rendered paste body---
pbx*CLI>
<-- SIP read from 83.125.8.46:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.15:5060;branch=z9hG4bK7ff7d6bb;received=81.214.188.91;rport=5060
From: "1008" <sip:000387233745@192.168.2.15>;tag=as09059a05
To: <sip:0049305110511@voip.dus.net>;tag=as0af265a6
Call-ID: 40f97cd50b4b182b5ed9d76438b60c5a@192.168.2.15
CSeq: 104 BYE
User-Agent: SIP_GW_1.4.2.PL3 talos.dus.net
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:0049305110511@83.125.8.46>
Content-Length: 0
--- (11 headers 0 lines) ---
Destroying call '40f97cd50b4b182b5ed9d76438b60c5a@192.168.2.15'
Destroying call '21969694911200041956@10.0.0.2'
pbx*CLI>
<-- SIP read from 10.0.0.2:5060:
INVITE sip:071311298862@10.0.0.1;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.0.0.2;branch=z9hG4bKac236762393
Max-Forwards: 70
From: <sip:1024@10.0.0.1>;tag=1c236760662
To: <sip:071311298862@10.0.0.1;user=phone>
Call-ID: 23676042311200042010@10.0.0.2
CSeq: 1 INVITE
Contact: <sip:1024@10.0.0.2>
Supported: em,100rel,timer,replaces,path,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Audiocodes-Sip-Gateway-MP-124 FXS/v.5.00A.027.005
Content-Type: application/sdp
Content-Length: 272
v=0
o=AudiocodesGW 236753870 236753787 IN IP4 10.0.0.2
s=Phone-Call
c=IN IP4 10.0.0.2
t=0 0
m=audio 6230 RTP/AVP 18 96
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=ptime:20
a=sendrecv
a=rtcp:6231 IN IP4 10.0.0.2
--- (13 headers 13 lines) ---
Using INVITE request as basis request - 23676042311200042010@10.0.0.2
Sending to 10.0.0.2 : 5060 (non-NAT)
Reliably Transmitting (no NAT) to 10.0.0.2:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.0.0.2;branch=z9hG4bKac236762393;received=10.0.0.2
From: <sip:1024@10.0.0.1>;tag=1c236760662
To: <sip:071311298862@10.0.0.1;user=phone>;tag=as50a3ddcd
Call-ID: 23676042311200042010@10.0.0.2
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="3797ccf7"
Content-Length: 0
---
Scheduling destruction of call '23676042311200042010@10.0.0.2' in 15000 ms
Found user '1024'
pbx*CLI>
<-- SIP read from 10.0.0.2:5060:
ACK sip:071311298862@10.0.0.1;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.0.0.2;branch=z9hG4bKac236762393
Max-Forwards: 70
From: <sip:1024@10.0.0.1>;tag=1c236760662
To: <sip:071311298862@10.0.0.1;user=phone>;tag=as50a3ddcd
Call-ID: 23676042311200042010@10.0.0.2
CSeq: 1 ACK
Contact: <sip:1024@10.0.0.2>
Supported: em,timer,replaces,path,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Audiocodes-Sip-Gateway-MP-124 FXS/v.5.00A.027.005
Content-Length: 0
--- (12 headers 0 lines) ---
pbx*CLI>
<-- SIP read from 10.0.0.2:5060:
INVITE sip:071311298862@10.0.0.1;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.0.0.2;branch=z9hG4bKac236789509
Max-Forwards: 70
From: <sip:1024@10.0.0.1>;tag=1c236760662
To: <sip:071311298862@10.0.0.1;user=phone>
Call-ID: 23676042311200042010@10.0.0.2
CSeq: 2 INVITE
Proxy-Authorization: Digest username="1024",realm="asterisk",nonce="3797ccf7",uri="sip:071311298862@10.0.0.1",algorithm=MD5,response="2c03f960b4d9ebe940c111c74d0bf7d0"
Contact: <sip:1024@10.0.0.2>
Supported: em,100rel,timer,replaces,path,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Audiocodes-Sip-Gateway-MP-124 FXS/v.5.00A.027.005
Content-Type: application/sdp
Content-Length: 272
v=0
o=AudiocodesGW 236753870 236753787 IN IP4 10.0.0.2
s=Phone-Call
c=IN IP4 10.0.0.2
t=0 0
m=audio 6230 RTP/AVP 18 96
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=ptime:20
a=sendrecv
a=rtcp:6231 IN IP4 10.0.0.2
--- (14 headers 13 lines) ---
Using INVITE request as basis request - 23676042311200042010@10.0.0.2
Sending to 10.0.0.2 : 5060 (non-NAT)
Found user '1024'
Found RTP audio format 18
Found RTP audio format 96
Peer audio RTP is at port 10.0.0.2:6230
Found description format G729
Found description format telephone-event
Capabilities: us - 0x50e (gsm|ulaw|alaw|g729|ilbc), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 071311298862 in masus (domain 10.0.0.1;user=phone)
list_route: hop: <sip:1024@10.0.0.2>
Transmitting (no NAT) to 10.0.0.2:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.0.2;branch=z9hG4bKac236789509;received=10.0.0.2
From: <sip:1024@10.0.0.1>;tag=1c236760662
To: <sip:071311298862@10.0.0.1;user=phone>
Call-ID: 23676042311200042010@10.0.0.2
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:071311298862@10.0.0.1>
Content-Length: 0
---
We're at 192.168.2.15 port 29314
Adding codec 0x100 (g729) to SDP
Adding codec 0x400 (ilbc) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 15 lines
Reliably Transmitting (NAT) to 83.125.8.46:5060:
INVITE sip:004971311298862@voip.dus.net SIP/2.0
Via: SIP/2.0/UDP 192.168.2.15:5060;branch=z9hG4bK01dcd9f7;rport
From: "1024" <sip:000387233745@192.168.2.15>;tag=as20b9cbea
To: <sip:004971311298862@voip.dus.net>
Contact: <sip:000387233745@192.168.2.15>
Call-ID: 70e04fe371dd539c6cab3d4627e544f2@192.168.2.15
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 22 Apr 2008 15:39:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 334
v=0
o=root 8992 8992 IN IP4 192.168.2.15
s=session
c=IN IP4 192.168.2.15
t=0 0
m=audio 29314 RTP/AVP 18 97 3 0 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:97 iLBC/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
pbx*CLI>
<-- SIP read from 83.125.8.46:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.2.15:5060;branch=z9hG4bK01dcd9f7;received=81.214.188.91;rport=5060
From: "1024" <sip:000387233745@192.168.2.15>;tag=as20b9cbea
To: <sip:004971311298862@voip.dus.net>;tag=as71a26820
Call-ID: 70e04fe371dd539c6cab3d4627e544f2@192.168.2.15
CSeq: 102 INVITE
User-Agent: SIP_GW_1.4.2.PL3 talos.dus.net
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="dus.net", nonce="10993390"
Content-Length: 0
--- (11 headers 0 lines) ---
Transmitting (NAT) to 83.125.8.46:5060:
ACK sip:004971311298862@voip.dus.net SIP/2.0
Via: SIP/2.0/UDP 192.168.2.15:5060;branch=z9hG4bK01dcd9f7;rport
From: "1024" <sip:000387233745@192.168.2.15>;tag=as20b9cbea
To: <sip:004971311298862@voip.dus.net>;tag=as71a26820
Contact: <sip:000387233745@192.168.2.15>
Call-ID: 70e04fe371dd539c6cab3d4627e544f2@192.168.2.15
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
We're at 192.168.2.15 port 29314
Adding codec 0x100 (g729) to SDP
Adding codec 0x400 (ilbc) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 83.125.8.46:5060:
INVITE sip:004971311298862@voip.dus.net SIP/2.0
Via: SIP/2.0/UDP 192.168.2.15:5060;branch=z9hG4bK501ab53e;rport
From: "1024" <sip:000387233745@192.168.2.15>;tag=as20b9cbea
To: <sip:004971311298862@voip.dus.net>
Contact: <sip:000387233745@192.168.2.15>
Call-ID: 70e04fe371dd539c6cab3d4627e544f2@192.168.2.15
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Proxy-Authorization: Digest username="000387233745", realm="dus.net", algorithm=MD5, uri="sip:004971311298862@voip.dus.net", nonce="10993390", response="11835a648761c7ce8d3c4c6a2670fb05", opaque=""
Date: Tue, 22 Apr 2008 15:39:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 334
v=0
o=root 8992 8993 IN IP4 192.168.2.15
s=session
c=IN IP4 192.168.2.15
t=0 0
m=audio 29314 RTP/AVP 18 97 3 0 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:97 iLBC/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
