<------------->
--- (8 headers 0 lines) ---
ares*CLI>
<--- SIP read from UDP://217.64.48.36:1162 --->
INVITE sip:0143620918@217.64.49.43;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 217.64.48.36:5060;branch=z9hG4bK-d8754z-9aa41781bdf61ce6-1---d8754z-
Max-Forwards: 70
Contact: <sip:gaia@217.64.48.36:5060;transport=UDP>
To: <sip:0143620918@217.64.49.43;transport=UDP>
From: "gaia"<sip:gaia@217.64.49.43;transport=UDP>;tag=411f2d60
Call-ID: Mjk3YWVhNDM2NGFmZjFjZTNhZmMxNzlhMmQ5OWNjOTQ.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
User-Agent: Zoiper rev.4829
Authorization: Digest username="gaia",realm="asterisk",nonce="66c0d26b",uri="sip:0143620918@217.64.49.43;transport=UDP",response="6aeed9d9c56f5311f2d023a970ad9977",algorithm=MD5
Content-Length: 336

v=0
o=Zoiper_pc-francois 0 0 IN IP4 192.168.1.100
s=Zoiper_session
c=IN IP4 192.168.1.100
t=0 0
m=audio 8000 RTP/AVP 3 0 8 110 98 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:110 speex/8000
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

<------------->
--- (13 headers 15 lines) ---
Sending to 217.64.48.36 : 1162 (NAT)
Using INVITE request as basis request - Mjk3YWVhNDM2NGFmZjFjZTNhZmMxNzlhMmQ5OWNjOTQ.
Found user 'gaia' for 'gaia'
Found RTP audio format 3
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 110
Found RTP audio format 98
Found RTP audio format 101
Peer audio RTP is at port 192.168.1.100:8000
Found audio description format GSM for ID 3
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format speex for ID 110
Found audio description format iLBC for ID 98
Got unsupported a:fmtp in SDP offer
Found audio description format telephone-event for ID 101
Got unsupported a:fmtp in SDP offer
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x60e (gsm|ulaw|alaw|speex|ilbc)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.1.100:8000
Looking for 0143620918 in t38 (domain 217.64.49.43)
list_route: hop: <sip:gaia@217.64.48.36:5060;transport=UDP>
ares*CLI>
<--- Transmitting (NAT) to 217.64.48.36:1162 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 217.64.48.36:5060;branch=z9hG4bK-d8754z-9aa41781bdf61ce6-1---d8754z-;received=217.64.48.36
From: "gaia"<sip:gaia@217.64.49.43;transport=UDP>;tag=411f2d60
To: <sip:0143620918@217.64.49.43;transport=UDP>
Call-ID: Mjk3YWVhNDM2NGFmZjFjZTNhZmMxNzlhMmQ5OWNjOTQ.
CSeq: 2 INVITE
User-Agent: Asterisk PBX 1.6.0.14-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:0143620918@217.64.49.43>
Content-Length: 0


<------------>
    -- Executing [0143620918@t38:1] SIPAddHeader("SIP/gaia-0824b708", "x-acro-client: Fcois") in new stack
    -- Executing [0143620918@t38:2] SIPAddHeader("SIP/gaia-0824b708", "x-acro-credit: 0.00000") in new stack
    -- Executing [0143620918@t38:3] SIPAddHeader("SIP/gaia-0824b708", "x-acro-called: france_fixe") in new stack
    -- Executing [0143620918@t38:4] SIPAddHeader("SIP/gaia-0824b708", "x-acro-caller: france") in new stack
    -- Executing [0143620918@t38:5] SIPAddHeader("SIP/gaia-0824b708", "x-acro-masquage: 1") in new stack
    -- Executing [0143620918@t38:6] SIPAddHeader("SIP/gaia-0824b708", "x-acro-codec: t38") in new stack
    -- Executing [0143620918@t38:7] SIPAddHeader("SIP/gaia-0824b708", "x-acro-dst_ext: t38") in new stack
    -- Executing [0143620918@t38:8] SIPAddHeader("SIP/gaia-0824b708", "x-acro-nbappel: 946") in new stack
    -- Executing [0143620918@t38:9] SIPAddHeader("SIP/gaia-0824b708", "x-acro-t38: oui") in new stack
    -- Executing [0143620918@t38:10] Dial("SIP/gaia-0824b708", "SIP/ser_sei0/0143620918") in new stack
  == Using SIP RTP CoS mark 5
  == Using UDPTL CoS mark 5
Audio is at 192.168.7.40 port 11220
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.7.36:5060:
INVITE sip:0143620918@192.168.7.36 SIP/2.0
Via: SIP/2.0/UDP 192.168.7.40:5060;branch=z9hG4bK15b9df72;rport
Max-Forwards: 70
From: "gaia" <sip:gaia@192.168.7.40>;tag=as318313ba
To: <sip:0143620918@192.168.7.36>
Contact: <sip:gaia@192.168.7.40>
Call-ID: 2177e0776ef133741b8423691ce29b5f@192.168.7.40
Seq: 102 INVITE
User-Agent: Asterisk PBX 1.6.0.14-rc1
Date: Fri, 28 Aug 2009 06:58:11 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
x-acro-t38: oui
x-acro-nbappel: 946
x-acro-dst_ext: t38
x-acro-codec: t38
x-acro-masquage: 1
x-acro-caller: france
x-acro-called: france_fixe
x-acro-credit: 0.00000
x-acro-client: Fcois
Content-Type: application/sdp
Content-Length: 292

v=0
o=root 1820870008 1820870008 IN IP4 192.168.7.40
s=Asterisk PBX 1.6.0.14-rc1
c=IN IP4 192.168.7.40
t=0 0
m=audio 11220 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=sendrecv

---
    -- Called ser_sei0/0143620918
ares*CLI>
<--- SIP read from UDP://192.168.7.36:5060 --->
SIP/2.0 100 Giving a try
Via: SIP/2.0/UDP 192.168.7.40:5060;branch=z9hG4bK15b9df72;rport=5060
From: "gaia" <sip:gaia@192.168.7.40>;tag=as318313ba
To: <sip:0143620918@192.168.7.36>
Call-ID: 2177e0776ef133741b8423691ce29b5f@192.168.7.40
CSeq: 102 INVITE
Server: Kamailio (1.4.3-notls (i386/linux))
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
ares*CLI>
<--- SIP read from UDP://192.168.7.36:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.7.40:5060;branch=z9hG4bK15b9df72;rport=5060
Record-Route: <sip:0.0.0.0;lr=on;ftag=as318313ba>
From: "gaia" <sip:gaia@192.168.7.40>;tag=as318313ba
To: <sip:0143620918@192.168.7.36>;tag=1880379773
Call-ID: 2177e0776ef133741b8423691ce29b5f@192.168.7.40
CSeq: 102 INVITE
Contact: <sip:0143620918@192.168.8.40:5060>
Server: Patton SN4961 4E30V 00A0BA04A020 R5.3 2009-01-15 H323 RBS SIP M5T SIP Stack/4.0.28.28
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
    -- SIP/ser_sei0-0825feb8 is ringing

