
Use 'exit' when done

[0;37;40m[0m[1;30;40m  == [0;37;40mParsing '/etc/asterisk/asterisk.conf': Found
[1;30;40m  == [0;37;40mParsing '/etc/asterisk/extconfig.conf': Found
Asterisk 1.2.9.1, Copyright (C) 1999 - 2006 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'show license' for details.
=========================================================================
Connected to Asterisk 1.2.9.1 currently running on devcentos64 (pid = 24142)
devcentos64*CLI> Verbosity was 4 and is now 5
Core debug is at least 4
[Kdevcentos64*CLI> 
<-- SIP read from 192.168.1.38:5060: 
INVITE sip:58@192.168.1.10 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.38:5060;branch=z9hG4bK-44ba0616
From: "Jerry Geis" <sip:404@192.168.1.10>;tag=48dd2341203ea66o0
To: <sip:58@192.168.1.10>
Call-ID: de93b0cc-f369dcae@192.168.1.38
CSeq: 101 INVITE
Max-Forwards: 70
Contact: "Jerry Geis" <sip:404@192.168.1.38:5060>
Expires: 240
User-Agent: Linksys/SPA942-4.1.12
Proxy-Require: 192.168.1.10
Content-Length: 397
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Content-Type: application/sdp

v=0
o=- 2589835 2589835 IN IP4 192.168.1.38
s=-
c=IN IP4 192.168.1.38
t=0 0
m=audio 16422 RTP/AVP 0 2 4 8 18 96 97 98 101
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 0: INVITE sip:58@192.168.1.10 SIP/2.0 (34)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 1: Via: SIP/2.0/UDP 192.168.1.38:5060;branch=z9hG4bK-44ba0616 (58)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 2: From: "Jerry Geis" <sip:404@192.168.1.10>;tag=48dd2341203ea66o0 (63)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 3: To: <sip:58@192.168.1.10> (25)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 4: Call-ID: de93b0cc-f369dcae@192.168.1.38 (39)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 5: CSeq: 101 INVITE (16)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 6: Max-Forwards: 70 (16)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 7: Contact: "Jerry Geis" <sip:404@192.168.1.38:5060> (49)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 8: Expires: 240 (12)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 9: User-Agent: Linksys/SPA942-4.1.12 (33)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 10: Proxy-Require: 192.168.1.10 (27)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 11: Content-Length: 397 (19)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 12: Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER (61)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 13: Content-Type: application/sdp (29)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 14:  (0)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3395 parse_request:  Line: v=0 (3)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3395 parse_request:  Line: o=- 2589835 2589835 IN IP4 192.168.1.38 (39)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3395 parse_request:  Line: s=- (3)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3395 parse_request:  Line: c=IN IP4 192.168.1.38 (21)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3395 parse_request:  Line: t=0 0 (5)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3395 parse_request:  Line: m=audio 16422 RTP/AVP 0 2 4 8 18 96 97 98 101 (45)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3395 parse_request:  Line: a=rtpmap:0 PCMU/8000 (20)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3395 parse_request:  Line: a=rtpmap:2 G726-32/8000 (23)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3395 parse_request:  Line: a=rtpmap:4 G723/8000 (20)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3395 parse_request:  Line: a=rtpmap:8 PCMA/8000 (20)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3395 parse_request:  Line: a=rtpmap:18 G729a/8000 (22)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3395 parse_request:  Line: a=rtpmap:96 G726-40/8000 (24)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3395 parse_request:  Line: a=rtpmap:97 G726-24/8000 (24)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3395 parse_request:  Line: a=rtpmap:98 G726-16/8000 (24)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3395 parse_request:  Line: a=rtpmap:101 telephone-event/8000 (33)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3395 parse_request:  Line: a=fmtp:101 0-15 (15)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3395 parse_request:  Line: a=ptime:30 (10)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3395 parse_request:  Line: a=sendrecv (10)
 --- (14 headers 18 lines)---
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3147 sip_alloc:  Allocating new SIP dialog for de93b0cc-f369dcae@192.168.1.38 - INVITE (With RTP)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:11137 handle_request:  **** Received INVITE (5) - Command in SIP INVITE
 Using INVITE request as basis request - de93b0cc-f369dcae@192.168.1.38
 Sending to 192.168.1.38 : 5060 (non-NAT)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:7155 check_user_full:  Setting NAT on RTP to 0
 Reliably Transmitting (no NAT) to 192.168.1.38:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.38:5060;branch=z9hG4bK-44ba0616;received=192.168.1.38
From: "Jerry Geis" <sip:404@192.168.1.10>;tag=48dd2341203ea66o0
To: <sip:58@192.168.1.10>;tag=as509a40e1
Call-ID: de93b0cc-f369dcae@192.168.1.38
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:58@192.168.1.10>
Proxy-Authenticate: Digest realm="asterisk", nonce="360d1285"
Content-Length: 0


---
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:1293 __sip_reliable_xmit:  *** SIP TIMER: Initalizing retransmit timer on packet: Id  #12
 Scheduling destruction of call 'de93b0cc-f369dcae@192.168.1.38' in 15000 ms
 Found user '404'
 [Kdevcentos64*CLI> 
<-- SIP read from 192.168.1.38:5060: 
ACK sip:58@192.168.1.10 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.38:5060;branch=z9hG4bK-44ba0616
From: "Jerry Geis" <sip:404@192.168.1.10>;tag=48dd2341203ea66o0
To: <sip:58@192.168.1.10>;tag=as509a40e1
Call-ID: de93b0cc-f369dcae@192.168.1.38
CSeq: 101 ACK
Max-Forwards: 70
Contact: "Jerry Geis" <sip:404@192.168.1.38:5060>
User-Agent: Linksys/SPA942-4.1.12
Proxy-Require: 192.168.1.10
Content-Length: 0