pbx*CLI>
<-- SIP read from 83.125.8.46:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.15:5060;branch=z9hG4bK501ab53e;received=81.214.188.91;rport=5060
From: "1024" <sip:000387233745@192.168.2.15>;tag=as20b9cbea
To: <sip:004971311298862@voip.dus.net>
Call-ID: 70e04fe371dd539c6cab3d4627e544f2@192.168.2.15
CSeq: 103 INVITE
User-Agent: SIP_GW_1.4.2.PL3 talos.dus.net
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:004971311298862@83.125.8.46>
Content-Length: 0
--- (11 headers 0 lines) ---
pbx*CLI>
<-- SIP read from 83.125.8.46:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.15:5060;branch=z9hG4bK48238559;received=81.214.188.91;rport=5060
From: "1001" <sip:000387233745@192.168.2.15>;tag=as1387477f
To: <sip:004950519159871@voip.dus.net>;tag=as3c1e61a9
Call-ID: 0b760e162c18bdc90283a21f38126586@192.168.2.15
CSeq: 103 INVITE
User-Agent: SIP_GW_1.4.2.PL3 talos.dus.net
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:004950519159871@83.125.8.46>
Content-Type: application/sdp
Content-Length: 351
v=0
o=root 1387 1388 IN IP4 83.125.8.46
s=session
c=IN IP4 83.125.8.46
t=0 0
m=audio 58704 RTP/AVP 8 3 97 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
--- (12 headers 17 lines) ---
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 97
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 83.125.8.46:58704
Found description format PCMA
Found description format GSM
Found description format iLBC
Found description format G729
Found description format telephone-event
Capabilities: us - 0x50e (gsm|ulaw|alaw|g729|ilbc), peer - audio=0x50a (gsm|alaw|g729|ilbc)/video=0x0 (nothing), combined - 0x50a (gsm|alaw|g729|ilbc)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
list_route: hop: <sip:004950519159871@83.125.8.46>
set_destination: Parsing <sip:004950519159871@83.125.8.46> for address/port to send to
set_destination: set destination to 83.125.8.46, port 5060
Transmitting (NAT) to 83.125.8.46:5060:
ACK sip:004950519159871@83.125.8.46 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.15:5060;branch=z9hG4bK62715351;rport
From: "1001" <sip:000387233745@192.168.2.15>;tag=as1387477f
To: <sip:004950519159871@voip.dus.net>;tag=as3c1e61a9
Contact: <sip:000387233745@192.168.2.15>
Call-ID: 0b760e162c18bdc90283a21f38126586@192.168.2.15
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
We're at 10.0.0.1 port 24560
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.0.0.2:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.2;branch=z9hG4bKac224598576;received=10.0.0.2
From: <sip:1001@10.0.0.1>;tag=1c224543595
To: <sip:050519159871@10.0.0.1;user=phone>;tag=as3b3a07ab
Call-ID: 2245433591120004200@10.0.0.2
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:050519159871@10.0.0.1>
Content-Type: application/sdp
Content-Length: 226
v=0
o=root 8992 8993 IN IP4 10.0.0.1
s=session
c=IN IP4 10.0.0.1
t=0 0
m=audio 24560 RTP/AVP 18 96
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=silenceSupp:off - - - -
---
pbx*CLI>
<-- SIP read from 10.0.0.2:5060:
ACK sip:050519159871@10.0.0.1 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.2;branch=z9hG4bKac237196027
Max-Forwards: 70
From: <sip:1001@10.0.0.1>;tag=1c224543595
To: <sip:050519159871@10.0.0.1;user=phone>;tag=as3b3a07ab
Call-ID: 2245433591120004200@10.0.0.2
CSeq: 2 ACK
Contact: <sip:1001@10.0.0.2>
Supported: em,timer,replaces,path,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Audiocodes-Sip-Gateway-MP-124 FXS/v.5.00A.027.005
Content-Length: 0
--- (12 headers 0 lines) ---
pbx*CLI>
<-- SIP read from 83.125.8.46:5060:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.2.15:5060;branch=z9hG4bK501ab53e;received=81.214.188.91;rport=5060
From: "1024" <sip:000387233745@192.168.2.15>;tag=as20b9cbea
To: <sip:004971311298862@voip.dus.net>;tag=as73d777d3
Call-ID: 70e04fe371dd539c6cab3d4627e544f2@192.168.2.15
CSeq: 103 INVITE
User-Agent: SIP_GW_1.4.2.PL3 talos.dus.net
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:004971311298862@83.125.8.46>
Content-Type: application/sdp
Content-Length: 351
v=0
o=root 1387 1387 IN IP4 83.125.8.46
s=session
c=IN IP4 83.125.8.46
t=0 0
m=audio 20182 RTP/AVP 8 3 97 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
--- (12 headers 17 lines) ---
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 97
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 83.125.8.46:20182
Found description format PCMA
Found description format GSM
Found description format iLBC
Found description format G729
Found description format telephone-event
Capabilities: us - 0x50e (gsm|ulaw|alaw|g729|ilbc), peer - audio=0x50a (gsm|alaw|g729|ilbc)/video=0x0 (nothing), combined - 0x50a (gsm|alaw|g729|ilbc)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
We're at 10.0.0.1 port 12932
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Transmitting (no NAT) to 10.0.0.2:5060:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.0.0.2;branch=z9hG4bKac236789509;received=10.0.0.2
From: <sip:1024@10.0.0.1>;tag=1c236760662
To: <sip:071311298862@10.0.0.1;user=phone>;tag=as35f994c9
Call-ID: 23676042311200042010@10.0.0.2
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:071311298862@10.0.0.1>
Content-Type: application/sdp
Content-Length: 226
v=0
o=root 8992 8992 IN IP4 10.0.0.1
s=session
c=IN IP4 10.0.0.1
t=0 0
m=audio 12932 RTP/AVP 18 96
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=silenceSupp:off - - - -
---
pbx*CLI>
<-- SIP read from 10.0.0.2:5060:
INVITE sip:02452155048@10.0.0.1;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.0.0.2;branch=z9hG4bKac239637731
Max-Forwards: 70
From: <sip:1008@10.0.0.1>;tag=1c239635997
To: <sip:02452155048@10.0.0.1;user=phone>
Call-ID: 23963575711200042012@10.0.0.2
CSeq: 1 INVITE
Contact: <sip:1008@10.0.0.2>
Supported: em,100rel,timer,replaces,path,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Audiocodes-Sip-Gateway-MP-124 FXS/v.5.00A.027.005
Content-Type: application/sdp
Content-Length: 272
v=0
o=AudiocodesGW 239629226 239629145 IN IP4 10.0.0.2
s=Phone-Call
c=IN IP4 10.0.0.2
t=0 0
m=audio 6070 RTP/AVP 18 96
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=ptime:20
a=sendrecv
a=rtcp:6071 IN IP4 10.0.0.2
--- (13 headers 13 lines) ---
Using INVITE request as basis request - 23963575711200042012@10.0.0.2
Sending to 10.0.0.2 : 5060 (non-NAT)
Reliably Transmitting (no NAT) to 10.0.0.2:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.0.0.2;branch=z9hG4bKac239637731;received=10.0.0.2
From: <sip:1008@10.0.0.1>;tag=1c239635997
To: <sip:02452155048@10.0.0.1;user=phone>;tag=as752da0c8
Call-ID: 23963575711200042012@10.0.0.2
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="05560f2b"
Content-Length: 0
---
Scheduling destruction of call '23963575711200042012@10.0.0.2' in 15000 ms
Found user '1008'
pbx*CLI>
<-- SIP read from 10.0.0.2:5060:
ACK sip:02452155048@10.0.0.1;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.0.0.2;branch=z9hG4bKac239637731
Max-Forwards: 70
From: <sip:1008@10.0.0.1>;tag=1c239635997
To: <sip:02452155048@10.0.0.1;user=phone>;tag=as752da0c8
Call-ID: 23963575711200042012@10.0.0.2
CSeq: 1 ACK
Contact: <sip:1008@10.0.0.2>