<--- Transmitting (NAT) to 217.64.48.36:1162 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 217.64.48.36:5060;branch=z9hG4bK-d8754z-9aa41781bdf61ce6-1---d8754z-;received=217.64.48.36
From: "gaia"<sip:gaia@217.64.49.43;transport=UDP>;tag=411f2d60
To: <sip:0143620918@217.64.49.43;transport=UDP>;tag=as6297643f
Call-ID: Mjk3YWVhNDM2NGFmZjFjZTNhZmMxNzlhMmQ5OWNjOTQ.
CSeq: 2 INVITE
User-Agent: Asterisk PBX 1.6.0.14-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:0143620918@217.64.49.43>
Content-Length: 0


<------------>
ares*CLI>
<--- SIP read from UDP://192.168.7.36:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.7.40:5060;branch=z9hG4bK15b9df72;rport=5060
Record-Route: <sip:0.0.0.0;lr=on;ftag=as318313ba>
From: "gaia" <sip:gaia@192.168.7.40>;tag=as318313ba
To: <sip:0143620918@192.168.7.36>;tag=1880379773
Call-ID: 2177e0776ef133741b8423691ce29b5f@192.168.7.40
CSeq: 102 INVITE
Contact: <sip:0143620918@192.168.8.40:5060>
Server: Patton SN4961 4E30V 00A0BA04A020 R5.3 2009-01-15 H323 RBS SIP M5T SIP Stack/4.0.28.28
Content-Type: application/sdp
Content-Length: 221

v=0
o=MxSIP 0 1501 IN IP4 192.168.8.40
s=SIP Call
c=IN IP4 192.168.8.40
t=0 0
m=audio 5316 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=sendrecv

<------------->
--- (11 headers 11 lines) ---
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 192.168.8.40:5316
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Got unsupported a:fmtp in SDP offer
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.8.40:5316
    -- SIP/ser_sei0-0825feb8 is making progress passing it to SIP/gaia-0824b708
Audio is at 217.64.49.43 port 10108
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
ares*CLI>
<--- Transmitting (NAT) to 217.64.48.36:1162 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 217.64.48.36:5060;branch=z9hG4bK-d8754z-9aa41781bdf61ce6-1---d8754z-;received=217.64.48.36
From: "gaia"<sip:gaia@217.64.49.43;transport=UDP>;tag=411f2d60
To: <sip:0143620918@217.64.49.43;transport=UDP>;tag=as6297643f
Call-ID: Mjk3YWVhNDM2NGFmZjFjZTNhZmMxNzlhMmQ5OWNjOTQ.
CSeq: 2 INVITE
User-Agent: Asterisk PBX 1.6.0.14-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:0143620918@217.64.49.43>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 80690669 80690669 IN IP4 217.64.49.43
s=Asterisk PBX 1.6.0.14-rc1
c=IN IP4 217.64.49.43
t=0 0
m=audio 10108 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=sendrecv

<------------>
ares*CLI>
<--- SIP read from UDP://192.168.7.36:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.7.40:5060;branch=z9hG4bK15b9df72;rport=5060
Record-Route: <sip:0.0.0.0;lr=on;ftag=as318313ba>
From: "gaia" <sip:gaia@192.168.7.40>;tag=as318313ba
To: <sip:0143620918@192.168.7.36>;tag=1880379773
Call-ID: 2177e0776ef133741b8423691ce29b5f@192.168.7.40
CSeq: 102 INVITE
Contact: <sip:0143620918@192.168.8.40:5060>
Server: Patton SN4961 4E30V 00A0BA04A020 R5.3 2009-01-15 H323 RBS SIP M5T SIP Stack/4.0.28.28
Content-Type: application/sdp
Content-Length: 221

v=0
o=MxSIP 0 1501 IN IP4 192.168.8.40
s=SIP Call
c=IN IP4 192.168.8.40
t=0 0
m=audio 5316 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=sendrecv