 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 0: ACK sip:58@192.168.1.10 SIP/2.0 (31)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 1: Via: SIP/2.0/UDP 192.168.1.38:5060;branch=z9hG4bK-44ba0616 (58)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 2: From: "Jerry Geis" <sip:404@192.168.1.10>;tag=48dd2341203ea66o0 (63)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 3: To: <sip:58@192.168.1.10>;tag=as509a40e1 (40)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 4: Call-ID: de93b0cc-f369dcae@192.168.1.38 (39)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 5: CSeq: 101 ACK (13)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 6: Max-Forwards: 70 (16)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 7: Contact: "Jerry Geis" <sip:404@192.168.1.38:5060> (49)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 8: User-Agent: Linksys/SPA942-4.1.12 (33)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 9: Proxy-Require: 192.168.1.10 (27)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 10: Content-Length: 0 (17)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 11:  (0)
 --- (11 headers 0 lines)---
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:11137 handle_request:  **** Received ACK (6) - Command in SIP ACK
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:1390 __sip_ack:  ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #12
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:1401 __sip_ack:  Stopping retransmission on 'de93b0cc-f369dcae@192.168.1.38' of Response 101: Match Found
 [Kdevcentos64*CLI> 
<-- SIP read from 192.168.1.38:5060: 
INVITE sip:58@192.168.1.10 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.38:5060;branch=z9hG4bK-62355802
From: "Jerry Geis" <sip:404@192.168.1.10>;tag=48dd2341203ea66o0
To: <sip:58@192.168.1.10>
Call-ID: de93b0cc-f369dcae@192.168.1.38
CSeq: 102 INVITE
Max-Forwards: 70
Proxy-Authorization: Digest username="404",realm="asterisk",nonce="360d1285",uri="sip:58@192.168.1.10",algorithm=MD5,response="69c167c9fe14c34048ee83974b737254"
Contact: "Jerry Geis" <sip:404@192.168.1.38:5060>
Expires: 240
User-Agent: Linksys/SPA942-4.1.12
Proxy-Require: 192.168.1.10
Content-Length: 397
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Content-Type: application/sdp

v=0
o=- 2589835 2589835 IN IP4 192.168.1.38
s=-
c=IN IP4 192.168.1.38
t=0 0
m=audio 16422 RTP/AVP 0 2 4 8 18 96 97 98 101
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 0: INVITE sip:58@192.168.1.10 SIP/2.0 (34)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 1: Via: SIP/2.0/UDP 192.168.1.38:5060;branch=z9hG4bK-62355802 (58)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 2: From: "Jerry Geis" <sip:404@192.168.1.10>;tag=48dd2341203ea66o0 (63)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 3: To: <sip:58@192.168.1.10> (25)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 4: Call-ID: de93b0cc-f369dcae@192.168.1.38 (39)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 5: CSeq: 102 INVITE (16)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 6: Max-Forwards: 70 (16)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 7: Proxy-Authorization: Digest username="404",realm="asterisk",nonce="360d1285",uri="sip:58@192.168.1.10",algorithm=MD5,response="69c167c9fe14c34048ee83974b737254" (160)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 8: Contact: "Jerry Geis" <sip:404@192.168.1.38:5060> (49)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 9: Expires: 240 (12)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 10: User-Agent: Linksys/SPA942-4.1.12 (33)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 11: Proxy-Require: 192.168.1.10 (27)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 12: Content-Length: 397 (19)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 13: Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER (61)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 14: Content-Type: application/sdp (29)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 15:  (0)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3395 parse_request:  Line: v=0 (3)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3395 parse_request:  Line: o=- 2589835 2589835 IN IP4 192.168.1.38 (39)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3395 parse_request:  Line: s=- (3)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3395 parse_request:  Line: c=IN IP4 192.168.1.38 (21)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3395 parse_request:  Line: t=0 0 (5)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3395 parse_request:  Line: m=audio 16422 RTP/AVP 0 2 4 8 18 96 97 98 101 (45)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3395 parse_request:  Line: a=rtpmap:0 PCMU/8000 (20)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3395 parse_request:  Line: a=rtpmap:2 G726-32/8000 (23)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3395 parse_request:  Line: a=rtpmap:4 G723/8000 (20)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3395 parse_request:  Line: a=rtpmap:8 PCMA/8000 (20)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3395 parse_request:  Line: a=rtpmap:18 G729a/8000 (22)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3395 parse_request:  Line: a=rtpmap:96 G726-40/8000 (24)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3395 parse_request:  Line: a=rtpmap:97 G726-24/8000 (24)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3395 parse_request:  Line: a=rtpmap:98 G726-16/8000 (24)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3395 parse_request:  Line: a=rtpmap:101 telephone-event/8000 (33)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3395 parse_request:  Line: a=fmtp:101 0-15 (15)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3395 parse_request:  Line: a=ptime:30 (10)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3395 parse_request:  Line: a=sendrecv (10)
 --- (15 headers 18 lines)---
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:11137 handle_request:  **** Received INVITE (5) - Command in SIP INVITE
 Using INVITE request as basis request - de93b0cc-f369dcae@192.168.1.38
 Sending to 192.168.1.38 : 5060 (non-NAT)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:7155 check_user_full:  Setting NAT on RTP to 0
 Found user '404'
 Found RTP audio format 0
 Found RTP audio format 2
 Found RTP audio format 4
 Found RTP audio format 8
 Found RTP audio format 18
 Found RTP audio format 96
 Found RTP audio format 97
 Found RTP audio format 98
 Found RTP audio format 101
 Peer audio RTP is at port 192.168.1.38:16422
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3604 process_sdp:  Peer audio RTP is at port 192.168.1.38:16422
 Found description format PCMU
 Found description format G726-32
 Found description format G723
 Found description format PCMA
 Found description format G729a
 Found description format G726-40
 Found description format G726-24
 Found description format G726-16
 Found description format telephone-event
 Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x51d (g723|ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
 Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:10497 handle_request_invite:  Checking SIP call limits for device 404
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:2209 update_call_counter:  Updating call counter for incoming call
 Looking for 58 in smvoice-sip (domain 192.168.1.10)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:6137 build_route:  build_route: Contact hop: "Jerry Geis" <sip:404@192.168.1.38:5060>
 list_route: hop: <sip:404@192.168.1.38:5060>
 Transmitting (no NAT) to 192.168.1.38:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.38:5060;branch=z9hG4bK-62355802;received=192.168.1.38
From: "Jerry Geis" <sip:404@192.168.1.10>;tag=48dd2341203ea66o0
To: <sip:58@192.168.1.10>
Call-ID: de93b0cc-f369dcae@192.168.1.38
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:58@192.168.1.10>
Content-Length: 0