Supported: em,timer,replaces,path,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Audiocodes-Sip-Gateway-MP-124 FXS/v.5.00A.027.005
Content-Length: 0
--- (12 headers 0 lines) ---
pbx*CLI>
<-- SIP read from 10.0.0.2:5060:
INVITE sip:02452155048@10.0.0.1;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.0.0.2;branch=z9hG4bKac239664674
Max-Forwards: 70
From: <sip:1008@10.0.0.1>;tag=1c239635997
To: <sip:02452155048@10.0.0.1;user=phone>
Call-ID: 23963575711200042012@10.0.0.2
CSeq: 2 INVITE
Proxy-Authorization: Digest username="1008",realm="asterisk",nonce="05560f2b",uri="sip:02452155048@10.0.0.1",algorithm=MD5,response="b26489cd71bf709ba926cb989a6584cd"
Contact: <sip:1008@10.0.0.2>
Supported: em,100rel,timer,replaces,path,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Audiocodes-Sip-Gateway-MP-124 FXS/v.5.00A.027.005
Content-Type: application/sdp
Content-Length: 272
v=0
o=AudiocodesGW 239629226 239629145 IN IP4 10.0.0.2
s=Phone-Call
c=IN IP4 10.0.0.2
t=0 0
m=audio 6070 RTP/AVP 18 96
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=ptime:20
a=sendrecv
a=rtcp:6071 IN IP4 10.0.0.2
--- (14 headers 13 lines) ---
Using INVITE request as basis request - 23963575711200042012@10.0.0.2
Sending to 10.0.0.2 : 5060 (non-NAT)
Found user '1008'
Found RTP audio format 18
Found RTP audio format 96
Peer audio RTP is at port 10.0.0.2:6070
Found description format G729
Found description format telephone-event
Capabilities: us - 0x50e (gsm|ulaw|alaw|g729|ilbc), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 02452155048 in masus (domain 10.0.0.1;user=phone)
list_route: hop: <sip:1008@10.0.0.2>
Transmitting (no NAT) to 10.0.0.2:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.0.2;branch=z9hG4bKac239664674;received=10.0.0.2
From: <sip:1008@10.0.0.1>;tag=1c239635997
To: <sip:02452155048@10.0.0.1;user=phone>
Call-ID: 23963575711200042012@10.0.0.2
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:02452155048@10.0.0.1>
Content-Length: 0
---
We're at 192.168.2.15 port 5638
Adding codec 0x100 (g729) to SDP
Adding codec 0x400 (ilbc) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 15 lines
Reliably Transmitting (NAT) to 83.125.8.46:5060:
INVITE sip:00492452155048@voip.dus.net SIP/2.0
Via: SIP/2.0/UDP 192.168.2.15:5060;branch=z9hG4bK22c4e836;rport
From: "1008" <sip:000387233745@192.168.2.15>;tag=as26adf52c
To: <sip:00492452155048@voip.dus.net>
Contact: <sip:000387233745@192.168.2.15>
Call-ID: 262ccf6115d4bbb4467624dc0262a62f@192.168.2.15
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 22 Apr 2008 15:39:10 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 333
v=0
o=root 8992 8992 IN IP4 192.168.2.15
s=session
c=IN IP4 192.168.2.15
t=0 0
m=audio 5638 RTP/AVP 18 97 3 0 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:97 iLBC/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
pbx*CLI>
<-- SIP read from 83.125.8.46:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.2.15:5060;branch=z9hG4bK22c4e836;received=81.214.188.91;rport=5060
From: "1008" <sip:000387233745@192.168.2.15>;tag=as26adf52c
To: <sip:00492452155048@voip.dus.net>;tag=as2760b903
Call-ID: 262ccf6115d4bbb4467624dc0262a62f@192.168.2.15
CSeq: 102 INVITE
User-Agent: SIP_GW_1.4.2.PL3 talos.dus.net
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="dus.net", nonce="7cff320c"
Content-Length: 0
--- (11 headers 0 lines) ---
Transmitting (NAT) to 83.125.8.46:5060:
ACK sip:00492452155048@voip.dus.net SIP/2.0
Via: SIP/2.0/UDP 192.168.2.15:5060;branch=z9hG4bK22c4e836;rport
From: "1008" <sip:000387233745@192.168.2.15>;tag=as26adf52c
To: <sip:00492452155048@voip.dus.net>;tag=as2760b903
Contact: <sip:000387233745@192.168.2.15>
Call-ID: 262ccf6115d4bbb4467624dc0262a62f@192.168.2.15
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
We're at 192.168.2.15 port 5638
Adding codec 0x100 (g729) to SDP
Adding codec 0x400 (ilbc) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 83.125.8.46:5060:
INVITE sip:00492452155048@voip.dus.net SIP/2.0
Via: SIP/2.0/UDP 192.168.2.15:5060;branch=z9hG4bK2ff645d4;rport
From: "1008" <sip:000387233745@192.168.2.15>;tag=as26adf52c
To: <sip:00492452155048@voip.dus.net>
Contact: <sip:000387233745@192.168.2.15>
Call-ID: 262ccf6115d4bbb4467624dc0262a62f@192.168.2.15
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Proxy-Authorization: Digest username="000387233745", realm="dus.net", algorithm=MD5, uri="sip:00492452155048@voip.dus.net", nonce="7cff320c", response="1d30e905cb4367b250e3119ea85849e3", opaque=""
Date: Tue, 22 Apr 2008 15:39:10 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
ontent-Type: application/sdp
Content-Length: 333
v=0
o=root 8992 8993 IN IP4 192.168.2.15
s=session
c=IN IP4 192.168.2.15
t=0 0
m=audio 5638 RTP/AVP 18 97 3 0 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:97 iLBC/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
pbx*CLI>
<-- SIP read from 83.125.8.46:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.15:5060;branch=z9hG4bK2ff645d4;received=81.214.188.91;rport=5060
From: "1008" <sip:000387233745@192.168.2.15>;tag=as26adf52c
To: <sip:00492452155048@voip.dus.net>
Call-ID: 262ccf6115d4bbb4467624dc0262a62f@192.168.2.15
CSeq: 103 INVITE
User-Agent: SIP_GW_1.4.2.PL3 talos.dus.net
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:00492452155048@83.125.8.46>
Content-Length: 0
--- (11 headers 0 lines) ---
pbx*CLI>
<-- SIP read from 10.0.0.2:5060:
BYE sip:050519159871@10.0.0.1 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.2;branch=z9hG4bKac239998965
Max-Forwards: 70
From: <sip:1001@10.0.0.1>;tag=1c224543595
To: <sip:050519159871@10.0.0.1;user=phone>;tag=as3b3a07ab
Call-ID: 2245433591120004200@10.0.0.2
CSeq: 3 BYE
Supported: em,timer,replaces,path,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Audiocodes-Sip-Gateway-MP-124 FXS/v.5.00A.027.005
Reason: Q.850 ;cause=16
Content-Length: 0
--- (12 headers 0 lines) ---
Sending to 10.0.0.2 : 5060 (non-NAT)
Transmitting (no NAT) to 10.0.0.2:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.2;branch=z9hG4bKac239998965;received=10.0.0.2
From: <sip:1001@10.0.0.1>;tag=1c224543595
To: <sip:050519159871@10.0.0.1;user=phone>;tag=as3b3a07ab
Call-ID: 2245433591120004200@10.0.0.2
CSeq: 3 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:050519159871@10.0.0.1>
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing
---
Scheduling destruction of call '0b760e162c18bdc90283a21f38126586@192.168.2.15' in 32000 ms
set_destination: Parsing <sip:004950519159871@83.125.8.46> for address/port to send to
set_destination: set destination to 83.125.8.46, port 5060
Reliably Transmitting (NAT) to 83.125.8.46:5060:
BYE sip:004950519159871@83.125.8.46 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.15:5060;branch=z9hG4bK0e08051e;rport
From: "1001" <sip:000387233745@192.168.2.15>;tag=as1387477f
To: <sip:004950519159871@voip.dus.net>;tag=as3c1e61a9
Call-ID: 0b760e162c18bdc90283a21f38126586@192.168.2.15
CSeq: 104 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Proxy-Authorization: Digest username="000387233745", realm="dus.net", algorithm=MD5, uri="sip:004950519159871@83.125.8.46", nonce="389ba3cb", response="4065d10f74da0932830075057affa1a8", opaque=""
