
<--- SIP read from 192.168.0.61:5060 --->
INVITE sip:6000@asterisk.abc.dom:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.61;branch=z9hG4bK5dfe9fc875C4C893
From: "6001" <sip:6001@asterisk.abc.dom>;tag=15A5FE72-EE07032F
To: <sip:6000@asterisk.abc.dom;user=phone>
CSeq: 1 INVITE
Call-ID: 962b51f6-9936784-c0ef42b1@192.168.0.61
Contact: <sip:6001@192.168.0.61>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.3.0067
Supported: 100rel,replace
Allow-Events: talk,hold,conference
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 247

v=0
o=- 978308461 978308461 IN IP4 192.168.0.61
s=Polycom IP Phone
c=IN IP4 192.168.0.61
t=0 0
a=sendrecv
m=audio 2222 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000

<------------->
 --- (14 headers 11 lines) ---
 Sending to 192.168.0.61 : 5060 (NAT)
 Using INVITE request as basis request - 962b51f6-9936784-c0ef42b1@192.168.0.61
 Found peer '6003'
 
<--- Reliably Transmitting (NAT) to 192.168.0.61:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.0.61;branch=z9hG4bK5dfe9fc875C4C893;received=192.168.0.61
From: "6001" <sip:6001@asterisk.abc.dom>;tag=15A5FE72-EE07032F
To: <sip:6000@asterisk.abc.dom;user=phone>;tag=as443c2479
Call-ID: 962b51f6-9936784-c0ef42b1@192.168.0.61
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk.abc.dom", nonce="1a3a77d3"
Content-Length: 0


<------------>
 Scheduling destruction of SIP dialog '962b51f6-9936784-c0ef42b1@192.168.0.61' in 32000 ms (Method: INVITE)
 [Kasterisk*CLI> 
<--- SIP read from 192.168.0.61:5060 --->
ACK sip:6000@asterisk.abc.dom:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.61;branch=z9hG4bK5dfe9fc875C4C893
From: "6001" <sip:6001@asterisk.abc.dom>;tag=15A5FE72-EE07032F
To: <sip:6000@asterisk.abc.dom;user=phone>;tag=as443c2479
CSeq: 1 ACK
Call-ID: 962b51f6-9936784-c0ef42b1@192.168.0.61
Contact: <sip:6001@192.168.0.61>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.3.0067
Max-Forwards: 70
Content-Length: 0


<------------->
 --- (11 headers 0 lines) ---
 [Kasterisk*CLI> 
<--- SIP read from 192.168.0.61:5060 --->
INVITE sip:6000@asterisk.abc.dom:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.61;branch=z9hG4bK657e980d892894C0
From: "6001" <sip:6001@asterisk.abc.dom>;tag=15A5FE72-EE07032F
To: <sip:6000@asterisk.abc.dom;user=phone>
CSeq: 2 INVITE
Call-ID: 962b51f6-9936784-c0ef42b1@192.168.0.61
Contact: <sip:6001@192.168.0.61>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.3.0067
Supported: 100rel,replace
Allow-Events: talk,hold,conference
Proxy-Authorization: Digest username="6001", realm="asterisk.abc.dom", nonce="1a3a77d3", uri="sip:6000@asterisk.abc.dom:5060;user=phone", response="212a9ea03e1fbda251b3c0df95f5f338", algorithm=MD5
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 247

v=0
o=- 978308461 978308461 IN IP4 192.168.0.61
s=Polycom IP Phone
c=IN IP4 192.168.0.61
t=0 0
a=sendrecv
m=audio 2222 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000

<------------->
 --- (15 headers 11 lines) ---
 Sending to 192.168.0.61 : 5060 (NAT)
 Using INVITE request as basis request - 962b51f6-9936784-c0ef42b1@192.168.0.61
 Found peer '6003'
 [Feb 13 22:34:35] [1;31;40mWARNING[0;37;40m[6780]: [1;37;40mchan_sip.c[0;37;40m:[1;37;40m8023[0;37;40m [1;37;40mcheck_auth[0;37;40m:  username mismatch, have <6003>, digest has <6001>
 [Feb 13 22:34:35] [1;33;40mNOTICE[0;37;40m[6780]: [1;37;40mchan_sip.c[0;37;40m:[1;37;40m13200[0;37;40m [1;37;40mhandle_request_invite[0;37;40m:  Failed to authenticate user "6001" <sip:6001@asterisk.abc.dom>;tag=15A5FE72-EE07032F
 
<--- Reliably Transmitting (NAT) to 192.168.0.61:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.0.61;branch=z9hG4bK657e980d892894C0;received=192.168.0.61
From: "6001" <sip:6001@asterisk.abc.dom>;tag=15A5FE72-EE07032F
To: <sip:6000@asterisk.abc.dom;user=phone>;tag=as443c2479
Call-ID: 962b51f6-9936784-c0ef42b1@192.168.0.61
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