---
 [Kdevcentos64*CLI> Jul 10 16:31:29 DEBUG[24146]: chan_sip.c:11668 sip_devicestate:  Checking device state for peer 404
 Jul 10 16:31:29 DEBUG[24146]: devicestate.c:187 do_state_change:  Changing state for SIP/404 - state 2 (In use)
 Jul 10 16:31:29 DEBUG[24178]: pbx.c:1677 pbx_extension_helper:  Launching 'Dial'
     -- Executing Dial("SIP/404-56d1", "SIP/402|20|tTD(101:101)") in new stack
 Jul 10 16:31:29 DEBUG[24178]: chan_sip.c:3147 sip_alloc:  Allocating new SIP dialog for (No Call-ID) - INVITE (With RTP)
 Jul 10 16:31:29 DEBUG[24178]: chan_sip.c:1874 create_addr_from_peer:  Setting NAT on RTP to 0
 Jul 10 16:31:29 DEBUG[24178]: channel.c:2823 ast_channel_inherit_variables:  Not copying variable STACK-smvoice-sip-58-1.
 Jul 10 16:31:29 DEBUG[24178]: channel.c:2823 ast_channel_inherit_variables:  Not copying variable SIPCALLID.
 Jul 10 16:31:29 DEBUG[24178]: channel.c:2823 ast_channel_inherit_variables:  Not copying variable SIPUSERAGENT.
 Jul 10 16:31:29 DEBUG[24178]: channel.c:2823 ast_channel_inherit_variables:  Not copying variable SIPDOMAIN.
 Jul 10 16:31:29 DEBUG[24178]: channel.c:2823 ast_channel_inherit_variables:  Not copying variable SIPURI.
 Jul 10 16:31:29 DEBUG[24178]: chan_sip.c:2068 sip_call:  Outgoing Call for 402
 Jul 10 16:31:29 DEBUG[24178]: chan_sip.c:2209 update_call_counter:  Updating call counter for outgoing call
 We're at 192.168.1.10 port 12804
 Adding codec 0x4 (ulaw) to SDP
 Adding codec 0x8 (alaw) to SDP
 Adding codec 0x2 (gsm) to SDP
 Adding non-codec 0x1 (telephone-event) to SDP
 Jul 10 16:31:29 DEBUG[24178]: chan_sip.c:3363 parse_request:  Header 0: INVITE sip:402@192.168.1.162 SIP/2.0 (36)
 Jul 10 16:31:29 DEBUG[24178]: chan_sip.c:3363 parse_request:  Header 1: Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK5a5f23ae;rport (63)
 Jul 10 16:31:29 DEBUG[24178]: chan_sip.c:3363 parse_request:  Header 2: From: "A 404 A 404" <sip:404@192.168.1.10>;tag=as2908229a (57)
 Jul 10 16:31:29 DEBUG[24178]: chan_sip.c:3363 parse_request:  Header 3: To: <sip:402@192.168.1.162> (27)
 Jul 10 16:31:29 DEBUG[24178]: chan_sip.c:3363 parse_request:  Header 4: Contact: <sip:404@192.168.1.10> (31)
 Jul 10 16:31:29 DEBUG[24178]: chan_sip.c:3363 parse_request:  Header 5: Call-ID: 06001a807b1d2e3513fbc6d866253c7e@192.168.1.10 (54)
 Jul 10 16:31:29 DEBUG[24178]: chan_sip.c:3363 parse_request:  Header 6: CSeq: 102 INVITE (16)
 Jul 10 16:31:29 DEBUG[24178]: chan_sip.c:3363 parse_request:  Header 7: User-Agent: Asterisk PBX (24)
 Jul 10 16:31:29 DEBUG[24178]: chan_sip.c:3363 parse_request:  Header 8: Max-Forwards: 70 (16)
 [Kdevcentos64*CLI> Jul 10 16:31:29 DEBUG[24178]: chan_sip.c:3363 parse_request:  Header 9: Date: Mon, 10 Jul 2006 20:31:29 GMT (35)
 Jul 10 16:31:29 DEBUG[24178]: chan_sip.c:3363 parse_request:  Header 10: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66)
 Jul 10 16:31:29 DEBUG[24178]: chan_sip.c:3363 parse_request:  Header 11: Content-Type: application/sdp (29)
 Jul 10 16:31:29 DEBUG[24178]: chan_sip.c:3363 parse_request:  Header 12: Content-Length: 263 (19)
 Jul 10 16:31:29 DEBUG[24178]: chan_sip.c:3363 parse_request:  Header 13:  (0)
 Jul 10 16:31:29 DEBUG[24178]: chan_sip.c:3395 parse_request:  Line: v=0 (3)
 Jul 10 16:31:29 DEBUG[24178]: chan_sip.c:3395 parse_request:  Line: o=root 24142 24142 IN IP4 192.168.1.10 (38)
 Jul 10 16:31:29 DEBUG[24178]: chan_sip.c:3395 parse_request:  Line: s=session (9)
 Jul 10 16:31:29 DEBUG[24178]: chan_sip.c:3395 parse_request:  Line: c=IN IP4 192.168.1.10 (21)
 Jul 10 16:31:29 DEBUG[24178]: chan_sip.c:3395 parse_request:  Line: t=0 0 (5)
 Jul 10 16:31:29 DEBUG[24178]: chan_sip.c:3395 parse_request:  Line: m=audio 12804 RTP/AVP 0 8 3 101 (31)
 Jul 10 16:31:29 DEBUG[24178]: chan_sip.c:3395 parse_request:  Line: a=rtpmap:0 PCMU/8000 (20)
 Jul 10 16:31:29 DEBUG[24178]: chan_sip.c:3395 parse_request:  Line: a=rtpmap:8 PCMA/8000 (20)
 Jul 10 16:31:29 DEBUG[24178]: chan_sip.c:3395 parse_request:  Line: a=rtpmap:3 GSM/8000 (19)
 Jul 10 16:31:29 DEBUG[24178]: chan_sip.c:3395 parse_request:  Line: a=rtpmap:101 telephone-event/8000 (33)
 Jul 10 16:31:29 DEBUG[24178]: chan_sip.c:3395 parse_request:  Line: a=fmtp:101 0-16 (15)
 Jul 10 16:31:29 DEBUG[24178]: chan_sip.c:3395 parse_request:  Line: a=silenceSupp:off - - - - (25)
 13 headers, 12 lines
 Reliably Transmitting (no NAT) to 192.168.1.162:5060:
INVITE sip:402@192.168.1.162 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK5a5f23ae;rport
From: "A 404 A 404" <sip:404@192.168.1.10>;tag=as2908229a
To: <sip:402@192.168.1.162>
Contact: <sip:404@192.168.1.10>
Call-ID: 06001a807b1d2e3513fbc6d866253c7e@192.168.1.10
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 10 Jul 2006 20:31:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 263