Content-Length: 0
---
pbx*CLI>
<-- SIP read from 83.125.8.46:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.15:5060;branch=z9hG4bK0e08051e;received=81.214.188.91;rport=5060
From: "1001" <sip:000387233745@192.168.2.15>;tag=as1387477f
To: <sip:004950519159871@voip.dus.net>;tag=as3c1e61a9
Call-ID: 0b760e162c18bdc90283a21f38126586@192.168.2.15
CSeq: 104 BYE
User-Agent: SIP_GW_1.4.2.PL3 talos.dus.net
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:004950519159871@83.125.8.46>
Content-Length: 0
--- (11 headers 0 lines) ---
Destroying call '0b760e162c18bdc90283a21f38126586@192.168.2.15'
Destroying call '2245433591120004200@10.0.0.2'
pbx*CLI>
<-- SIP read from 83.125.8.46:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.15:5060;branch=z9hG4bK6ec06309;received=81.214.188.91;rport=5060
From: "1010" <sip:000387233745@192.168.2.15>;tag=as018eec44
To: <sip:0049226157403@voip.dus.net>;tag=as66795132
Call-ID: 0adee35a7e61c7d10bffba995ec07c88@192.168.2.15
CSeq: 103 INVITE
User-Agent: SIP_GW_1.4.2.PL3 talos.dus.net
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:0049226157403@83.125.8.46>
ontent-Type: application/sdp
Content-Length: 351
v=0
o=root 1387 1388 IN IP4 83.125.8.46
s=session
c=IN IP4 83.125.8.46
t=0 0
m=audio 16736 RTP/AVP 8 3 97 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
--- (12 headers 17 lines) ---
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 97
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 83.125.8.46:16736
Found description format PCMA
Found description format GSM
Found description format iLBC
Found description format G729
Found description format telephone-event
Capabilities: us - 0x50e (gsm|ulaw|alaw|g729|ilbc), peer - audio=0x50a (gsm|alaw|g729|ilbc)/video=0x0 (nothing), combined - 0x50a (gsm|alaw|g729|ilbc)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
list_route: hop: <sip:0049226157403@83.125.8.46>
set_destination: Parsing <sip:0049226157403@83.125.8.46> for address/port to send to
set_destination: set destination to 83.125.8.46, port 5060
Transmitting (NAT) to 83.125.8.46:5060:
ACK sip:0049226157403@83.125.8.46 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.15:5060;branch=z9hG4bK58f07536;rport
From: "1010" <sip:000387233745@192.168.2.15>;tag=as018eec44
To: <sip:0049226157403@voip.dus.net>;tag=as66795132
Contact: <sip:000387233745@192.168.2.15>
Call-ID: 0adee35a7e61c7d10bffba995ec07c88@192.168.2.15
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
We're at 10.0.0.1 port 26290
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.0.0.2:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.2;branch=z9hG4bKac223785312;received=10.0.0.2
From: <sip:1010@10.0.0.1>;tag=1c223759377
To: <sip:0226157403@10.0.0.1;user=phone>;tag=as57837876
Call-ID: 22375913911200041959@10.0.0.2
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:0226157403@10.0.0.1>
Content-Type: application/sdp
Content-Length: 226
v=0
o=root 8992 8993 IN IP4 10.0.0.1
s=session
c=IN IP4 10.0.0.1
t=0 0
m=audio 26290 RTP/AVP 18 96
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=silenceSupp:off - - - -
---
pbx*CLI>
<-- SIP read from 10.0.0.2:5060:
ACK sip:0226157403@10.0.0.1 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.2;branch=z9hG4bKac241725686
Max-Forwards: 70
From: <sip:1010@10.0.0.1>;tag=1c223759377
To: <sip:0226157403@10.0.0.1;user=phone>;tag=as57837876
Call-ID: 22375913911200041959@10.0.0.2
CSeq: 2 ACK
Contact: <sip:1010@10.0.0.2>
Supported: em,timer,replaces,path,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Audiocodes-Sip-Gateway-MP-124 FXS/v.5.00A.027.005
Content-Length: 0
--- (12 headers 0 lines) ---
pbx*CLI>
<-- SIP read from 83.125.8.46:5060:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.2.15:5060;branch=z9hG4bK2ff645d4;received=81.214.188.91;rport=5060
From: "1008" <sip:000387233745@192.168.2.15>;tag=as26adf52c
To: <sip:00492452155048@voip.dus.net>;tag=as53cf50c2
Call-ID: 262ccf6115d4bbb4467624dc0262a62f@192.168.2.15
CSeq: 103 INVITE
User-Agent: SIP_GW_1.4.2.PL3 talos.dus.net
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:00492452155048@83.125.8.46>
Content-Type: application/sdp
Content-Length: 351
v=0
o=root 1387 1387 IN IP4 83.125.8.46
s=session
c=IN IP4 83.125.8.46
t=0 0
m=audio 24116 RTP/AVP 8 3 97 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
--- (12 headers 17 lines) ---
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 97
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 83.125.8.46:24116
Found description format PCMA
Found description format GSM
Found description format iLBC
Found description format G729
Found description format telephone-event
Capabilities: us - 0x50e (gsm|ulaw|alaw|g729|ilbc), peer - audio=0x50a (gsm|alaw|g729|ilbc)/video=0x0 (nothing), combined - 0x50a (gsm|alaw|g729|ilbc)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
We're at 10.0.0.1 port 11754
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Transmitting (no NAT) to 10.0.0.2:5060:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.0.0.2;branch=z9hG4bKac239664674;received=10.0.0.2
From: <sip:1008@10.0.0.1>;tag=1c239635997
To: <sip:02452155048@10.0.0.1;user=phone>;tag=as07d9b14a
Call-ID: 23963575711200042012@10.0.0.2
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:02452155048@10.0.0.1>
Content-Type: application/sdp
Content-Length: 226
v=0
o=root 8992 8992 IN IP4 10.0.0.1
s=session
c=IN IP4 10.0.0.1
t=0 0
m=audio 11754 RTP/AVP 18 96
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=silenceSupp:off - - - -
---
pbx*CLI>
<-- SIP read from 10.0.0.2:5060:
INVITE sip:02771819382@10.0.0.1;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.0.0.2;branch=z9hG4bKac244172515
Max-Forwards: 70
From: <sip:1002@10.0.0.1>;tag=1c244170773
To: <sip:02771819382@10.0.0.1;user=phone>
Call-ID: 24417053211200042016@10.0.0.2
CSeq: 1 INVITE
Contact: <sip:1002@10.0.0.2>
Supported: em,100rel,timer,replaces,path,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Audiocodes-Sip-Gateway-MP-124 FXS/v.5.00A.027.005
Content-Type: application/sdp
Content-Length: 272
v=0
o=AudiocodesGW 244164123 244164040 IN IP4 10.0.0.2
s=Phone-Call
c=IN IP4 10.0.0.2
t=0 0
m=audio 6010 RTP/AVP 18 96
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=ptime:20
a=sendrecv
a=rtcp:6011 IN IP4 10.0.0.2
--- (13 headers 13 lines) ---
Using INVITE request as basis request - 24417053211200042016@10.0.0.2
Sending to 10.0.0.2 : 5060 (non-NAT)
Reliably Transmitting (no NAT) to 10.0.0.2:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.0.0.2;branch=z9hG4bKac244172515;received=10.0.0.2
From: <sip:1002@10.0.0.1>;tag=1c244170773
To: <sip:02771819382@10.0.0.1;user=phone>;tag=as1dc1fe85
Call-ID: 24417053211200042016@10.0.0.2
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4dbc9192"
Content-Length: 0
---
Scheduling destruction of call '24417053211200042016@10.0.0.2' in 15000 ms
Found user '1002'
pbx*CLI>
<-- SIP read from 10.0.0.2:5060:
ACK sip:02771819382@10.0.0.1;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.0.0.2;branch=z9hG4bKac244172515
Max-Forwards: 70
From: <sip:1002@10.0.0.1>;tag=1c244170773
To: <sip:02771819382@10.0.0.1;user=phone>;tag=as1dc1fe85