<------------->
--- (11 headers 11 lines) ---
    -- SIP/ser_sei0-0825feb8 is ringing
    -- SIP/ser_sei0-0825feb8 is making progress passing it to SIP/gaia-0824b708
Reliably Transmitting (no NAT) to 192.168.7.36:5060:
OPTIONS sip:192.168.7.36 SIP/2.0
Via: SIP/2.0/UDP 192.168.7.40:5060;branch=z9hG4bK2e3e3cb6;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.7.40>;tag=as4fa92f64
To: <sip:192.168.7.36>
Contact: <sip:asterisk@192.168.7.40>
Call-ID: 2b9b7240242050ce49d5d82b1777d507@192.168.7.40
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.0.14-rc1
Date: Fri, 28 Aug 2009 06:58:14 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
ares*CLI>
<--- SIP read from UDP://192.168.7.36:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.7.40:5060;branch=z9hG4bK2e3e3cb6;rport=5060
From: "asterisk" <sip:asterisk@192.168.7.40>;tag=as4fa92f64
To: <sip:192.168.7.36>;tag=62f18ef1efcf42af43028c33e1d47f4a.d554
Call-ID: 2b9b7240242050ce49d5d82b1777d507@192.168.7.40
CSeq: 102 OPTIONS
Server: Kamailio (1.4.3-notls (i386/linux))
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '2b9b7240242050ce49d5d82b1777d507@192.168.7.40' Method: OPTIONS
Reliably Transmitting (NAT) to 217.64.49.49:5060:
OPTIONS sip:217.64.49.49 SIP/2.0
Via: SIP/2.0/UDP 217.64.49.43:5060;branch=z9hG4bK775e00de;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@217.64.49.43>;tag=as0fbc3c31
To: <sip:217.64.49.49>
Contact: <sip:asterisk@217.64.49.43>
Call-ID: 0569716061e7a568420812f24822ed54@217.64.49.43
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.0.14-rc1
Date: Fri, 28 Aug 2009 06:58:14 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
ares*CLI>
<--- SIP read from UDP://217.64.49.49:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.64.49.43:5060;branch=z9hG4bK775e00de;rport=5060
From: "asterisk" <sip:asterisk@217.64.49.43>;tag=as0fbc3c31
To: <sip:217.64.49.49>;tag=5838f2f36eb9689a2709b2fd51d40ba0.b3f0
Call-ID: 0569716061e7a568420812f24822ed54@217.64.49.43
CSeq: 102 OPTIONS
Server: Kamailio (1.4.3-notls (i386/linux))
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '0569716061e7a568420812f24822ed54@217.64.49.43' Method: OPTIONS
Reliably Transmitting (NAT) to 217.64.49.49:5060:
OPTIONS sip:217.64.49.49 SIP/2.0
Via: SIP/2.0/UDP 217.64.49.43:5060;branch=z9hG4bK0c6323cb;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@217.64.49.43>;tag=as4623537b
To: <sip:217.64.49.49>
Contact: <sip:asterisk@217.64.49.43>
Call-ID: 05799b4410728cd61dfe5e26388d14f5@217.64.49.43
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.0.14-rc1
Date: Fri, 28 Aug 2009 06:58:14 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
ares*CLI>
<--- SIP read from UDP://217.64.49.49:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.64.49.43:5060;branch=z9hG4bK0c6323cb;rport=5060
From: "asterisk" <sip:asterisk@217.64.49.43>;tag=as4623537b
To: <sip:217.64.49.49>;tag=5838f2f36eb9689a2709b2fd51d40ba0.ea32
Call-ID: 05799b4410728cd61dfe5e26388d14f5@217.64.49.43
CSeq: 102 OPTIONS
Server: Kamailio (1.4.3-notls (i386/linux))
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '05799b4410728cd61dfe5e26388d14f5@217.64.49.43' Method: OPTIONS
Reliably Transmitting (NAT) to 217.64.49.49:5060:
OPTIONS sip:217.64.49.49 SIP/2.0
Via: SIP/2.0/UDP 217.64.49.43:5060;branch=z9hG4bK2e691abb;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@217.64.49.43>;tag=as4be96bea
To: <sip:217.64.49.49>
Contact: <sip:asterisk@217.64.49.43>
Call-ID: 6cc637b7178a9885106aca1b7409b2aa@217.64.49.43
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.0.14-rc1
Date: Fri, 28 Aug 2009 06:58:15 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
ares*CLI>
<--- SIP read from UDP://217.64.49.49:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.64.49.43:5060;branch=z9hG4bK2e691abb;rport=5060
From: "asterisk" <sip:asterisk@217.64.49.43>;tag=as4be96bea
To: <sip:217.64.49.49>;tag=5838f2f36eb9689a2709b2fd51d40ba0.ae63
Call-ID: 6cc637b7178a9885106aca1b7409b2aa@217.64.49.43
CSeq: 102 OPTIONS
Server: Kamailio (1.4.3-notls (i386/linux))
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '6cc637b7178a9885106aca1b7409b2aa@217.64.49.43' Method: OPTIONS
Reliably Transmitting (NAT) to 217.64.49.49:5060:
OPTIONS sip:217.64.49.49 SIP/2.0
Via: SIP/2.0/UDP 217.64.49.43:5060;branch=z9hG4bK2686e8af;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@217.64.49.43>;tag=as50c63794
To: <sip:217.64.49.49>
Contact: <sip:asterisk@217.64.49.43>
Call-ID: 75ae71b86aa22f70169f308d2a4807b4@217.64.49.43
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.0.14-rc1
Date: Fri, 28 Aug 2009 06:58:15 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
ares*CLI>
<--- SIP read from UDP://217.64.49.49:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.64.49.43:5060;branch=z9hG4bK2686e8af;rport=5060
From: "asterisk" <sip:asterisk@217.64.49.43>;tag=as50c63794
To: <sip:217.64.49.49>;tag=5838f2f36eb9689a2709b2fd51d40ba0.e5b6
Call-ID: 75ae71b86aa22f70169f308d2a4807b4@217.64.49.43
CSeq: 102 OPTIONS
Server: Kamailio (1.4.3-notls (i386/linux))
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '75ae71b86aa22f70169f308d2a4807b4@217.64.49.43' Method: OPTIONS
ares*CLI>
<--- SIP read from UDP://192.168.7.36:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.7.40:5060;branch=z9hG4bK15b9df72;rport=5060
Record-Route: <sip:0.0.0.0;lr=on;ftag=as318313ba>
From: "gaia" <sip:gaia@192.168.7.40>;tag=as318313ba
To: <sip:0143620918@192.168.7.36>;tag=1880379773
Call-ID: 2177e0776ef133741b8423691ce29b5f@192.168.7.40
CSeq: 102 INVITE
Contact: <sip:0143620918@192.168.8.40:5060>
Server: Patton SN4961 4E30V 00A0BA04A020 R5.3 2009-01-15 H323 RBS SIP M5T SIP Stack/4.0.28.28
Supported: replaces
Content-Type: application/sdp
Content-Length: 221

v=0
o=MxSIP 0 1501 IN IP4 192.168.8.40
s=SIP Call
c=IN IP4 192.168.8.40
t=0 0
m=audio 5316 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=sendrecv

<------------->
--- (12 headers 11 lines) ---
list_route: hop: <sip:0.0.0.0;lr=on;ftag=as318313ba>
set_destination: Parsing <sip:0.0.0.0;lr=on;ftag=as318313ba> for address/port to send to
set_destination: set destination to 0.0.0.0, port 5060
Transmitting (no NAT) to 0.0.0.0:5060:
ACK sip:0143620918@192.168.8.40:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.7.40:5060;branch=z9hG4bK66925893;rport
Route: <sip:0.0.0.0;lr=on;ftag=as318313ba>
Max-Forwards: 70
From: "gaia" <sip:gaia@192.168.7.40>;tag=as318313ba
To: <sip:0143620918@192.168.7.36>;tag=1880379773
Contact: <sip:gaia@192.168.7.40>
Call-ID: 2177e0776ef133741b8423691ce29b5f@192.168.7.40
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.0.14-rc1
Content-Length: 0


---

<--- SIP read from UDP://127.0.0.1:5060 --->
ACK sip:0143620918@192.168.8.40:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.7.40:5060;branch=z9hG4bK66925893;rport
Route: <sip:0.0.0.0;lr=on;ftag=as318313ba>
Max-Forwards: 70
From: "gaia" <sip:gaia@192.168.7.40>;tag=as318313ba
To: <sip:0143620918@192.168.7.36>;tag=1880379773
Contact: <sip:gaia@192.168.7.40>
Call-ID: 2177e0776ef133741b8423691ce29b5f@192.168.7.40
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.0.14-rc1
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
    -- SIP/ser_sei0-0825feb8 answered SIP/gaia-0824b708
Audio is at 217.64.49.43 port 10108
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 217.64.48.36:1162 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.64.48.36:5060;branch=z9hG4bK-d8754z-9aa41781bdf61ce6-1---d8754z-;received=217.64.48.36
From: "gaia"<sip:gaia@217.64.49.43;transport=UDP>;tag=411f2d60
To: <sip:0143620918@217.64.49.43;transport=UDP>;tag=as6297643f
Call-ID: Mjk3YWVhNDM2NGFmZjFjZTNhZmMxNzlhMmQ5OWNjOTQ.
CSeq: 2 INVITE
User-Agent: Asterisk PBX 1.6.0.14-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:0143620918@217.64.49.43>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 80690669 80690670 IN IP4 217.64.49.43
s=Asterisk PBX 1.6.0.14-rc1
c=IN IP4 217.64.49.43
t=0 0
m=audio 10108 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=sendrecv