v=0
o=root 24142 24142 IN IP4 192.168.1.10
s=session
c=IN IP4 192.168.1.10
t=0 0
m=audio 12804 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
 Jul 10 16:31:29 DEBUG[24178]: chan_sip.c:1293 __sip_reliable_xmit:  *** SIP TIMER: Initalizing retransmit timer on packet: Id  #14
     -- Called 402
 Jul 10 16:31:29 DEBUG[24178]: channel.c:2350 set_format:  Set channel SIP/402-66ef to read format ulaw
 Jul 10 16:31:29 DEBUG[24178]: channel.c:2350 set_format:  Set channel SIP/404-56d1 to write format ulaw
 Jul 10 16:31:29 DEBUG[24178]: channel.c:2350 set_format:  Set channel SIP/404-56d1 to read format ulaw
 Jul 10 16:31:29 DEBUG[24178]: channel.c:2350 set_format:  Set channel SIP/402-66ef to write format ulaw
 [Kdevcentos64*CLI> Jul 10 16:31:29 DEBUG[24179]: app_queue.c:523 changethread:  Device 'SIP/404' changed to state '2' (In use) but we don't care because they're not a member of any queue.
 [Kdevcentos64*CLI> 
<-- SIP read from 192.168.1.162:5060: 
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK5a5f23ae;rport
From: "A 404 A 404" <sip:404@192.168.1.10>;tag=as2908229a
To: <sip:402@192.168.1.162>
Call-ID: 06001a807b1d2e3513fbc6d866253c7e@192.168.1.10
CSeq: 102 INVITE
User-Agent: Grandstream BT100 1.0.6.7
Content-Length: 0


 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 0: SIP/2.0 100 Trying (18)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 1: Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK5a5f23ae;rport (63)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 2: From: "A 404 A 404" <sip:404@192.168.1.10>;tag=as2908229a (57)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 3: To: <sip:402@192.168.1.162> (27)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 4: Call-ID: 06001a807b1d2e3513fbc6d866253c7e@192.168.1.10 (54)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 5: CSeq: 102 INVITE (16)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 6: User-Agent: Grandstream BT100 1.0.6.7 (37)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 7: Content-Length: 0 (17)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 8:  (0)
 --- (8 headers 0 lines)---
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:1445 __sip_semi_ack:  *** SIP TIMER: Cancelling retransmission #14 - INVITE (got response)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:1454 __sip_semi_ack:  (Provisional) Stopping retransmission (but retaining packet) on '06001a807b1d2e3513fbc6d866253c7e@192.168.1.10' Request 102: Found
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:9584 handle_response_invite:  SIP response 100 to standard invite
 [Kdevcentos64*CLI> 
<-- SIP read from 192.168.1.162:5060: 
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK5a5f23ae;rport
From: "A 404 A 404" <sip:404@192.168.1.10>;tag=as2908229a
To: <sip:402@192.168.1.162>;tag=e1801b68b530c818
Call-ID: 06001a807b1d2e3513fbc6d866253c7e@192.168.1.10
CSeq: 102 INVITE
User-Agent: Grandstream BT100 1.0.6.7
Content-Length: 0


 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 0: SIP/2.0 180 Ringing (19)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 1: Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK5a5f23ae;rport (63)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 2: From: "A 404 A 404" <sip:404@192.168.1.10>;tag=as2908229a (57)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 3: To: <sip:402@192.168.1.162>;tag=e1801b68b530c818 (48)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 4: Call-ID: 06001a807b1d2e3513fbc6d866253c7e@192.168.1.10 (54)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 5: CSeq: 102 INVITE (16)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 6: User-Agent: Grandstream BT100 1.0.6.7 (37)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 7: Content-Length: 0 (17)
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 8:  (0)
 --- (8 headers 0 lines)---
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:1454 __sip_semi_ack:  (Provisional) Stopping retransmission (but retaining packet) on '06001a807b1d2e3513fbc6d866253c7e@192.168.1.10' Request 102: Found
 Jul 10 16:31:29 DEBUG[24156]: chan_sip.c:9584 handle_response_invite:  SIP response 180 to standard invite
 [Kdevcentos64*CLI>     -- SIP/402-66ef is ringing
 Transmitting (no NAT) to 192.168.1.38:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.38:5060;branch=z9hG4bK-62355802;received=192.168.1.38
From: "Jerry Geis" <sip:404@192.168.1.10>;tag=48dd2341203ea66o0
To: <sip:58@192.168.1.10>;tag=as12041276
Call-ID: de93b0cc-f369dcae@192.168.1.38
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:58@192.168.1.10>
Content-Length: 0