Call-ID: 24417053211200042016@10.0.0.2
CSeq: 1 ACK
Contact: <sip:1002@10.0.0.2>
Supported: em,timer,replaces,path,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Audiocodes-Sip-Gateway-MP-124 FXS/v.5.00A.027.005
Content-Length: 0
--- (12 headers 0 lines) ---
pbx*CLI>
<-- SIP read from 10.0.0.2:5060:
INVITE sip:02771819382@10.0.0.1;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.0.0.2;branch=z9hG4bKac244195942
Max-Forwards: 70
From: <sip:1002@10.0.0.1>;tag=1c244170773
To: <sip:02771819382@10.0.0.1;user=phone>
Call-ID: 24417053211200042016@10.0.0.2
CSeq: 2 INVITE
Proxy-Authorization: Digest username="1002",realm="asterisk",nonce="4dbc9192",uri="sip:02771819382@10.0.0.1",algorithm=MD5,response="92bc14b5bb273b85e539bebf71a494e5"
Contact: <sip:1002@10.0.0.2>
Supported: em,100rel,timer,replaces,path,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Audiocodes-Sip-Gateway-MP-124 FXS/v.5.00A.027.005
Content-Type: application/sdp
Content-Length: 272
v=0
o=AudiocodesGW 244164123 244164040 IN IP4 10.0.0.2
s=Phone-Call
c=IN IP4 10.0.0.2
t=0 0
m=audio 6010 RTP/AVP 18 96
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=ptime:20
a=sendrecv
a=rtcp:6011 IN IP4 10.0.0.2
--- (14 headers 13 lines) ---
Using INVITE request as basis request - 24417053211200042016@10.0.0.2
Sending to 10.0.0.2 : 5060 (non-NAT)
Found user '1002'
Found RTP audio format 18
Found RTP audio format 96
Peer audio RTP is at port 10.0.0.2:6010
Found description format G729
Found description format telephone-event
Capabilities: us - 0x50e (gsm|ulaw|alaw|g729|ilbc), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 02771819382 in masus (domain 10.0.0.1;user=phone)
list_route: hop: <sip:1002@10.0.0.2>
Transmitting (no NAT) to 10.0.0.2:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.0.2;branch=z9hG4bKac244195942;received=10.0.0.2
From: <sip:1002@10.0.0.1>;tag=1c244170773
To: <sip:02771819382@10.0.0.1;user=phone>
Call-ID: 24417053211200042016@10.0.0.2
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:02771819382@10.0.0.1>
Content-Length: 0
---
We're at 192.168.2.15 port 24582
Adding codec 0x100 (g729) to SDP
Adding codec 0x400 (ilbc) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 15 lines
Reliably Transmitting (NAT) to 83.125.8.46:5060:
INVITE sip:00492771819382@voip.dus.net SIP/2.0
Via: SIP/2.0/UDP 192.168.2.15:5060;branch=z9hG4bK1dffbf90;rport
From: "1002" <sip:000387233745@192.168.2.15>;tag=as495acd44
To: <sip:00492771819382@voip.dus.net>
Contact: <sip:000387233745@192.168.2.15>
Call-ID: 00d9911a36d325e378cc75451986412f@192.168.2.15
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 22 Apr 2008 15:39:14 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 334
v=0
o=root 8992 8992 IN IP4 192.168.2.15
s=session
c=IN IP4 192.168.2.15
t=0 0
m=audio 24582 RTP/AVP 18 97 3 0 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:97 iLBC/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
pbx*CLI>
<-- SIP read from 83.125.8.46:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.2.15:5060;branch=z9hG4bK1dffbf90;received=81.214.188.91;rport=5060
From: "1002" <sip:000387233745@192.168.2.15>;tag=as495acd44
To: <sip:00492771819382@voip.dus.net>;tag=as33330631
Call-ID: 00d9911a36d325e378cc75451986412f@192.168.2.15
CSeq: 102 INVITE
User-Agent: SIP_GW_1.4.2.PL3 talos.dus.net
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="dus.net", nonce="1e752fb2"
Content-Length: 0
--- (11 headers 0 lines) ---
Transmitting (NAT) to 83.125.8.46:5060:
ACK sip:00492771819382@voip.dus.net SIP/2.0
Via: SIP/2.0/UDP 192.168.2.15:5060;branch=z9hG4bK1dffbf90;rport
From: "1002" <sip:000387233745@192.168.2.15>;tag=as495acd44
To: <sip:00492771819382@voip.dus.net>;tag=as33330631
Contact: <sip:000387233745@192.168.2.15>
Call-ID: 00d9911a36d325e378cc75451986412f@192.168.2.15
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
We're at 192.168.2.15 port 24582
Adding codec 0x100 (g729) to SDP
Adding codec 0x400 (ilbc) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 83.125.8.46:5060:
INVITE sip:00492771819382@voip.dus.net SIP/2.0
Via: SIP/2.0/UDP 192.168.2.15:5060;branch=z9hG4bK5cf7c4c2;rport
From: "1002" <sip:000387233745@192.168.2.15>;tag=as495acd44
To: <sip:00492771819382@voip.dus.net>
Contact: <sip:000387233745@192.168.2.15>
Call-ID: 00d9911a36d325e378cc75451986412f@192.168.2.15
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Proxy-Authorization: Digest username="000387233745", realm="dus.net", algorithm=MD5, uri="sip:00492771819382@voip.dus.net", nonce="1e752fb2", response="f7059c8df9a5908bf0bd0501757ed16b", opaque=""
Date: Tue, 22 Apr 2008 15:39:14 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 334
v=0
o=root 8992 8993 IN IP4 192.168.2.15
s=session
c=IN IP4 192.168.2.15
t=0 0
m=audio 24582 RTP/AVP 18 97 3 0 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:97 iLBC/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
pbx*CLI>
<-- SIP read from 83.125.8.46:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.15:5060;branch=z9hG4bK5cf7c4c2;received=81.214.188.91;rport=5060
From: "1002" <sip:000387233745@192.168.2.15>;tag=as495acd44
To: <sip:00492771819382@voip.dus.net>
Call-ID: 00d9911a36d325e378cc75451986412f@192.168.2.15
CSeq: 103 INVITE
User-Agent: SIP_GW_1.4.2.PL3 talos.dus.net
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:00492771819382@83.125.8.46>
Content-Length: 0
--- (11 headers 0 lines) ---
pbx*CLI>
<-- SIP read from 83.125.8.46:5060:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.2.15:5060;branch=z9hG4bK5cf7c4c2;received=81.214.188.91;rport=5060
From: "1002" <sip:000387233745@192.168.2.15>;tag=as495acd44
To: <sip:00492771819382@voip.dus.net>;tag=as1a8e074b
Call-ID: 00d9911a36d325e378cc75451986412f@192.168.2.15
CSeq: 103 INVITE
User-Agent: SIP_GW_1.4.2.PL3 talos.dus.net
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:00492771819382@83.125.8.46>
Content-Type: application/sdp
Content-Length: 350
v=0
o=root 1387 1387 IN IP4 83.125.8.46
s=session
c=IN IP4 83.125.8.46
t=0 0
m=audio 5948 RTP/AVP 8 3 97 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
--- (12 headers 17 lines) ---
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 97
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 83.125.8.46:5948
Found description format PCMA
Found description format GSM
Found description format iLBC
Found description format G729
Found description format telephone-event
Capabilities: us - 0x50e (gsm|ulaw|alaw|g729|ilbc), peer - audio=0x50a (gsm|alaw|g729|ilbc)/video=0x0 (nothing), combined - 0x50a (gsm|alaw|g729|ilbc)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
We're at 10.0.0.1 port 25312
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Transmitting (no NAT) to 10.0.0.2:5060:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.0.0.2;branch=z9hG4bKac244195942;received=10.0.0.2
From: <sip:1002@10.0.0.1>;tag=1c244170773
To: <sip:02771819382@10.0.0.1;user=phone>;tag=as3430dbc1
Call-ID: 24417053211200042016@10.0.0.2
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:02771819382@10.0.0.1>
Content-Type: application/sdp
Content-Length: 226
v=0
o=root 8992 8992 IN IP4 10.0.0.1
s=session