<------------>
    -- Packet2Packet bridging SIP/gaia-0824b708 and SIP/ser_sei0-0825feb8
ares*CLI>
<--- SIP read from UDP://217.64.48.36:1162 --->
ACK sip:0143620918@217.64.49.43 SIP/2.0
Via: SIP/2.0/UDP 217.64.48.36:5060;branch=z9hG4bK-d8754z-223345decd18128b-1---d8754z-
Max-Forwards: 70
Contact: <sip:gaia@217.64.48.36:5060;transport=UDP>
To: <sip:0143620918@217.64.49.43;transport=UDP>;tag=as6297643f
From: "gaia"<sip:gaia@217.64.49.43;transport=UDP>;tag=411f2d60
Call-ID: Mjk3YWVhNDM2NGFmZjFjZTNhZmMxNzlhMmQ5OWNjOTQ.
CSeq: 2 ACK
User-Agent: Zoiper rev.4829
Authorization: Digest username="gaia",realm="asterisk",nonce="66c0d26b",uri="sip:0143620918@217.64.49.43;transport=UDP",response="6aeed9d9c56f5311f2d023a970ad9977",algorithm=MD5
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
ares*CLI>
<--- SIP read from UDP://217.64.48.36:1162 --->
INVITE sip:0143620918@217.64.49.43 SIP/2.0
Via: SIP/2.0/UDP 217.64.48.36:5060;branch=z9hG4bK-d8754z-2e6d40454e268f44-1---d8754z-
Max-Forwards: 70
Contact: <sip:gaia@217.64.48.36:5060;transport=UDP>
To: <sip:0143620918@217.64.49.43;transport=UDP>;tag=as6297643f
From: "gaia"<sip:gaia@217.64.49.43;transport=UDP>;tag=411f2d60
Call-ID: Mjk3YWVhNDM2NGFmZjFjZTNhZmMxNzlhMmQ5OWNjOTQ.
CSeq: 3 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
User-Agent: Zoiper rev.4829
Authorization: Digest username="gaia",realm="asterisk",nonce="66c0d26b",uri="sip:0143620918@217.64.49.43",response="f66af77d00061b7bca143b580179416b",algorithm=MD5
Content-Length: 382

v=0
o=Zoiper_pc-francois 924326799 1049062413 IN IP4 192.168.1.100
s=Zoiper_session
c=IN IP4 192.168.1.100
t=0 0
m=image 8000 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:400
a=T38FaxMaxDatagram:400
a=T38FaxUdpEC:t38UDPRedundancy

<------------->
--- (13 headers 15 lines) ---
Sending to 217.64.48.36 : 1162 (NAT)
Got T.38 offer in SDP in dialog Mjk3YWVhNDM2NGFmZjFjZTNhZmMxNzlhMmQ5OWNjOTQ.
Got T.38 Re-invite without audio. Keeping RTP active during T.38 session. Callid Mjk3YWVhNDM2NGFmZjFjZTNhZmMxNzlhMmQ5OWNjOTQ.
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x0 (nothing)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x0 (nothing)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
ares*CLI>
<--- Transmitting (NAT) to 217.64.48.36:1162 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 217.64.48.36:5060;branch=z9hG4bK-d8754z-2e6d40454e268f44-1---d8754z-;received=217.64.48.36
From: "gaia"<sip:gaia@217.64.49.43;transport=UDP>;tag=411f2d60
To: <sip:0143620918@217.64.49.43;transport=UDP>;tag=as6297643f
Call-ID: Mjk3YWVhNDM2NGFmZjFjZTNhZmMxNzlhMmQ5OWNjOTQ.
CSeq: 3 INVITE
User-Agent: Asterisk PBX 1.6.0.14-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:0143620918@217.64.49.43>
Content-Length: 0


<------------>
set_destination: Parsing <sip:0.0.0.0;lr=on;ftag=as318313ba> for address/port to send to
set_destination: set destination to 0.0.0.0, port 5060
Reliably Transmitting (no NAT) to 0.0.0.0:5060:
INVITE sip:0143620918@192.168.8.40:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.7.40:5060;branch=z9hG4bK13e481cf;rport
Route: <sip:0.0.0.0;lr=on;ftag=as318313ba>
Max-Forwards: 70
From: "gaia" <sip:gaia@192.168.7.40>;tag=as318313ba
To: <sip:0143620918@192.168.7.36>;tag=1880379773
Contact: <sip:gaia@192.168.7.40>
Call-ID: 2177e0776ef133741b8423691ce29b5f@192.168.7.40
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.6.0.14-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 269

v=0
o=root 1820870008 1820870009 IN IP4 192.168.7.40
s=Asterisk PBX 1.6.0.14-rc1
c=IN IP4 192.168.7.40
t=0 0
m=image 4181 udptl t38
a=T38Faxversion:0
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxDatagram:195
a=T38FaxUdpEC:t38UDPFEC

---

<--- SIP read from UDP://127.0.0.1:5060 --->
INVITE sip:0143620918@192.168.8.40:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.7.40:5060;branch=z9hG4bK13e481cf;rport
Route: <sip:0.0.0.0;lr=on;ftag=as318313ba>
Max-Forwards: 70
From: "gaia" <sip:gaia@192.168.7.40>;tag=as318313ba
To: <sip:0143620918@192.168.7.36>;tag=1880379773
Contact: <sip:gaia@192.168.7.40>
Call-ID: 2177e0776ef133741b8423691ce29b5f@192.168.7.40
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.6.0.14-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 269

v=0
o=root 1820870008 1820870009 IN IP4 192.168.7.40
s=Asterisk PBX 1.6.0.14-rc1
c=IN IP4 192.168.7.40
t=0 0
m=image 4181 udptl t38
a=T38Faxversion:0
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxDatagram:195
a=T38FaxUdpEC:t38UDPFEC

<------------->
--- (15 headers 11 lines) ---
[Aug 28 08:58:20] WARNING[339]: chan_sip.c:6737 get_ip_and_port_from_sdp: Failed to read an alternate host or port in SDP. Expect audio problems
[Aug 28 08:58:20] WARNING[339]: chan_sip.c:17425 handle_request_invite: Failed to set an alternate media source on glared reinvite. Audio may not work properly on this call.