---
 Jul 10 16:31:29 DEBUG[24146]: chan_sip.c:11668 sip_devicestate:  Checking device state for peer 402
 Jul 10 16:31:29 DEBUG[24146]: devicestate.c:187 do_state_change:  Changing state for SIP/402 - state 6 (Ringing)
 [Kdevcentos64*CLI> Jul 10 16:31:29 DEBUG[24180]: app_queue.c:523 changethread:  Device 'SIP/402' changed to state '6' (Ringing) but we don't care because they're not a member of any queue.
 [Kdevcentos64*CLI> 
<-- SIP read from 192.168.1.162:5060: 
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK5a5f23ae;rport
From: "A 404 A 404" <sip:404@192.168.1.10>;tag=as2908229a
To: <sip:402@192.168.1.162>;tag=e1801b68b530c818
Call-ID: 06001a807b1d2e3513fbc6d866253c7e@192.168.1.10
CSeq: 102 INVITE
User-Agent: Grandstream BT100 1.0.6.7
Contact: <sip:402@192.168.1.162>
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Type: application/sdp
Supported: replaces
Content-Length: 212

v=0
o=402 8000 8000 IN IP4 192.168.1.162
s=SIP Call
c=IN IP4 192.168.1.162
t=0 0
m=audio 5004 RTP/AVP 0 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11

 Jul 10 16:31:33 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 0: SIP/2.0 200 OK (14)
 Jul 10 16:31:33 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 1: Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK5a5f23ae;rport (63)
 Jul 10 16:31:33 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 2: From: "A 404 A 404" <sip:404@192.168.1.10>;tag=as2908229a (57)
 Jul 10 16:31:33 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 3: To: <sip:402@192.168.1.162>;tag=e1801b68b530c818 (48)
 Jul 10 16:31:33 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 4: Call-ID: 06001a807b1d2e3513fbc6d866253c7e@192.168.1.10 (54)
 Jul 10 16:31:33 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 5: CSeq: 102 INVITE (16)
 Jul 10 16:31:33 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 6: User-Agent: Grandstream BT100 1.0.6.7 (37)
 Jul 10 16:31:33 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 7: Contact: <sip:402@192.168.1.162> (32)
 Jul 10 16:31:33 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 8: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE (64)
 Jul 10 16:31:33 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 9: Content-Type: application/sdp (29)
 Jul 10 16:31:33 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 10: Supported: replaces (19)
 Jul 10 16:31:33 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 11: Content-Length: 212 (19)
 Jul 10 16:31:33 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 12:  (0)
 Jul 10 16:31:33 DEBUG[24156]: chan_sip.c:3395 parse_request:  Line: v=0 (3)
 Jul 10 16:31:33 DEBUG[24156]: chan_sip.c:3395 parse_request:  Line: o=402 8000 8000 IN IP4 192.168.1.162 (36)
 Jul 10 16:31:33 DEBUG[24156]: chan_sip.c:3395 parse_request:  Line: s=SIP Call (10)
 Jul 10 16:31:33 DEBUG[24156]: chan_sip.c:3395 parse_request:  Line: c=IN IP4 192.168.1.162 (22)
 Jul 10 16:31:33 DEBUG[24156]: chan_sip.c:3395 parse_request:  Line: t=0 0 (5)
 Jul 10 16:31:33 DEBUG[24156]: chan_sip.c:3395 parse_request:  Line: m=audio 5004 RTP/AVP 0 101 (26)
 Jul 10 16:31:33 DEBUG[24156]: chan_sip.c:3395 parse_request:  Line: a=sendrecv (10)
 Jul 10 16:31:33 DEBUG[24156]: chan_sip.c:3395 parse_request:  Line: a=rtpmap:0 PCMU/8000 (20)
 Jul 10 16:31:33 DEBUG[24156]: chan_sip.c:3395 parse_request:  Line: a=ptime:20 (10)
 Jul 10 16:31:33 DEBUG[24156]: chan_sip.c:3395 parse_request:  Line: a=rtpmap:101 telephone-event/8000 (33)
 Jul 10 16:31:33 DEBUG[24156]: chan_sip.c:3395 parse_request:  Line: a=fmtp:101 0-11 (15)
 --- (12 headers 11 lines)---
 Jul 10 16:31:33 DEBUG[24156]: chan_sip.c:1379 __sip_ack:  Acked pending invite 102
 Jul 10 16:31:33 DEBUG[24156]: chan_sip.c:1401 __sip_ack:  Stopping retransmission on '06001a807b1d2e3513fbc6d866253c7e@192.168.1.10' of Request 102: Match Found
 Jul 10 16:31:33 DEBUG[24156]: chan_sip.c:9584 handle_response_invite:  SIP response 200 to standard invite
 Found RTP audio format 0
 Found RTP audio format 101
 Peer audio RTP is at port 192.168.1.162:5004
 Jul 10 16:31:33 DEBUG[24156]: chan_sip.c:3604 process_sdp:  Peer audio RTP is at port 192.168.1.162:5004
 Found description format PCMU
 Found description format telephone-event
 Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
 Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
 Jul 10 16:31:33 DEBUG[24156]: chan_sip.c:6137 build_route:  build_route: Contact hop: <sip:402@192.168.1.162>
 list_route: hop: <sip:402@192.168.1.162>
 set_destination: Parsing <sip:402@192.168.1.162> for address/port to send to
 set_destination: set destination to 192.168.1.162, port 5060
 Transmitting (no NAT) to 192.168.1.162:5060:
ACK sip:402@192.168.1.162 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK1e830804;rport
From: "A 404 A 404" <sip:404@192.168.1.10>;tag=as2908229a
To: <sip:402@192.168.1.162>;tag=e1801b68b530c818
Contact: <sip:404@192.168.1.10>
Call-ID: 06001a807b1d2e3513fbc6d866253c7e@192.168.1.10
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
 [Kdevcentos64*CLI>     -- SIP/402-66ef answered SIP/404-56d1
     -- Sending DTMF '101' to the called party.
 Jul 10 16:31:33 DEBUG[24146]: chan_sip.c:11668 sip_devicestate:  Checking device state for peer 402
 Jul 10 16:31:33 DEBUG[24146]: devicestate.c:187 do_state_change:  Changing state for SIP/402 - state 2 (In use)
 [Kdevcentos64*CLI> Jul 10 16:31:33 DEBUG[24182]: app_queue.c:523 changethread:  Device 'SIP/402' changed to state '2' (In use) but we don't care because they're not a member of any queue.
 [Kdevcentos64*CLI>     -- Sending DTMF '101' to the calling party.
 [Kdevcentos64*CLI> Jul 10 16:31:35 DEBUG[24178]: channel.c:2350 set_format:  Set channel SIP/404-56d1 to read format ulaw
 Jul 10 16:31:35 DEBUG[24178]: channel.c:2350 set_format:  Set channel SIP/402-66ef to write format ulaw
 Jul 10 16:31:35 DEBUG[24178]: channel.c:2350 set_format:  Set channel SIP/402-66ef to read format ulaw
 Jul 10 16:31:35 DEBUG[24178]: channel.c:2350 set_format:  Set channel SIP/404-56d1 to write format ulaw
 Jul 10 16:31:35 DEBUG[24178]: chan_sip.c:2540 sip_answer:  sip_answer(SIP/404-56d1)
 We're at 192.168.1.10 port 11192
 Adding codec 0x4 (ulaw) to SDP
 Adding codec 0x8 (alaw) to SDP
 Adding non-codec 0x1 (telephone-event) to SDP
 Reliably Transmitting (no NAT) to 192.168.1.38:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.38:5060;branch=z9hG4bK-62355802;received=192.168.1.38
From: "Jerry Geis" <sip:404@192.168.1.10>;tag=48dd2341203ea66o0
To: <sip:58@192.168.1.10>;tag=as12041276
Call-ID: de93b0cc-f369dcae@192.168.1.38
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:58@192.168.1.10>
Content-Type: application/sdp
Content-Length: 240