<------------>
 Scheduling destruction of SIP dialog '962b51f6-9936784-c0ef42b1@192.168.0.61' in 32000 ms (Method: INVITE)
 [Kasterisk*CLI> 
<--- SIP read from 192.168.0.61:5060 --->
ACK sip:6000@asterisk.abc.dom:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.61;branch=z9hG4bK657e980d892894C0
From: "6001" <sip:6001@asterisk.abc.dom>;tag=15A5FE72-EE07032F
To: <sip:6000@asterisk.abc.dom;user=phone>;tag=as443c2479
CSeq: 2 ACK
Call-ID: 962b51f6-9936784-c0ef42b1@192.168.0.61
Contact: <sip:6001@192.168.0.61>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.3.0067
Max-Forwards: 70
Content-Length: 0





+++++++++++++++++++++++++++++++++++++ Call from 6003 (on same phone that works) ++++++++++++++++++++++++++++=



<------------->
 --- (11 headers 0 lines) ---
 [Kasterisk*CLI> 
<--- SIP read from 192.168.0.61:5060 --->
INVITE sip:6000@asterisk.abc.dom:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.61;branch=z9hG4bK765cdfeaA0646F45
From: "6003" <sip:6003@asterisk.abc.dom>;tag=5EA6DD74-E948D121
To: <sip:6000@asterisk.abc.dom;user=phone>
CSeq: 1 INVITE
Call-ID: 655fdfb8-5047f4e6-5aeb9c03@192.168.0.61
Contact: <sip:6003@192.168.0.61>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.3.0067
Supported: 100rel,replace
Allow-Events: talk,hold,conference
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 247

v=0
o=- 978308469 978308469 IN IP4 192.168.0.61
s=Polycom IP Phone
c=IN IP4 192.168.0.61
t=0 0
a=sendrecv
m=audio 2224 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000

<------------->
 --- (14 headers 11 lines) ---
 Sending to 192.168.0.61 : 5060 (NAT)
 Using INVITE request as basis request - 655fdfb8-5047f4e6-5aeb9c03@192.168.0.61
 Found peer '6003'
 
<--- Reliably Transmitting (NAT) to 192.168.0.61:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.0.61;branch=z9hG4bK765cdfeaA0646F45;received=192.168.0.61
From: "6003" <sip:6003@asterisk.abc.dom>;tag=5EA6DD74-E948D121
To: <sip:6000@asterisk.abc.dom;user=phone>;tag=as46c6eb05
Call-ID: 655fdfb8-5047f4e6-5aeb9c03@192.168.0.61
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk.abc.dom", nonce="19965183"
Content-Length: 0


<------------>
 Scheduling destruction of SIP dialog '655fdfb8-5047f4e6-5aeb9c03@192.168.0.61' in 32000 ms (Method: INVITE)
 [Kasterisk*CLI> 
<--- SIP read from 192.168.0.61:5060 --->
ACK sip:6000@asterisk.abc.dom:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.61;branch=z9hG4bK765cdfeaA0646F45
From: "6003" <sip:6003@asterisk.abc.dom>;tag=5EA6DD74-E948D121
To: <sip:6000@asterisk.abc.dom;user=phone>;tag=as46c6eb05
CSeq: 1 ACK
Call-ID: 655fdfb8-5047f4e6-5aeb9c03@192.168.0.61
Contact: <sip:6003@192.168.0.61>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.3.0067
Max-Forwards: 70
Content-Length: 0


<------------->
 --- (11 headers 0 lines) ---
 [Kasterisk*CLI> 
<--- SIP read from 192.168.0.61:5060 --->
INVITE sip:6000@asterisk.abc.dom:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.61;branch=z9hG4bKc0de3c9fED26B762
From: "6003" <sip:6003@asterisk.abc.dom>;tag=5EA6DD74-E948D121
To: <sip:6000@asterisk.abc.dom;user=phone>
CSeq: 2 INVITE
Call-ID: 655fdfb8-5047f4e6-5aeb9c03@192.168.0.61
Contact: <sip:6003@192.168.0.61>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.3.0067
Supported: 100rel,replace
Allow-Events: talk,hold,conference
Proxy-Authorization: Digest username="6003", realm="asterisk.abc.dom", nonce="19965183", uri="sip:6000@asterisk.abc.dom:5060;user=phone", response="86b2bda4d1f48ede8f2bd5be27cef720", algorithm=MD5
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 247

v=0
o=- 978308469 978308469 IN IP4 192.168.0.61
s=Polycom IP Phone
c=IN IP4 192.168.0.61
t=0 0
a=sendrecv
m=audio 2224 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000

<------------->
 --- (15 headers 11 lines) ---
 Sending to 192.168.0.61 : 5060 (NAT)
 Using INVITE request as basis request - 655fdfb8-5047f4e6-5aeb9c03@192.168.0.61
 Found peer '6003'
 Found RTP audio format 0
 Found RTP audio format 8
 Found RTP audio format 18
 Found RTP audio format 101
 Peer audio RTP is at port 192.168.0.61:2224
 Found description format PCMU for ID 0
 Found description format PCMA for ID 8
 Found description format G729 for ID 18
 Found description format telephone-event for ID 101
 Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x104 (ulaw|g729)
 Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
 Peer audio RTP is at port 192.168.0.61:2224
 Looking for 6000 in numberplan-custom-1 (domain asterisk.abc.dom)
 list_route: hop: <sip:6003@192.168.0.61>
 