<--- Reliably Transmitting (no NAT) to 0.0.0.0:5060 --->
SIP/2.0 491 Request Pending
Via: SIP/2.0/UDP 192.168.7.40:5060;branch=z9hG4bK13e481cf;received=127.0.0.1;rport=5060
From: "gaia" <sip:gaia@192.168.7.40>;tag=as318313ba
To: <sip:0143620918@192.168.7.36>;tag=1880379773
Call-ID: 2177e0776ef133741b8423691ce29b5f@192.168.7.40
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.6.0.14-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16


<------------>
ares*CLI>
<--- SIP read from UDP://127.0.0.1:5060 --->
SIP/2.0 491 Request Pending
Via: SIP/2.0/UDP 192.168.7.40:5060;branch=z9hG4bK13e481cf;received=127.0.0.1;rport=5060
From: "gaia" <sip:gaia@192.168.7.40>;tag=as318313ba
To: <sip:0143620918@192.168.7.36>;tag=1880379773
Call-ID: 2177e0776ef133741b8423691ce29b5f@192.168.7.40
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.6.0.14-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16


<------------->
--- (12 headers 0 lines) ---
set_destination: Parsing <sip:0.0.0.0;lr=on;ftag=as318313ba> for address/port to send to
set_destination: set destination to 0.0.0.0, port 5060
Transmitting (no NAT) to 0.0.0.0:5060:
ACK sip:0143620918@192.168.8.40:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.7.40:5060;branch=z9hG4bK13e481cf;rport
Route: <sip:0.0.0.0;lr=on;ftag=as318313ba>
Max-Forwards: 70
From: "gaia" <sip:gaia@192.168.7.40>;tag=as318313ba
To: <sip:0143620918@192.168.7.36>;tag=1880379773
Contact: <sip:gaia@192.168.7.40>
Call-ID: 2177e0776ef133741b8423691ce29b5f@192.168.7.40
CSeq: 103 ACK
User-Agent: Asterisk PBX 1.6.0.14-rc1
Content-Length: 0


---
ares*CLI>
<--- SIP read from UDP://127.0.0.1:5060 --->
ACK sip:0143620918@192.168.8.40:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.7.40:5060;branch=z9hG4bK13e481cf;rport
Route: <sip:0.0.0.0;lr=on;ftag=as318313ba>
Max-Forwards: 70
From: "gaia" <sip:gaia@192.168.7.40>;tag=as318313ba
To: <sip:0143620918@192.168.7.36>;tag=1880379773
Contact: <sip:gaia@192.168.7.40>
Call-ID: 2177e0776ef133741b8423691ce29b5f@192.168.7.40
CSeq: 103 ACK
User-Agent: Asterisk PBX 1.6.0.14-rc1
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
ares*CLI>
<--- SIP read from UDP://192.168.7.36:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.7.40:5060;branch=z9hG4bK15b9df72;rport=5060
Record-Route: <sip:0.0.0.0;lr=on;ftag=as318313ba>
From: "gaia" <sip:gaia@192.168.7.40>;tag=as318313ba
To: <sip:0143620918@192.168.7.36>;tag=1880379773
Call-ID: 2177e0776ef133741b8423691ce29b5f@192.168.7.40
CSeq: 102 INVITE
Contact: <sip:0143620918@192.168.8.40:5060>
Server: Patton SN4961 4E30V 00A0BA04A020 R5.3 2009-01-15 H323 RBS SIP M5T SIP Stack/4.0.28.28
Supported: replaces
Content-Type: application/sdp
Content-Length: 221

v=0
o=MxSIP 0 1501 IN IP4 192.168.8.40
s=SIP Call
c=IN IP4 192.168.8.40
t=0 0
m=audio 5316 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=sendrecv

<------------->
--- (12 headers 11 lines) ---
ares*CLI>
<--- SIP read from UDP://192.168.7.36:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.7.40:5060;branch=z9hG4bK15b9df72;rport=5060
Record-Route: <sip:0.0.0.0;lr=on;ftag=as318313ba>
From: "gaia" <sip:gaia@192.168.7.40>;tag=as318313ba
To: <sip:0143620918@192.168.7.36>;tag=1880379773
Call-ID: 2177e0776ef133741b8423691ce29b5f@192.168.7.40
CSeq: 102 INVITE
Contact: <sip:0143620918@192.168.8.40:5060>
Server: Patton SN4961 4E30V 00A0BA04A020 R5.3 2009-01-15 H323 RBS SIP M5T SIP Stack/4.0.28.28
Supported: replaces
Content-Type: application/sdp
Content-Length: 221

v=0
o=MxSIP 0 1501 IN IP4 192.168.8.40
s=SIP Call
c=IN IP4 192.168.8.40
t=0 0
m=audio 5316 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=sendrecv

<------------->
--- (12 headers 11 lines) ---
ares*CLI>
<--- SIP read from UDP://192.168.7.36:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.7.40:5060;branch=z9hG4bK15b9df72;rport=5060
Record-Route: <sip:0.0.0.0;lr=on;ftag=as318313ba>
From: "gaia" <sip:gaia@192.168.7.40>;tag=as318313ba
To: <sip:0143620918@192.168.7.36>;tag=1880379773
Call-ID: 2177e0776ef133741b8423691ce29b5f@192.168.7.40
CSeq: 102 INVITE
Contact: <sip:0143620918@192.168.8.40:5060>
Server: Patton SN4961 4E30V 00A0BA04A020 R5.3 2009-01-15 H323 RBS SIP M5T SIP Stack/4.0.28.28
Supported: replaces
Content-Type: application/sdp
Content-Length: 221

v=0
o=MxSIP 0 1501 IN IP4 192.168.8.40
s=SIP Call
c=IN IP4 192.168.8.40
t=0 0
m=audio 5316 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=sendrecv

<------------->
--- (12 headers 11 lines) ---
set_destination: Parsing <sip:0.0.0.0;lr=on;ftag=as318313ba> for address/port to send to
set_destination: set destination to 0.0.0.0, port 5060
Reliably Transmitting (no NAT) to 0.0.0.0:5060:
INVITE sip:0143620918@192.168.8.40:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.7.40:5060;branch=z9hG4bK5907e379;rport
Route: <sip:0.0.0.0;lr=on;ftag=as318313ba>
Max-Forwards: 70
From: "gaia" <sip:gaia@192.168.7.40>;tag=as318313ba
To: <sip:0143620918@192.168.7.36>;tag=1880379773
Contact: <sip:gaia@192.168.7.40>
Call-ID: 2177e0776ef133741b8423691ce29b5f@192.168.7.40
CSeq: 104 INVITE
User-Agent: Asterisk PBX 1.6.0.14-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 269

v=0
o=root 1820870008 1820870010 IN IP4 192.168.7.40
s=Asterisk PBX 1.6.0.14-rc1
c=IN IP4 192.168.7.40
t=0 0
m=image 4181 udptl t38
a=T38Faxversion:0
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxDatagram:195
a=T38FaxUdpEC:t38UDPFEC