v=0
o=root 24142 24142 IN IP4 192.168.1.10
s=session
c=IN IP4 192.168.1.10
t=0 0
m=audio 11192 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
 Jul 10 16:31:35 DEBUG[24178]: chan_sip.c:1293 __sip_reliable_xmit:  *** SIP TIMER: Initalizing retransmit timer on packet: Id  #15
     -- Attempting native bridge of SIP/404-56d1 and SIP/402-66ef
 [Kdevcentos64*CLI> Jul 10 16:31:35 DEBUG[24146]: chan_sip.c:11668 sip_devicestate:  Checking device state for peer 404
 Jul 10 16:31:35 DEBUG[24146]: devicestate.c:187 do_state_change:  Changing state for SIP/404 - state 2 (In use)
 [Kdevcentos64*CLI> Jul 10 16:31:35 DEBUG[24183]: app_queue.c:523 changethread:  Device 'SIP/404' changed to state '2' (In use) but we don't care because they're not a member of any queue.
 [Kdevcentos64*CLI> 
<-- SIP read from 192.168.1.38:5060: 
ACK sip:58@192.168.1.10 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.38:5060;branch=z9hG4bK-9c1d4b98
From: "Jerry Geis" <sip:404@192.168.1.10>;tag=48dd2341203ea66o0
To: <sip:58@192.168.1.10>;tag=as12041276
Call-ID: de93b0cc-f369dcae@192.168.1.38
CSeq: 102 ACK
Max-Forwards: 70
Proxy-Authorization: Digest username="404",realm="asterisk",nonce="360d1285",uri="sip:58@192.168.1.10",algorithm=MD5,response="a50eaab8fd9ad73edb8e7b26fb924d4a"
Contact: "Jerry Geis" <sip:404@192.168.1.38:5060>
User-Agent: Linksys/SPA942-4.1.12
Proxy-Require: 192.168.1.10
Content-Length: 0


 Jul 10 16:31:35 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 0: ACK sip:58@192.168.1.10 SIP/2.0 (31)
 Jul 10 16:31:35 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 1: Via: SIP/2.0/UDP 192.168.1.38:5060;branch=z9hG4bK-9c1d4b98 (58)
 Jul 10 16:31:35 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 2: From: "Jerry Geis" <sip:404@192.168.1.10>;tag=48dd2341203ea66o0 (63)
 Jul 10 16:31:35 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 3: To: <sip:58@192.168.1.10>;tag=as12041276 (40)
 Jul 10 16:31:35 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 4: Call-ID: de93b0cc-f369dcae@192.168.1.38 (39)
 Jul 10 16:31:35 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 5: CSeq: 102 ACK (13)
 Jul 10 16:31:35 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 6: Max-Forwards: 70 (16)
 Jul 10 16:31:35 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 7: Proxy-Authorization: Digest username="404",realm="asterisk",nonce="360d1285",uri="sip:58@192.168.1.10",algorithm=MD5,response="a50eaab8fd9ad73edb8e7b26fb924d4a" (160)
 Jul 10 16:31:35 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 8: Contact: "Jerry Geis" <sip:404@192.168.1.38:5060> (49)
 Jul 10 16:31:35 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 9: User-Agent: Linksys/SPA942-4.1.12 (33)
 Jul 10 16:31:35 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 10: Proxy-Require: 192.168.1.10 (27)
 Jul 10 16:31:35 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 11: Content-Length: 0 (17)
 Jul 10 16:31:35 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 12:  (0)
 --- (12 headers 0 lines)---
 Jul 10 16:31:35 DEBUG[24156]: chan_sip.c:11137 handle_request:  **** Received ACK (6) - Command in SIP ACK
 Jul 10 16:31:35 DEBUG[24156]: chan_sip.c:1390 __sip_ack:  ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #15
 Jul 10 16:31:35 DEBUG[24156]: chan_sip.c:1401 __sip_ack:  Stopping retransmission on 'de93b0cc-f369dcae@192.168.1.38' of Response 102: Match Found
 [Kdevcentos64*CLI> Jul 10 16:31:35 DEBUG[24178]: rtp.c:1352 ast_rtp_write:  Ooh, format changed from unknown to ulaw
 [Kdevcentos64*CLI> Jul 10 16:31:35 DEBUG[24178]: rtp.c:1352 ast_rtp_write:  Ooh, format changed from unknown to ulaw
 [Kdevcentos64*CLI> 
<-- SIP read from 192.168.1.162:5060: 
BYE sip:404@192.168.1.10 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.162;branch=z9hG4bKb0b3c7fa4f4e6885
From: <sip:402@192.168.1.162>;tag=e1801b68b530c818
To: "A 404 A 404" <sip:404@192.168.1.10>;tag=as2908229a
Call-ID: 06001a807b1d2e3513fbc6d866253c7e@192.168.1.10
CSeq: 1627 BYE
User-Agent: Grandstream BT100 1.0.6.7
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE
Content-Length: 0