c=IN IP4 10.0.0.1
t=0 0
m=audio 25312 RTP/AVP 18 96
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=silenceSupp:off - - - -
---
pbx*CLI>
<-- SIP read from 10.0.0.2:5060:
INVITE sip:0545173819@10.0.0.1;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.0.0.2;branch=z9hG4bKac249298169
Max-Forwards: 70
From: <sip:1004@10.0.0.1>;tag=1c249296417
To: <sip:0545173819@10.0.0.1;user=phone>
Call-ID: 24929617611200042020@10.0.0.2
CSeq: 1 INVITE
Contact: <sip:1004@10.0.0.2>
Supported: em,100rel,timer,replaces,path,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Audiocodes-Sip-Gateway-MP-124 FXS/v.5.00A.027.005
Content-Type: application/sdp
Content-Length: 272
v=0
o=AudiocodesGW 249289592 249289509 IN IP4 10.0.0.2
s=Phone-Call
c=IN IP4 10.0.0.2
t=0 0
m=audio 6030 RTP/AVP 18 96
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=ptime:20
a=sendrecv
a=rtcp:6031 IN IP4 10.0.0.2
--- (13 headers 13 lines) ---
Using INVITE request as basis request - 24929617611200042020@10.0.0.2
Sending to 10.0.0.2 : 5060 (non-NAT)
Reliably Transmitting (no NAT) to 10.0.0.2:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.0.0.2;branch=z9hG4bKac249298169;received=10.0.0.2
From: <sip:1004@10.0.0.1>;tag=1c249296417
To: <sip:0545173819@10.0.0.1;user=phone>;tag=as2f5000a2
Call-ID: 24929617611200042020@10.0.0.2
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6c97d623"
Content-Length: 0
---
Scheduling destruction of call '24929617611200042020@10.0.0.2' in 15000 ms
Found user '1004'
pbx*CLI>
<-- SIP read from 10.0.0.2:5060:
ACK sip:0545173819@10.0.0.1;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.0.0.2;branch=z9hG4bKac249298169
Max-Forwards: 70
From: <sip:1004@10.0.0.1>;tag=1c249296417
To: <sip:0545173819@10.0.0.1;user=phone>;tag=as2f5000a2
Call-ID: 24929617611200042020@10.0.0.2
CSeq: 1 ACK
Contact: <sip:1004@10.0.0.2>
Supported: em,timer,replaces,path,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Audiocodes-Sip-Gateway-MP-124 FXS/v.5.00A.027.005
Content-Length: 0
--- (12 headers 0 lines) ---
pbx*CLI>
<-- SIP read from 10.0.0.2:5060:
INVITE sip:0545173819@10.0.0.1;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.0.0.2;branch=z9hG4bKac249321137
Max-Forwards: 70
From: <sip:1004@10.0.0.1>;tag=1c249296417
To: <sip:0545173819@10.0.0.1;user=phone>
Call-ID: 24929617611200042020@10.0.0.2
CSeq: 2 INVITE
Proxy-Authorization: Digest username="1004",realm="asterisk",nonce="6c97d623",uri="sip:0545173819@10.0.0.1",algorithm=MD5,response="44a88de12c2cf00002e7ec68784557fe"
Contact: <sip:1004@10.0.0.2>
Supported: em,100rel,timer,replaces,path,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Audiocodes-Sip-Gateway-MP-124 FXS/v.5.00A.027.005
Content-Type: application/sdp
Content-Length: 272
v=0
o=AudiocodesGW 249289592 249289509 IN IP4 10.0.0.2
s=Phone-Call
c=IN IP4 10.0.0.2
t=0 0
m=audio 6030 RTP/AVP 18 96
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=ptime:20
a=sendrecv
a=rtcp:6031 IN IP4 10.0.0.2
--- (14 headers 13 lines) ---
Using INVITE request as basis request - 24929617611200042020@10.0.0.2
Sending to 10.0.0.2 : 5060 (non-NAT)
Found user '1004'
Found RTP audio format 18
Found RTP audio format 96
Peer audio RTP is at port 10.0.0.2:6030
Found description format G729
Found description format telephone-event
Capabilities: us - 0x50e (gsm|ulaw|alaw|g729|ilbc), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 0545173819 in masus (domain 10.0.0.1;user=phone)
list_route: hop: <sip:1004@10.0.0.2>
Transmitting (no NAT) to 10.0.0.2:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.0.2;branch=z9hG4bKac249321137;received=10.0.0.2
From: <sip:1004@10.0.0.1>;tag=1c249296417
To: <sip:0545173819@10.0.0.1;user=phone>
Call-ID: 24929617611200042020@10.0.0.2
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:0545173819@10.0.0.1>
Content-Length: 0
---
We're at 192.168.2.15 port 19604
Adding codec 0x100 (g729) to SDP
Adding codec 0x400 (ilbc) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 15 lines
Reliably Transmitting (NAT) to 83.125.8.46:5060:
INVITE sip:0049545173819@voip.dus.net SIP/2.0
Via: SIP/2.0/UDP 192.168.2.15:5060;branch=z9hG4bK100cd22e;rport
From: "1004" <sip:000387233745@192.168.2.15>;tag=as284f8fbf
To: <sip:0049545173819@voip.dus.net>
Contact: <sip:000387233745@192.168.2.15>
Call-ID: 50e49713563aa63e5e1457881e006e58@192.168.2.15
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 22 Apr 2008 15:39:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 334
v=0
o=root 8992 8992 IN IP4 192.168.2.15
s=session
c=IN IP4 192.168.2.15
t=0 0
m=audio 19604 RTP/AVP 18 97 3 0 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:97 iLBC/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
pbx*CLI>
<-- SIP read from 83.125.8.46:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.2.15:5060;branch=z9hG4bK100cd22e;received=81.214.188.91;rport=5060
From: "1004" <sip:000387233745@192.168.2.15>;tag=as284f8fbf
To: <sip:0049545173819@voip.dus.net>;tag=as65e54448
Call-ID: 50e49713563aa63e5e1457881e006e58@192.168.2.15
CSeq: 102 INVITE
User-Agent: SIP_GW_1.4.2.PL3 talos.dus.net
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="dus.net", nonce="562cd475"
Content-Length: 0
--- (11 headers 0 lines) ---
Transmitting (NAT) to 83.125.8.46:5060:
ACK sip:0049545173819@voip.dus.net SIP/2.0
Via: SIP/2.0/UDP 192.168.2.15:5060;branch=z9hG4bK100cd22e;rport
From: "1004" <sip:000387233745@192.168.2.15>;tag=as284f8fbf
To: <sip:0049545173819@voip.dus.net>;tag=as65e54448
Contact: <sip:000387233745@192.168.2.15>
Call-ID: 50e49713563aa63e5e1457881e006e58@192.168.2.15
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
We're at 192.168.2.15 port 19604
Adding codec 0x100 (g729) to SDP
Adding codec 0x400 (ilbc) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 83.125.8.46:5060:
INVITE sip:0049545173819@voip.dus.net SIP/2.0
Via: SIP/2.0/UDP 192.168.2.15:5060;branch=z9hG4bK727b201f;rport
From: "1004" <sip:000387233745@192.168.2.15>;tag=as284f8fbf
To: <sip:0049545173819@voip.dus.net>
Contact: <sip:000387233745@192.168.2.15>
Call-ID: 50e49713563aa63e5e1457881e006e58@192.168.2.15
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Proxy-Authorization: Digest username="000387233745", realm="dus.net", algorithm=MD5, uri="sip:0049545173819@voip.dus.net", nonce="562cd475", response="6616af67c44205bf87bd1504f6699fc0", opaque=""
Date: Tue, 22 Apr 2008 15:39:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 334
v=0
o=root 8992 8993 IN IP4 192.168.2.15
s=session
c=IN IP4 192.168.2.15
t=0 0
m=audio 19604 RTP/AVP 18 97 3 0 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:97 iLBC/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
pbx*CLI>
<-- SIP read from 83.125.8.46:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.15:5060;branch=z9hG4bK727b201f;received=81.214.188.91;rport=5060
From: "1004" <sip:000387233745@192.168.2.15>;tag=as284f8fbf
To: <sip:0049545173819@voip.dus.net>
Call-ID: 50e49713563aa63e5e1457881e006e58@192.168.2.15
CSeq: 103 INVITE
User-Agent: SIP_GW_1.4.2.PL3 talos.dus.net
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:0049545173819@83.125.8.46>
Content-Length: 0
--- (11 headers 0 lines) ---
pbx*CLI>