---
ares*CLI>
<--- SIP read from UDP://127.0.0.1:5060 --->
INVITE sip:0143620918@192.168.8.40:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.7.40:5060;branch=z9hG4bK5907e379;rport
Route: <sip:0.0.0.0;lr=on;ftag=as318313ba>
Max-Forwards: 70
From: "gaia" <sip:gaia@192.168.7.40>;tag=as318313ba
To: <sip:0143620918@192.168.7.36>;tag=1880379773
Contact: <sip:gaia@192.168.7.40>
Call-ID: 2177e0776ef133741b8423691ce29b5f@192.168.7.40
CSeq: 104 INVITE
User-Agent: Asterisk PBX 1.6.0.14-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 269

v=0
o=root 1820870008 1820870010 IN IP4 192.168.7.40
s=Asterisk PBX 1.6.0.14-rc1
c=IN IP4 192.168.7.40
t=0 0
m=image 4181 udptl t38
a=T38Faxversion:0
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxDatagram:195
a=T38FaxUdpEC:t38UDPFEC

<------------->
--- (15 headers 11 lines) ---
[Aug 28 08:58:24] WARNING[339]: chan_sip.c:6737 get_ip_and_port_from_sdp: Failed to read an alternate host or port in SDP. Expect audio problems
[Aug 28 08:58:24] WARNING[339]: chan_sip.c:17425 handle_request_invite: Failed to set an alternate media source on glared reinvite. Audio may not work properly on this call.

<--- Reliably Transmitting (no NAT) to 0.0.0.0:5060 --->
SIP/2.0 491 Request Pending
Via: SIP/2.0/UDP 192.168.7.40:5060;branch=z9hG4bK5907e379;received=127.0.0.1;rport=5060
From: "gaia" <sip:gaia@192.168.7.40>;tag=as318313ba
To: <sip:0143620918@192.168.7.36>;tag=1880379773
Call-ID: 2177e0776ef133741b8423691ce29b5f@192.168.7.40
CSeq: 104 INVITE
User-Agent: Asterisk PBX 1.6.0.14-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16


<------------>
ares*CLI>
<--- SIP read from UDP://127.0.0.1:5060 --->
SIP/2.0 491 Request Pending
Via: SIP/2.0/UDP 192.168.7.40:5060;branch=z9hG4bK5907e379;received=127.0.0.1;rport=5060
From: "gaia" <sip:gaia@192.168.7.40>;tag=as318313ba
To: <sip:0143620918@192.168.7.36>;tag=1880379773
Call-ID: 2177e0776ef133741b8423691ce29b5f@192.168.7.40
CSeq: 104 INVITE
User-Agent: Asterisk PBX 1.6.0.14-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16


<------------->
--- (12 headers 0 lines) ---
set_destination: Parsing <sip:0.0.0.0;lr=on;ftag=as318313ba> for address/port to send to
set_destination: set destination to 0.0.0.0, port 5060
Transmitting (no NAT) to 0.0.0.0:5060:
ACK sip:0143620918@192.168.8.40:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.7.40:5060;branch=z9hG4bK5907e379;rport
Route: <sip:0.0.0.0;lr=on;ftag=as318313ba>
Max-Forwards: 70
From: "gaia" <sip:gaia@192.168.7.40>;tag=as318313ba
To: <sip:0143620918@192.168.7.36>;tag=1880379773
Contact: <sip:gaia@192.168.7.40>
Call-ID: 2177e0776ef133741b8423691ce29b5f@192.168.7.40
CSeq: 104 ACK
User-Agent: Asterisk PBX 1.6.0.14-rc1
Content-Length: 0


---
ares*CLI>
<--- SIP read from UDP://127.0.0.1:5060 --->
ACK sip:0143620918@192.168.8.40:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.7.40:5060;branch=z9hG4bK5907e379;rport
Route: <sip:0.0.0.0;lr=on;ftag=as318313ba>
Max-Forwards: 70
From: "gaia" <sip:gaia@192.168.7.40>;tag=as318313ba
To: <sip:0143620918@192.168.7.36>;tag=1880379773
Contact: <sip:gaia@192.168.7.40>
Call-ID: 2177e0776ef133741b8423691ce29b5f@192.168.7.40
CSeq: 104 ACK
User-Agent: Asterisk PBX 1.6.0.14-rc1
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
ares*CLI>
<--- SIP read from UDP://217.64.48.36:5060 --->

<------------->
ares*CLI>
<--- Reliably Transmitting (NAT) to 217.64.48.36:1162 --->
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP 217.64.48.36:5060;branch=z9hG4bK-d8754z-2e6d40454e268f44-1---d8754z-;received=217.64.48.36
From: "gaia"<sip:gaia@217.64.49.43;transport=UDP>;tag=411f2d60
To: <sip:0143620918@217.64.49.43;transport=UDP>;tag=as6297643f
Call-ID: Mjk3YWVhNDM2NGFmZjFjZTNhZmMxNzlhMmQ5OWNjOTQ.
CSeq: 3 INVITE
User-Agent: Asterisk PBX 1.6.0.14-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16


<------------>
ares*CLI>
<--- SIP read from UDP://217.64.48.36:1162 --->
ACK sip:0143620918@217.64.49.43 SIP/2.0
Via: SIP/2.0/UDP 217.64.48.36:5060;branch=z9hG4bK-d8754z-2e6d40454e268f44-1---d8754z-
Max-Forwards: 70
To: <sip:0143620918@217.64.49.43;transport=UDP>;tag=as6297643f
From: "gaia"<sip:gaia@217.64.49.43;transport=UDP>;tag=411f2d60
Call-ID: Mjk3YWVhNDM2NGFmZjFjZTNhZmMxNzlhMmQ5OWNjOTQ.
CSeq: 3 ACK
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
ares*CLI>
<--- SIP read from UDP://217.64.48.36:1162 --->
BYE sip:0143620918@217.64.49.43 SIP/2.0
Via: SIP/2.0/UDP 217.64.48.36:5060;branch=z9hG4bK-d8754z-f2119dd9d4493135-1---d8754z-
Max-Forwards: 70
Contact: <sip:gaia@217.64.48.36:5060;transport=UDP>
To: <sip:0143620918@217.64.49.43;transport=UDP>;tag=as6297643f
From: "gaia"<sip:gaia@217.64.49.43;transport=UDP>;tag=411f2d60
Call-ID: Mjk3YWVhNDM2NGFmZjFjZTNhZmMxNzlhMmQ5OWNjOTQ.
CSeq: 4 BYE
User-Agent: Zoiper rev.4829
Authorization: Digest username="gaia",realm="asterisk",nonce="66c0d26b",uri="sip:0143620918@217.64.49.43",response="e8b8111690038e01743fee073d22e30d",algorithm=MD5
Content-Length: 0