 Jul 10 16:31:39 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 0: BYE sip:404@192.168.1.10 SIP/2.0 (32)
 Jul 10 16:31:39 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 1: Via: SIP/2.0/UDP 192.168.1.162;branch=z9hG4bKb0b3c7fa4f4e6885 (61)
 Jul 10 16:31:39 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 2: From: <sip:402@192.168.1.162>;tag=e1801b68b530c818 (50)
 Jul 10 16:31:39 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 3: To: "A 404 A 404" <sip:404@192.168.1.10>;tag=as2908229a (55)
 Jul 10 16:31:39 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 4: Call-ID: 06001a807b1d2e3513fbc6d866253c7e@192.168.1.10 (54)
 Jul 10 16:31:39 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 5: CSeq: 1627 BYE (14)
 Jul 10 16:31:39 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 6: User-Agent: Grandstream BT100 1.0.6.7 (37)
 Jul 10 16:31:39 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 7: Max-Forwards: 70 (16)
 Jul 10 16:31:39 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 8: Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE (64)
 Jul 10 16:31:39 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 9: Content-Length: 0 (17)
 Jul 10 16:31:39 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 10:  (0)
 --- (10 headers 0 lines)---
 Jul 10 16:31:39 DEBUG[24156]: chan_sip.c:11137 handle_request:  **** Received BYE (8) - Command in SIP BYE
 Sending to 192.168.1.162 : 5060 (non-NAT)
 Transmitting (no NAT) to 192.168.1.162:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.162;branch=z9hG4bKb0b3c7fa4f4e6885;received=192.168.1.162
From: <sip:402@192.168.1.162>;tag=e1801b68b530c818
To: "A 404 A 404" <sip:404@192.168.1.10>;tag=as2908229a
Call-ID: 06001a807b1d2e3513fbc6d866253c7e@192.168.1.10
CSeq: 1627 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:404@192.168.1.10>
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing


