalizing retransmit timer on packet: Id  #1356667
Jun 10 06:38:14 DEBUG[15969] channel.c: Set channel SIP/6720-0892aa10 to write format gsm
Jun 10 06:38:14 DEBUG[15969] rtp.c: Ooh, format changed from unknown to g729
Jun 10 06:38:14 VERBOSE[15969] logger.c:     -- Playing 'agent-pass' (language 'en')
Jun 10 06:38:14 DEBUG[15970] app_queue.c: Device 'SIP/6720' changed to state '2' (In use) but we don't care because they're not a member of any queue.
Jun 10 06:38:14 DEBUG[22353] chan_sip.c: Checking device state for peer 6720
Jun 10 06:38:14 DEBUG[22353] devicestate.c: Changing state for SIP/6720 - state 2 (In use)
Jun 10 06:38:14 DEBUG[22353] chan_sip.c: Checking device state for peer 6720
Jun 10 06:38:14 DEBUG[15971] app_queue.c: Device 'SIP/6720' changed to state '2' (In use) but we don't care because they're not a member of any queue.
Jun 10 06:38:14 VERBOSE[22367] logger.c:
<-- SIP read from 220.238.243.11:5060:
INVITE sip:0860121218@pabx.itws.co.za;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.1.1.3:5060;branch=z9hG4bK-90ifcmbqip4r;rport
From: "6720" <sip:6720@pabx.itws.co.za>;tag=2ps68pkrvk
To: <sip:0860121218@pabx.itws.co.za;user=phone>
Call-ID: 3c2a60a8493e-7gnq2vbwskte@snom320-000413242316
CSeq: 2 INVITE
Max-Forwards: 70
Contact: <sip:6720@10.1.1.3:5060;line=gv8x1x75>;flow-id=1
P-Key-Flags: keys="3"
User-Agent: snom320/6.5.8
Accept: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO
Allow-Events: talk, hold, refer
Supported: timer, 100rel, replaces, callerid
Session-Expires: 3600;refresher=uas
Min-SE: 90
Proxy-Authorization: Digest username="6720",realm="asterisk",nonce="6a3b79bb",uri="sip:0860121218@pabx.itws.co.za;user=phone",response="a3f15ff5371956d279c656252c3acace",algori$
Content-Type: application/sdp
Content-Length: 419

v=0
o=root 1713850934 1713850934 IN IP4 10.1.1.3