<------------->
--- (11 headers 0 lines) ---
Sending to 217.64.48.36 : 1162 (NAT)
ares*CLI>
<--- Transmitting (NAT) to 217.64.48.36:1162 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.64.48.36:5060;branch=z9hG4bK-d8754z-f2119dd9d4493135-1---d8754z-;received=217.64.48.36
From: "gaia"<sip:gaia@217.64.49.43;transport=UDP>;tag=411f2d60
To: <sip:0143620918@217.64.49.43;transport=UDP>;tag=as6297643f
Call-ID: Mjk3YWVhNDM2NGFmZjFjZTNhZmMxNzlhMmQ5OWNjOTQ.
CSeq: 4 BYE
User-Agent: Asterisk PBX 1.6.0.14-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '2177e0776ef133741b8423691ce29b5f@192.168.7.40' in 6400 ms (Method: ACK)
set_destination: Parsing <sip:0.0.0.0;lr=on;ftag=as318313ba> for address/port to send to
set_destination: set destination to 0.0.0.0, port 5060
Reliably Transmitting (no NAT) to 0.0.0.0:5060:
BYE sip:0143620918@192.168.8.40:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.7.40:5060;branch=z9hG4bK11b60a35;rport
Route: <sip:0.0.0.0;lr=on;ftag=as318313ba>
Max-Forwards: 70
From: "gaia" <sip:gaia@192.168.7.40>;tag=as318313ba
To: <sip:0143620918@192.168.7.36>;tag=1880379773
Call-ID: 2177e0776ef133741b8423691ce29b5f@192.168.7.40
CSeq: 105 BYE
User-Agent: Asterisk PBX 1.6.0.14-rc1
X-Asterisk-HangupCause: Interworking, unspecified
X-Asterisk-HangupCauseCode: 127
Content-Length: 0


---
  == Spawn extension (t38, 0143620918, 10) exited non-zero on 'SIP/gaia-0824b708'
ares*CLI>
<--- SIP read from UDP://127.0.0.1:5060 --->
BYE sip:0143620918@192.168.8.40:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.7.40:5060;branch=z9hG4bK11b60a35;rport
Route: <sip:0.0.0.0;lr=on;ftag=as318313ba>
Max-Forwards: 70
From: "gaia" <sip:gaia@192.168.7.40>;tag=as318313ba
To: <sip:0143620918@192.168.7.36>;tag=1880379773
Call-ID: 2177e0776ef133741b8423691ce29b5f@192.168.7.40
CSeq: 105 BYE
User-Agent: Asterisk PBX 1.6.0.14-rc1
X-Asterisk-HangupCause: Interworking, unspecified
X-Asterisk-HangupCauseCode: 127
Content-Length: 0


<------------->
--- (12 headers 0 lines) ---
Sending to 192.168.7.40 : 5060 (no NAT)
Scheduling destruction of SIP dialog '2177e0776ef133741b8423691ce29b5f@192.168.7.40' in 6400 ms (Method: BYE)
ares*CLI>
<--- Transmitting (no NAT) to 192.168.7.40:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.7.40:5060;branch=z9hG4bK11b60a35;received=127.0.0.1;rport=5060
From: "gaia" <sip:gaia@192.168.7.40>;tag=as318313ba
To: <sip:0143620918@192.168.7.36>;tag=1880379773
Call-ID: 2177e0776ef133741b8423691ce29b5f@192.168.7.40
CSeq: 105 BYE
User-Agent: Asterisk PBX 1.6.0.14-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------>
Really destroying SIP dialog 'Mjk3YWVhNDM2NGFmZjFjZTNhZmMxNzlhMmQ5OWNjOTQ.' Method: BYE
ares*CLI>
<--- SIP read from UDP://192.168.7.40:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.7.40:5060;branch=z9hG4bK11b60a35;received=127.0.0.1;rport=5060
From: "gaia" <sip:gaia@192.168.7.40>;tag=as318313ba
To: <sip:0143620918@192.168.7.36>;tag=1880379773
Call-ID: 2177e0776ef133741b8423691ce29b5f@192.168.7.40
CSeq: 105 BYE
User-Agent: Asterisk PBX 1.6.0.14-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
set_destination: Parsing <sip:0.0.0.0;lr=on;ftag=as318313ba> for address/port to send to
set_destination: set destination to 0.0.0.0, port 5060
Reliably Transmitting (no NAT) to 0.0.0.0:5060:
INVITE sip:0143620918@192.168.8.40:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.7.40:5060;branch=z9hG4bK330e1b14;rport
Route: <sip:0.0.0.0;lr=on;ftag=as318313ba>
Max-Forwards: 70
From: "gaia" <sip:gaia@192.168.7.40>;tag=as318313ba
To: <sip:0143620918@192.168.7.36>;tag=1880379773
Contact: <sip:gaia@192.168.7.40>
Call-ID: 2177e0776ef133741b8423691ce29b5f@192.168.7.40
CSeq: 106 INVITE
User-Agent: Asterisk PBX 1.6.0.14-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 269

v=0
o=root 1820870008 1820870011 IN IP4 192.168.7.40
s=Asterisk PBX 1.6.0.14-rc1
c=IN IP4 192.168.7.40
t=0 0
m=image 4181 udptl t38
a=T38Faxversion:0
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxDatagram:195
a=T38FaxUdpEC:t38UDPFEC

---
ares*CLI>
<--- SIP read from UDP://127.0.0.1:5060 --->
INVITE sip:0143620918@192.168.8.40:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.7.40:5060;branch=z9hG4bK330e1b14;rport
Route: <sip:0.0.0.0;lr=on;ftag=as318313ba>
Max-Forwards: 70
From: "gaia" <sip:gaia@192.168.7.40>;tag=as318313ba
To: <sip:0143620918@192.168.7.36>;tag=1880379773
Contact: <sip:gaia@192.168.7.40>
Call-ID: 2177e0776ef133741b8423691ce29b5f@192.168.7.40
CSeq: 106 INVITE
User-Agent: Asterisk PBX 1.6.0.14-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 269

v=0
o=root 1820870008 1820870011 IN IP4 192.168.7.40
s=Asterisk PBX 1.6.0.14-rc1
c=IN IP4 192.168.7.40
t=0 0
m=image 4181 udptl t38
a=T38Faxversion:0
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxDatagram:195
a=T38FaxUdpEC:t38UDPFEC

<------------->
--- (15 headers 11 lines) ---
[Aug 28 08:58:27] WARNING[339]: chan_sip.c:6737 get_ip_and_port_from_sdp: Failed to read an alternate host or port in SDP. Expect audio problems
[Aug 28 08:58:27] WARNING[339]: chan_sip.c:17425 handle_request_invite: Failed to set an alternate media source on glared reinvite. Audio may not work properly on this call.