<--- Transmitting (NAT) to 192.168.0.61:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.61;branch=z9hG4bKc0de3c9fED26B762;received=192.168.0.61
From: "6003" <sip:6003@asterisk.abc.dom>;tag=5EA6DD74-E948D121
To: <sip:6000@asterisk.abc.dom;user=phone>
Call-ID: 655fdfb8-5047f4e6-5aeb9c03@192.168.0.61
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:6000@192.168.0.200>
Content-Length: 0


<------------>
 [Kasterisk*CLI>     -- Executing [6000@numberplan-custom-1:1] [1;36;40mMacro[0;37;40m("[1;35;40mSIP/6003-082326d0[0;37;40m", "[1;35;40mstdexten|6000|Zap/1[0;37;40m") in new stack
 [Kasterisk*CLI>     -- Executing [s@macro-stdexten:1] [1;36;40mDial[0;37;40m("[1;35;40mSIP/6003-082326d0[0;37;40m", "[1;35;40mZap/1|20[0;37;40m") in new stack
 [Kasterisk*CLI>     -- Called 1
 [Kasterisk*CLI>     -- Zap/1-1 is ringing
 [Kasterisk*CLI> 
<--- Transmitting (NAT) to 192.168.0.61:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.61;branch=z9hG4bKc0de3c9fED26B762;received=192.168.0.61
From: "6003" <sip:6003@asterisk.abc.dom>;tag=5EA6DD74-E948D121
To: <sip:6000@asterisk.abc.dom;user=phone>;tag=as41a523cd
Call-ID: 655fdfb8-5047f4e6-5aeb9c03@192.168.0.61
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:6000@192.168.0.200>
Content-Length: 0


<------------>
 [Kasterisk*CLI> 
<--- SIP read from 192.168.0.61:5060 --->
CANCEL sip:6000@asterisk.abc.dom:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.0.61;branch=z9hG4bKc0de3c9fED26B762
From: "6003" <sip:6003@asterisk.abc.dom>;tag=5EA6DD74-E948D121
To: <sip:6000@asterisk.abc.dom;user=phone>
CSeq: 2 CANCEL
Call-ID: 655fdfb8-5047f4e6-5aeb9c03@192.168.0.61
Contact: <sip:6003@192.168.0.61>
User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.3.0067
Proxy-Authorization: Digest username="6003", realm="asterisk.abc.dom", nonce="19965183", uri="sip:6000@asterisk.abc.dom:5060;user=phone", response="86b2bda4d1f48ede8f2bd5be27cef720", algorithm=MD5
Max-Forwards: 70
Content-Length: 0


<------------->
 --- (11 headers 0 lines) ---
 Sending to 192.168.0.61 : 5060 (NAT)
 
<--- Reliably Transmitting (NAT) to 192.168.0.61:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.0.61;branch=z9hG4bKc0de3c9fED26B762;received=192.168.0.61
From: "6003" <sip:6003@asterisk.abc.dom>;tag=5EA6DD74-E948D121
To: <sip:6000@asterisk.abc.dom;user=phone>;tag=as41a523cd
Call-ID: 655fdfb8-5047f4e6-5aeb9c03@192.168.0.61
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0


<------------>
 
<--- Transmitting (NAT) to 192.168.0.61:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.61;branch=z9hG4bKc0de3c9fED26B762;received=192.168.0.61
From: "6003" <sip:6003@asterisk.abc.dom>;tag=5EA6DD74-E948D121
To: <sip:6000@asterisk.abc.dom;user=phone>;tag=as41a523cd
Call-ID: 655fdfb8-5047f4e6-5aeb9c03@192.168.0.61
CSeq: 2 CANCEL
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:6000@192.168.0.200>
Content-Length: 0


<------------>
 [Kasterisk*CLI>     -- Hungup 'Zap/1-1'
   == Spawn extension (macro-stdexten, s, 1) exited non-zero on 'SIP/6003-082326d0' in macro 'stdexten'
   == Spawn extension (macro-stdexten, s, 1) exited non-zero on 'SIP/6003-082326d0'
 [Kasterisk*CLI> 
<--- SIP read from 192.168.0.61:5060 --->
ACK sip:6000@192.168.0.200 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.61;branch=z9hG4bKc0de3c9fED26B762
From: "6003" <sip:6003@asterisk.abc.dom>;tag=5EA6DD74-E948D121
To: <sip:6000@asterisk.abc.dom;user=phone>;tag=as41a523cd
CSeq: 2 ACK
Call-ID: 655fdfb8-5047f4e6-5aeb9c03@192.168.0.61
Contact: <sip:6003@192.168.0.61>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_601-UA/1.6.3.0067
Max-Forwards: 70
Content-Length: 0


<------------->
 --- (11 headers 0 lines) ---
 Really destroying SIP dialog '655fdfb8-5047f4e6-5aeb9c03@192.168.0.61' Method: ACK
 [Kasterisk*CLI>