<-- SIP read from 83.125.8.46:5060:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.2.15:5060;branch=z9hG4bK727b201f;received=81.214.188.91;rport=5060
From: "1004" <sip:000387233745@192.168.2.15>;tag=as284f8fbf
To: <sip:0049545173819@voip.dus.net>;tag=as68d70d13
Call-ID: 50e49713563aa63e5e1457881e006e58@192.168.2.15
CSeq: 103 INVITE
User-Agent: SIP_GW_1.4.2.PL3 talos.dus.net
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:0049545173819@83.125.8.46>
Content-Type: application/sdp
Content-Length: 351
v=0
o=root 1387 1387 IN IP4 83.125.8.46
s=session
c=IN IP4 83.125.8.46
t=0 0
m=audio 10584 RTP/AVP 8 3 97 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
--- (12 headers 17 lines) ---
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 97
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 83.125.8.46:10584
Found description format PCMA
Found description format GSM
Found description format iLBC
Found description format G729
Found description format telephone-event
Capabilities: us - 0x50e (gsm|ulaw|alaw|g729|ilbc), peer - audio=0x50a (gsm|alaw|g729|ilbc)/video=0x0 (nothing), combined - 0x50a (gsm|alaw|g729|ilbc)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
We're at 10.0.0.1 port 21060
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Transmitting (no NAT) to 10.0.0.2:5060:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.0.0.2;branch=z9hG4bKac249321137;received=10.0.0.2
From: <sip:1004@10.0.0.1>;tag=1c249296417
To: <sip:0545173819@10.0.0.1;user=phone>;tag=as147d1b15
Call-ID: 24929617611200042020@10.0.0.2
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:0545173819@10.0.0.1>
Content-Type: application/sdp
Content-Length: 226
v=0
o=root 8992 8992 IN IP4 10.0.0.1
s=session
c=IN IP4 10.0.0.1
t=0 0
m=audio 21060 RTP/AVP 18 96
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=silenceSupp:off - - - -
---
pbx*CLI>
<-- SIP read from 83.125.8.46:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.15:5060;branch=z9hG4bK2ff645d4;received=81.214.188.91;rport=5060
From: "1008" <sip:000387233745@192.168.2.15>;tag=as26adf52c
To: <sip:00492452155048@voip.dus.net>;tag=as53cf50c2
Call-ID: 262ccf6115d4bbb4467624dc0262a62f@192.168.2.15
CSeq: 103 INVITE
User-Agent: SIP_GW_1.4.2.PL3 talos.dus.net
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:00492452155048@83.125.8.46>
Content-Type: application/sdp
Content-Length: 351
v=0
o=root 1387 1388 IN IP4 83.125.8.46
s=session
c=IN IP4 83.125.8.46
t=0 0
m=audio 24116 RTP/AVP 8 3 97 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
--- (12 headers 17 lines) ---
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 97
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 83.125.8.46:24116
Found description format PCMA
Found description format GSM
Found description format iLBC
Found description format G729
Found description format telephone-event
Capabilities: us - 0x50e (gsm|ulaw|alaw|g729|ilbc), peer - audio=0x50a (gsm|alaw|g729|ilbc)/video=0x0 (nothing), combined - 0x50a (gsm|alaw|g729|ilbc)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
list_route: hop: <sip:00492452155048@83.125.8.46>
set_destination: Parsing <sip:00492452155048@83.125.8.46> for address/port to send to
set_destination: set destination to 83.125.8.46, port 5060
Transmitting (NAT) to 83.125.8.46:5060:
ACK sip:00492452155048@83.125.8.46 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.15:5060;branch=z9hG4bK6827d41b;rport
From: "1008" <sip:000387233745@192.168.2.15>;tag=as26adf52c
To: <sip:00492452155048@voip.dus.net>;tag=as53cf50c2
Contact: <sip:000387233745@192.168.2.15>
Call-ID: 262ccf6115d4bbb4467624dc0262a62f@192.168.2.15
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
We're at 10.0.0.1 port 11754
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.0.0.2:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.2;branch=z9hG4bKac239664674;received=10.0.0.2
From: <sip:1008@10.0.0.1>;tag=1c239635997
To: <sip:02452155048@10.0.0.1;user=phone>;tag=as07d9b14a
Call-ID: 23963575711200042012@10.0.0.2
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:02452155048@10.0.0.1>
Content-Type: application/sdp
Content-Length: 226
v=0
o=root 8992 8993 IN IP4 10.0.0.1
s=session
c=IN IP4 10.0.0.1
t=0 0
m=audio 11754 RTP/AVP 18 96
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=silenceSupp:off - - - -
---
pbx*CLI> s
<-- SIP read from 10.0.0.2:5060:
ACK sip:02452155048@10.0.0.1 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.2;branch=z9hG4bKac256916755
Max-Forwards: 70
From: <sip:1008@10.0.0.1>;tag=1c239635997
To: <sip:02452155048@10.0.0.1;user=phone>;tag=as07d9b14a
Call-ID: 23963575711200042012@10.0.0.2
CSeq: 2 ACK
Contact: <sip:1008@10.0.0.2>
Supported: em,timer,replaces,path,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Audiocodes-Sip-Gateway-MP-124 FXS/v.5.00A.027.005
Content-Length: 0
--- (12 headers 0 lines) ---
pbx*CLI> sip deb
<-- SIP read from 10.0.0.2:5060:
CANCEL sip:071311298862@10.0.0.1;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.0.0.2;branch=z9hG4bKac236789509
Max-Forwards: 70
From: <sip:1024@10.0.0.1>;tag=1c236760662
To: <sip:071311298862@10.0.0.1;user=phone>
Call-ID: 23676042311200042010@10.0.0.2
CSeq: 2 CANCEL
Supported: em,timer,replaces,path,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Audiocodes-Sip-Gateway-MP-124 FXS/v.5.00A.027.005
Reason: Q.850 ;cause=16
Content-Length: 0
--- (12 headers 0 lines) ---
Sending to 10.0.0.2 : 5060 (non-NAT)
Reliably Transmitting (no NAT) to 10.0.0.2:5060:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 10.0.0.2;branch=z9hG4bKac236789509;received=10.0.0.2
From: <sip:1024@10.0.0.1>;tag=1c236760662
To: <sip:071311298862@10.0.0.1;user=phone>;tag=as35f994c9
Call-ID: 23676042311200042010@10.0.0.2
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0
---
Transmitting (no NAT) to 10.0.0.2:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.2;branch=z9hG4bKac236789509;received=10.0.0.2
From: <sip:1024@10.0.0.1>;tag=1c236760662
To: <sip:071311298862@10.0.0.1;user=phone>;tag=as35f994c9
Call-ID: 23676042311200042010@10.0.0.2
CSeq: 2 CANCEL
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:071311298862@10.0.0.1>
Content-Length: 0
---
Scheduling destruction of call '70e04fe371dd539c6cab3d4627e544f2@192.168.2.15' in 32000 ms
Reliably Transmitting (NAT) to 83.125.8.46:5060:
CANCEL sip:004971311298862@voip.dus.net SIP/2.0
Via: SIP/2.0/UDP 192.168.2.15:5060;branch=z9hG4bK501ab53e;rport
From: "1024" <sip:000387233745@192.168.2.15>;tag=as20b9cbea
To: <sip:004971311298862@voip.dus.net>
Call-ID: 70e04fe371dd539c6cab3d4627e544f2@192.168.2.15
CSeq: 103 CANCEL
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
pbx*CLI> sip deb
<-- SIP read from 10.0.0.2:5060:
ACK sip:071311298862@10.0.0.1;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.0.0.2;branch=z9hG4bKac236789509
Max-Forwards: 70
From: <sip:1024@10.0.0.1>;tag=1c236760662
To: <sip:071311298862@10.0.0.1;user=phone>;tag=as35f994c9
Call-ID: 23676042311200042010@10.0.0.2
CSeq: 2 ACK
Contact: <sip:1024@10.0.0.2>
Supported: em,timer,replaces,path,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Audiocodes-Sip-Gateway-MP-124 FXS/v.5.00A.027.005