<--- Reliably Transmitting (no NAT) to 0.0.0.0:5060 --->
SIP/2.0 491 Request Pending
Via: SIP/2.0/UDP 192.168.7.40:5060;branch=z9hG4bK330e1b14;received=127.0.0.1;rport=5060
From: "gaia" <sip:gaia@192.168.7.40>;tag=as318313ba
To: <sip:0143620918@192.168.7.36>;tag=1880379773
Call-ID: 2177e0776ef133741b8423691ce29b5f@192.168.7.40
CSeq: 106 INVITE
User-Agent: Asterisk PBX 1.6.0.14-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------>

<--- SIP read from UDP://127.0.0.1:5060 --->
SIP/2.0 491 Request Pending
Via: SIP/2.0/UDP 192.168.7.40:5060;branch=z9hG4bK330e1b14;received=127.0.0.1;rport=5060
From: "gaia" <sip:gaia@192.168.7.40>;tag=as318313ba
To: <sip:0143620918@192.168.7.36>;tag=1880379773
Call-ID: 2177e0776ef133741b8423691ce29b5f@192.168.7.40
CSeq: 106 INVITE
User-Agent: Asterisk PBX 1.6.0.14-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
ares*CLI>
<--- SIP read from UDP://192.168.7.36:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.7.40:5060;branch=z9hG4bK15b9df72;rport=5060
Record-Route: <sip:0.0.0.0;lr=on;ftag=as318313ba>
From: "gaia" <sip:gaia@192.168.7.40>;tag=as318313ba
To: <sip:0143620918@192.168.7.36>;tag=1880379773
Call-ID: 2177e0776ef133741b8423691ce29b5f@192.168.7.40
CSeq: 102 INVITE
Contact: <sip:0143620918@192.168.8.40:5060>
Server: Patton SN4961 4E30V 00A0BA04A020 R5.3 2009-01-15 H323 RBS SIP M5T SIP Stack/4.0.28.28
Supported: replaces
Content-Type: application/sdp
Content-Length: 221

v=0
o=MxSIP 0 1501 IN IP4 192.168.8.40
s=SIP Call
c=IN IP4 192.168.8.40
t=0 0
m=audio 5316 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=sendrecv

<------------->
--- (12 headers 11 lines) ---
Retransmitting #1 (no NAT) to 0.0.0.0:5060:
SIP/2.0 491 Request Pending
Via: SIP/2.0/UDP 192.168.7.40:5060;branch=z9hG4bK330e1b14;received=127.0.0.1;rport=5060
From: "gaia" <sip:gaia@192.168.7.40>;tag=as318313ba
To: <sip:0143620918@192.168.7.36>;tag=1880379773
Call-ID: 2177e0776ef133741b8423691ce29b5f@192.168.7.40
CSeq: 106 INVITE
User-Agent: Asterisk PBX 1.6.0.14-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
ares*CLI>
<--- SIP read from UDP://127.0.0.1:5060 --->
SIP/2.0 491 Request Pending
Via: SIP/2.0/UDP 192.168.7.40:5060;branch=z9hG4bK330e1b14;received=127.0.0.1;rport=5060
From: "gaia" <sip:gaia@192.168.7.40>;tag=as318313ba
To: <sip:0143620918@192.168.7.36>;tag=1880379773
Call-ID: 2177e0776ef133741b8423691ce29b5f@192.168.7.40
CSeq: 106 INVITE
User-Agent: Asterisk PBX 1.6.0.14-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
Retransmitting #2 (no NAT) to 0.0.0.0:5060:
SIP/2.0 491 Request Pending
Via: SIP/2.0/UDP 192.168.7.40:5060;branch=z9hG4bK330e1b14;received=127.0.0.1;rport=5060
From: "gaia" <sip:gaia@192.168.7.40>;tag=as318313ba
To: <sip:0143620918@192.168.7.36>;tag=1880379773
Call-ID: 2177e0776ef133741b8423691ce29b5f@192.168.7.40
CSeq: 106 INVITE
User-Agent: Asterisk PBX 1.6.0.14-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
ares*CLI>
<--- SIP read from UDP://127.0.0.1:5060 --->
SIP/2.0 491 Request Pending
Via: SIP/2.0/UDP 192.168.7.40:5060;branch=z9hG4bK330e1b14;received=127.0.0.1;rport=5060
From: "gaia" <sip:gaia@192.168.7.40>;tag=as318313ba
To: <sip:0143620918@192.168.7.36>;tag=1880379773
Call-ID: 2177e0776ef133741b8423691ce29b5f@192.168.7.40
CSeq: 106 INVITE
User-Agent: Asterisk PBX 1.6.0.14-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
Retransmitting #3 (no NAT) to 0.0.0.0:5060:
SIP/2.0 491 Request Pending
Via: SIP/2.0/UDP 192.168.7.40:5060;branch=z9hG4bK330e1b14;received=127.0.0.1;rport=5060
From: "gaia" <sip:gaia@192.168.7.40>;tag=as318313ba
To: <sip:0143620918@192.168.7.36>;tag=1880379773
Call-ID: 2177e0776ef133741b8423691ce29b5f@192.168.7.40
CSeq: 106 INVITE
User-Agent: Asterisk PBX 1.6.0.14-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
ares*CLI>
<--- SIP read from UDP://127.0.0.1:5060 --->
SIP/2.0 491 Request Pending
Via: SIP/2.0/UDP 192.168.7.40:5060;branch=z9hG4bK330e1b14;received=127.0.0.1;rport=5060
From: "gaia" <sip:gaia@192.168.7.40>;tag=as318313ba
To: <sip:0143620918@192.168.7.36>;tag=1880379773
Call-ID: 2177e0776ef133741b8423691ce29b5f@192.168.7.40
CSeq: 106 INVITE
User-Agent: Asterisk PBX 1.6.0.14-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
Retransmitting #4 (no NAT) to 0.0.0.0:5060:
SIP/2.0 491 Request Pending
Via: SIP/2.0/UDP 192.168.7.40:5060;branch=z9hG4bK330e1b14;received=127.0.0.1;rport=5060
From: "gaia" <sip:gaia@192.168.7.40>;tag=as318313ba
To: <sip:0143620918@192.168.7.36>;tag=1880379773
Call-ID: 2177e0776ef133741b8423691ce29b5f@192.168.7.40
CSeq: 106 INVITE
User-Agent: Asterisk PBX 1.6.0.14-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
ares*CLI>
<--- SIP read from UDP://127.0.0.1:5060 --->
SIP/2.0 491 Request Pending
Via: SIP/2.0/UDP 192.168.7.40:5060;branch=z9hG4bK330e1b14;received=127.0.0.1;rport=5060
From: "gaia" <sip:gaia@192.168.7.40>;tag=as318313ba
To: <sip:0143620918@192.168.7.36>;tag=1880379773
Call-ID: 2177e0776ef133741b8423691ce29b5f@192.168.7.40
CSeq: 106 INVITE
User-Agent: Asterisk PBX 1.6.0.14-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
ares*CLI>