---
 Jul 10 16:31:39 DEBUG[24178]: channel.c:3275 ast_generic_bridge:  Didn't get a frame from channel: SIP/402-66ef
 Jul 10 16:31:39 DEBUG[24178]: channel.c:3550 ast_channel_bridge:  Bridge stops bridging channels SIP/404-56d1 and SIP/402-66ef
 Jul 10 16:31:39 DEBUG[24178]: channel.c:1323 ast_hangup:  Hanging up channel 'SIP/402-66ef'
 Jul 10 16:31:39 DEBUG[24178]: chan_sip.c:2418 sip_hangup:  Hangup call SIP/402-66ef, SIP callid 06001a807b1d2e3513fbc6d866253c7e@192.168.1.10)
 Jul 10 16:31:39 DEBUG[24178]: chan_sip.c:2426 sip_hangup:  update_call_counter(402) - decrement call limit counter
 Jul 10 16:31:39 DEBUG[24178]: chan_sip.c:2209 update_call_counter:  Updating call counter for outgoing call
 Jul 10 16:31:39 DEBUG[24178]: app_dial.c:1619 dial_exec_full:  Exiting with DIALSTATUS=ANSWER.
 Jul 10 16:31:39 DEBUG[24178]: pbx.c:2316 __ast_pbx_run:  Spawn extension (smvoice-sip,58,1) exited non-zero on 'SIP/404-56d1'
 Jul 10 16:31:39 DEBUG[24178]: pbx.c:1522 pbx_substitute_variables_helper_full:  Function result is '"A 404 A 404" <404>'
 Jul 10 16:31:39 DEBUG[24178]: pbx.c:1522 pbx_substitute_variables_helper_full:  Function result is '404'
 Jul 10 16:31:39 DEBUG[24178]: pbx.c:1522 pbx_substitute_variables_helper_full:  Function result is '58'
 Jul 10 16:31:39 DEBUG[24178]: pbx.c:1522 pbx_substitute_variables_helper_full:  Function result is 'smvoice-sip'
 Jul 10 16:31:39 DEBUG[24178]: pbx.c:1522 pbx_substitute_variables_helper_full:  Function result is 'SIP/404-56d1'
 Jul 10 16:31:39 DEBUG[24178]: pbx.c:1522 pbx_substitute_variables_helper_full:  Function result is 'SIP/402-66ef'
 Jul 10 16:31:39 DEBUG[24178]: pbx.c:1522 pbx_substitute_variables_helper_full:  Function result is 'Dial'
 Jul 10 16:31:39 DEBUG[24178]: pbx.c:1522 pbx_substitute_variables_helper_full:  Function result is 'SIP/402|20|tTD(101:101)'
 Jul 10 16:31:39 DEBUG[24178]: pbx.c:1522 pbx_substitute_variables_helper_full:  Function result is '2006-07-10 16:31:29'
 Jul 10 16:31:39 DEBUG[24178]: pbx.c:1522 pbx_substitute_variables_helper_full:  Function result is '2006-07-10 16:31:35'
 Jul 10 16:31:39 DEBUG[24178]: pbx.c:1522 pbx_substitute_variables_helper_full:  Function result is '2006-07-10 16:31:39'
 Jul 10 16:31:39 DEBUG[24178]: pbx.c:1522 pbx_substitute_variables_helper_full:  Function result is '10'
 Jul 10 16:31:39 DEBUG[24178]: pbx.c:1522 pbx_substitute_variables_helper_full:  Function result is '4'
 Jul 10 16:31:39 DEBUG[24178]: pbx.c:1522 pbx_substitute_variables_helper_full:  Function result is 'ANSWERED'
 Jul 10 16:31:39 DEBUG[24178]: pbx.c:1522 pbx_substitute_variables_helper_full:  Function result is 'DOCUMENTATION'
 Jul 10 16:31:39 DEBUG[24178]: pbx.c:1522 pbx_substitute_variables_helper_full:  Function result is '(null)'
 Jul 10 16:31:39 DEBUG[24178]: pbx.c:1522 pbx_substitute_variables_helper_full:  Function result is '1152563489.2'
 Jul 10 16:31:39 DEBUG[24178]: pbx.c:1522 pbx_substitute_variables_helper_full:  Function result is '(null)'
 Jul 10 16:31:39 DEBUG[24178]: channel.c:1323 ast_hangup:  Hanging up channel 'SIP/404-56d1'
 Jul 10 16:31:39 DEBUG[24178]: chan_sip.c:2418 sip_hangup:  Hangup call SIP/404-56d1, SIP callid de93b0cc-f369dcae@192.168.1.38)
 Jul 10 16:31:39 DEBUG[24178]: chan_sip.c:2426 sip_hangup:  update_call_counter(404) - decrement call limit counter
 Jul 10 16:31:39 DEBUG[24178]: chan_sip.c:2209 update_call_counter:  Updating call counter for incoming call
 set_destination: Parsing <sip:404@192.168.1.38:5060> for address/port to send to
 set_destination: set destination to 192.168.1.38, port 5060
 Reliably Transmitting (no NAT) to 192.168.1.38:5060:
BYE sip:404@192.168.1.38:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK6c2073e2;rport
From: <sip:58@192.168.1.10>;tag=as12041276
To: "Jerry Geis" <sip:404@192.168.1.10>;tag=48dd2341203ea66o0
Contact: <sip:58@192.168.1.10>
Call-ID: de93b0cc-f369dcae@192.168.1.38
CSeq: 102 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
 Jul 10 16:31:39 DEBUG[24178]: chan_sip.c:1293 __sip_reliable_xmit:  *** SIP TIMER: Initalizing retransmit timer on packet: Id  #16
 Jul 10 16:31:39 DEBUG[24146]: chan_sip.c:11668 sip_devicestate:  Checking device state for peer 402
 Jul 10 16:31:39 DEBUG[24146]: devicestate.c:187 do_state_change:  Changing state for SIP/402 - state 1 (Not in use)
 Jul 10 16:31:39 DEBUG[24146]: chan_sip.c:11668 sip_devicestate:  Checking device state for peer 404
 [Kdevcentos64*CLI> Jul 10 16:31:39 DEBUG[24146]: devicestate.c:187 do_state_change:  Changing state for SIP/404 - state 1 (Not in use)
 Jul 10 16:31:39 DEBUG[24184]: app_queue.c:523 changethread:  Device 'SIP/402' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
 Jul 10 16:31:39 DEBUG[24185]: app_queue.c:523 changethread:  Device 'SIP/404' changed to state '1' (Not in use) but we don't care because they're not a member of any queue.
 [Kdevcentos64*CLI> 
<-- SIP read from 192.168.1.38:5060: 
SIP/2.0 200 OK
To: "Jerry Geis" <sip:404@192.168.1.10>;tag=48dd2341203ea66o0
From: <sip:58@192.168.1.10>;tag=as12041276
Call-ID: de93b0cc-f369dcae@192.168.1.38
CSeq: 102 BYE
Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK6c2073e2
Server: 192.168.1.10
Content-Length: 0


 Jul 10 16:31:39 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 0: SIP/2.0 200 OK (14)
 Jul 10 16:31:39 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 1: To: "Jerry Geis" <sip:404@192.168.1.10>;tag=48dd2341203ea66o0 (61)
 Jul 10 16:31:39 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 2: From: <sip:58@192.168.1.10>;tag=as12041276 (42)
 Jul 10 16:31:39 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 3: Call-ID: de93b0cc-f369dcae@192.168.1.38 (39)
 Jul 10 16:31:39 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 4: CSeq: 102 BYE (13)
 Jul 10 16:31:39 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 5: Via: SIP/2.0/UDP 192.168.1.10:5060;branch=z9hG4bK6c2073e2 (57)
 [Kdevcentos64*CLI> Jul 10 16:31:39 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 6: Server: 192.168.1.10 (20)
 Jul 10 16:31:39 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 7: Content-Length: 0 (17)
 Jul 10 16:31:39 DEBUG[24156]: chan_sip.c:3363 parse_request:  Header 8:  (0)
 --- (8 headers 0 lines)---
 Jul 10 16:31:39 DEBUG[24156]: chan_sip.c:1390 __sip_ack:  ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #16
 Jul 10 16:31:39 DEBUG[24156]: chan_sip.c:1401 __sip_ack:  Stopping retransmission on 'de93b0cc-f369dcae@192.168.1.38' of Request 102: Match Found
 Destroying call '06001a807b1d2e3513fbc6d866253c7e@192.168.1.10'
 Destroying call 'de93b0cc-f369dcae@192.168.1.38'
 [Kdevcentos64*CLI>     -- Remote UNIX connection
 [Kdevcentos64*CLI> Executing last minute cleanups
 [Kdevcentos64*CLI>   == Destroying musiconhold processes
 [Kdevcentos64*CLI> Jul 10 16:31:51 DEBUG[24188]: res_musiconhold.c:1087 ast_moh_destroy:  killing 24148!
 [Kdevcentos64*CLI> Executing last minute cleanups
Asterisk cleanly ending (0).
[0m