Content-Length: 0
--- (12 headers 0 lines) ---
Destroying call '23676042311200042010@10.0.0.2'
pbx*CLI> sip debu
<-- SIP read from 83.125.8.46:5060:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.2.15:5060;branch=z9hG4bK501ab53e;received=81.214.188.91;rport=5060
From: "1024" <sip:000387233745@192.168.2.15>;tag=as20b9cbea
To: <sip:004971311298862@voip.dus.net>;tag=as73d777d3
Call-ID: 70e04fe371dd539c6cab3d4627e544f2@192.168.2.15
CSeq: 103 INVITE
User-Agent: SIP_GW_1.4.2.PL3 talos.dus.net
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
--- (10 headers 0 lines) ---
Transmitting (NAT) to 83.125.8.46:5060:
ACK sip:004971311298862@voip.dus.net SIP/2.0
Via: SIP/2.0/UDP 192.168.2.15:5060;branch=z9hG4bK501ab53e;rport
From: "1024" <sip:000387233745@192.168.2.15>;tag=as20b9cbea
To: <sip:004971311298862@voip.dus.net>;tag=as73d777d3
Contact: <sip:000387233745@192.168.2.15>
Call-ID: 70e04fe371dd539c6cab3d4627e544f2@192.168.2.15
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
pbx*CLI> sip debu
<-- SIP read from 83.125.8.46:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.15:5060;branch=z9hG4bK501ab53e;received=81.214.188.91;rport=5060
From: "1024" <sip:000387233745@192.168.2.15>;tag=as20b9cbea
To: <sip:004971311298862@voip.dus.net>;tag=as73d777d3
Call-ID: 70e04fe371dd539c6cab3d4627e544f2@192.168.2.15
CSeq: 103 CANCEL
User-Agent: SIP_GW_1.4.2.PL3 talos.dus.net
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:004971311298862@83.125.8.46>
Content-Length: 0
--- (11 headers 0 lines) ---
Destroying call '70e04fe371dd539c6cab3d4627e544f2@192.168.2.15'
pbx*CLI> sip debug
<-- SIP read from 10.0.0.2:5060:
BYE sip:02452155048@10.0.0.1 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.2;branch=z9hG4bKac260466823
Max-Forwards: 70
From: <sip:1008@10.0.0.1>;tag=1c239635997
To: <sip:02452155048@10.0.0.1;user=phone>;tag=as07d9b14a
Call-ID: 23963575711200042012@10.0.0.2
CSeq: 3 BYE
Supported: em,timer,replaces,path,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Audiocodes-Sip-Gateway-MP-124 FXS/v.5.00A.027.005
Reason: Q.850 ;cause=16
Content-Length: 0
--- (12 headers 0 lines) ---
Sending to 10.0.0.2 : 5060 (non-NAT)
Transmitting (no NAT) to 10.0.0.2:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.2;branch=z9hG4bKac260466823;received=10.0.0.2
From: <sip:1008@10.0.0.1>;tag=1c239635997
To: <sip:02452155048@10.0.0.1;user=phone>;tag=as07d9b14a
Call-ID: 23963575711200042012@10.0.0.2
CSeq: 3 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:02452155048@10.0.0.1>
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing
pbx*CLI> sip debug
---
Scheduling destruction of call '262ccf6115d4bbb4467624dc0262a62f@192.168.2.15' in 32000 ms
set_destination: Parsing <sip:00492452155048@83.125.8.46> for address/port to send to
set_destination: set destination to 83.125.8.46, port 5060
Reliably Transmitting (NAT) to 83.125.8.46:5060:
BYE sip:00492452155048@83.125.8.46 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.15:5060;branch=z9hG4bK20fe8355;rport
From: "1008" <sip:000387233745@192.168.2.15>;tag=as26adf52c
To: <sip:00492452155048@voip.dus.net>;tag=as53cf50c2
Call-ID: 262ccf6115d4bbb4467624dc0262a62f@192.168.2.15
CSeq: 104 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Proxy-Authorization: Digest username="000387233745", realm="dus.net", algorithm=MD5, uri="sip:00492452155048@83.125.8.46", nonce="7cff320c", response="4e64b3fb70a09315f7f7c96b4fb3167c", opaque=""
Content-Length: 0
---
pbx*CLI> sip debug
<-- SIP read from 83.125.8.46:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.15:5060;branch=z9hG4bK20fe8355;received=81.214.188.91;rport=5060
From: "1008" <sip:000387233745@192.168.2.15>;tag=as26adf52c
To: <sip:00492452155048@voip.dus.net>;tag=as53cf50c2
Call-ID: 262ccf6115d4bbb4467624dc0262a62f@192.168.2.15
CSeq: 104 BYE
User-Agent: SIP_GW_1.4.2.PL3 talos.dus.net
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:00492452155048@83.125.8.46>
Content-Length: 0
--- (11 headers 0 lines) ---
Destroying call '262ccf6115d4bbb4467624dc0262a62f@192.168.2.15'
pbx*CLI> sip debug
<-- SIP read from 83.125.8.46:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.15:5060;branch=z9hG4bK5cf7c4c2;received=81.214.188.91;rport=5060
From: "1002" <sip:000387233745@192.168.2.15>;tag=as495acd44
To: <sip:00492771819382@voip.dus.net>;tag=as1a8e074b
Call-ID: 00d9911a36d325e378cc75451986412f@192.168.2.15
CSeq: 103 INVITE
User-Agent: SIP_GW_1.4.2.PL3 talos.dus.net
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:00492771819382@83.125.8.46>
Content-Type: application/sdp
Content-Length: 350
v=0
o=root 1387 1388 IN IP4 83.125.8.46
s=session
c=IN IP4 83.125.8.46
t=0 0
m=audio 5948 RTP/AVP 8 3 97 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
--- (12 headers 17 lines) ---
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 97
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 83.125.8.46:5948
Found description format PCMA
Found description format GSM
Found description format iLBC
Found description format G729
Found description format telephone-event
Capabilities: us - 0x50e (gsm|ulaw|alaw|g729|ilbc), peer - audio=0x50a (gsm|alaw|g729|ilbc)/video=0x0 (nothing), combined - 0x50a (gsm|alaw|g729|ilbc)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
list_route: hop: <sip:00492771819382@83.125.8.46>
set_destination: Parsing <sip:00492771819382@83.125.8.46> for address/port to send to
set_destination: set destination to 83.125.8.46, port 5060
Transmitting (NAT) to 83.125.8.46:5060:
ACK sip:00492771819382@83.125.8.46 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.15:5060;branch=z9hG4bK593d3c6b;rport
From: "1002" <sip:000387233745@192.168.2.15>;tag=as495acd44
To: <sip:00492771819382@voip.dus.net>;tag=as1a8e074b
Contact: <sip:000387233745@192.168.2.15>
Call-ID: 00d9911a36d325e378cc75451986412f@192.168.2.15
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
Destroying call '23963575711200042012@10.0.0.2'
We're at 10.0.0.1 port 25312
Adding codec 0x100 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.0.0.2:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.2;branch=z9hG4bKac244195942;received=10.0.0.2
From: <sip:1002@10.0.0.1>;tag=1c244170773
To: <sip:02771819382@10.0.0.1;user=phone>;tag=as3430dbc1
Call-ID: 24417053211200042016@10.0.0.2
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:02771819382@10.0.0.1>
Content-Type: application/sdp
Content-Length: 226
v=0
o=root 8992 8993 IN IP4 10.0.0.1
s=session
c=IN IP4 10.0.0.1
t=0 0
m=audio 25312 RTP/AVP 18 96
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-16
a=silenceSupp:off - - - -
---
pbx*CLI> sip debug
<-- SIP read from 10.0.0.2:5060:
ACK sip:02771819382@10.0.0.1 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.2;branch=z9hG4bKac260694128
Max-Forwards: 70
From: <sip:1002@10.0.0.1>;tag=1c244170773
To: <sip:02771819382@10.0.0.1;user=phone>;tag=as3430dbc1
Call-ID: 24417053211200042016@10.0.0.2
CSeq: 2 ACK
Contact: <sip:1002@10.0.0.2>
Supported: em,timer,replaces,path,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Audiocodes-Sip-Gateway-MP-124 FXS/v.5.00A.027.005
Content-Length: 0
--- (12 headers 0 lines) ---
pbx*CLI> sip no debug