This one is from Gigaset
<--- Transmitting (no NAT) to 192.168.0.101:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bKd3d75c495de6124bca61f620ce97fbd;received=192.168.0.101;rport=5060
From: "1" <sip:[email protected]>;tag=2907198688
To: <sip:[email protected];user=phone>;tag=as50fff393
Call-ID: 2656275381@192_168_0_101
CSeq: 3 INVITE
Server: FPBX-12.0.76.4(11.21.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:[email protected]:5060>
Remote-Party-ID: "Dahua" <sip:[email protected]>;party=called;privacy=off;screen=no
Content-Length: 0
<------------>
-- Connected line update to SIP/1-0000004a prevented.
-- SIP/8001-0000004b is ringing
<--- Transmitting (no NAT) to 192.168.0.101:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bKd3d75c495de6124bca61f620ce97fbd;received=192.168.0.101;rport=5060
From: "1" <sip:[email protected]>;tag=2907198688
To: <sip:[email protected];user=phone>;tag=as50fff393
Call-ID: 2656275381@192_168_0_101
CSeq: 3 INVITE
Server: FPBX-12.0.76.4(11.21.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:[email protected]:5060>
Content-Length: 0
<------------>
-- Connected line update to SIP/1-0000004a prevented.
-- SIP/8001-0000004b answered SIP/1-0000004a
Audio is at 15844
Adding codec 100003 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (no NAT) to 192.168.0.101:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bKd3d75c495de6124bca61f620ce97fbd;received=192.168.0.101;rport=5060
From: "1" <sip:[email protected]>;tag=2907198688
To: <sip:[email protected];user=phone>;tag=as50fff393
Call-ID: 2656275381@192_168_0_101
CSeq: 3 INVITE
Server: FPBX-12.0.76.4(11.21.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:[email protected]:5060>
Remote-Party-ID: "Dahua" <sip:[email protected]>;party=called;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 234
v=0
o=root 49481047 49481047 IN IP4 192.168.0.154
s=Asterisk PBX 11.21.0
c=IN IP4 192.168.0.154
t=0 0
m=audio 15844 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------>
> 0x7f10f4289d20 -- Probation passed - setting RTP source address to 192.168.0.101:5018
Retransmitting #1 (no NAT) to 192.168.0.101:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bKd3d75c495de6124bca61f620ce97fbd;received=192.168.0.101;rport=5060
From: "1" <sip:[email protected]>;tag=2907198688
To: <sip:[email protected];user=phone>;tag=as50fff393
Call-ID: 2656275381@192_168_0_101
CSeq: 3 INVITE
Server: FPBX-12.0.76.4(11.21.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:[email protected]:5060>
Remote-Party-ID: "Dahua" <sip:[email protected]>;party=called;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 234
v=0
o=root 49481047 49481047 IN IP4 192.168.0.154
s=Asterisk PBX 11.21.0
c=IN IP4 192.168.0.154
t=0 0
m=audio 15844 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
---
<--- SIP read from UDP:192.168.0.101:5060 --->
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bKfd7b6556238dac9dec0f203567b39ce6;rport
From: "1" <sip:[email protected]>;tag=2907198688
To: <sip:[email protected];user=phone>;tag=as50fff393
Call-ID: 2656275381@192_168_0_101
CSeq: 3 ACK
Contact: <sip:[email protected]:5060>
Authorization: Digest username="1", realm="asterisk", algorithm=MD5, uri="sip:[email protected];user=phone", nonce="6ade81af", response="09f9da7ec817abb78eb9235c39303c0e"
Max-Forwards: 70
User-Agent: C610 IP/42.076.00.000.000
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
<--- SIP read from UDP:192.168.0.101:5060 --->
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.101:5060;branch=z9hG4bKfd3f422bd105d125d315074885afca59;rport
From: "1" <sip:[email protected]>;tag=2907198688
To: <sip:[email protected];user=phone>;tag=as50fff393
Call-ID: 2656275381@192_168_0_101
CSeq: 3 ACK
Contact: <sip:[email protected]:5060>
Authorization: Digest username="1", realm="asterisk", algorithm=MD5, uri="sip:[email protected];user=phone", nonce="6ade81af", response="09f9da7ec817abb78eb9235c39303c0e"
Max-Forwards: 70
User-Agent: C610 IP/42.076.00.000.000
Content-Length: 0
<------------->
This one is from the intercom debug:
<--- Transmitting (no NAT) to 192.168.0.166:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.166:5060;branch=z9hG4bK1949793720;received=192.168.0.166;rport=5060
From: <sip:[email protected]>;tag=392762144
To: <sip:[email protected]:5060>;tag=as119cc3c6
Call-ID: [email protected]
CSeq: 21 INVITE
Server: FPBX-12.0.76.4(11.21.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:[email protected]:5060>
Content-Length: 0
<------------>
-- Connected line update to SIP/8001-00000050 prevented.
-- SIP/1-00000051 is ringing
<--- Transmitting (no NAT) to 192.168.0.166:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.166:5060;branch=z9hG4bK1949793720;received=192.168.0.166;rport=5060
From: <sip:[email protected]>;tag=392762144
To: <sip:[email protected]:5060>;tag=as119cc3c6
Call-ID: [email protected]
CSeq: 21 INVITE
Server: FPBX-12.0.76.4(11.21.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:[email protected]:5060>
Content-Length: 0
<------------>
-- Connected line update to SIP/8001-00000050 prevented.
-- SIP/1-00000051 answered SIP/8001-00000050
Audio is at 15324
Adding codec 100003 (ulaw) to SDP
<--- Reliably Transmitting (no NAT) to 192.168.0.166:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.166:5060;branch=z9hG4bK1949793720;received=192.168.0.166;rport=5060
From: <sip:[email protected]>;tag=392762144
To: <sip:[email protected]:5060>;tag=as119cc3c6
Call-ID: [email protected]
CSeq: 21 INVITE
Server: FPBX-12.0.76.4(11.21.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:[email protected]:5060>
Content-Type: application/sdp
Content-Length: 204
v=0
o=root 1821112262 1821112262 IN IP4 192.168.0.154
s=Asterisk PBX 11.21.0
c=IN IP4 192.168.0.154
t=0 0
m=video 0 RTP/AVP 96
m=audio 15324 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20
a=sendrecv
<------------>
<--- SIP read from UDP:192.168.0.166:5060 --->
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.166:5060;rport;branch=z9hG4bK929229632
From: <sip:[email protected]>;tag=392762144
To: <sip:[email protected]:5060>;tag=as119cc3c6
Call-ID: [email protected]
CSeq: 21 ACK
Contact: <sip:[email protected]:5060>
Max-Forwards: 70
User-Agent: Dahua UAC/3.0 VTO2000A V1.200.1000.0
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
> 0x21a8d10 -- Probation passed - setting RTP source address to 192.168.0.101:5004
> 0x7f10f4279f70 -- Probation passed - setting RTP source address to 192.168.0.166:20000
<--- SIP read from UDP:192.168.0.166:5060 --->
BYE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.166:5060;rport;branch=z9hG4bK2010863511
From: <sip:[email protected]>;tag=392762144
To: <sip:[email protected]:5060>;tag=as119cc3c6
Call-ID: [email protected]
CSeq: 22 BYE
Contact: <sip:[email protected]:5060>
Authorization: Digest username="8001", realm="asterisk", nonce="2e87e257", uri="sip:[email protected]:5060", response="55cda55e97e16028a57f1e1f8ee572e2", algorithm=MD5
Max-Forwards: 70
User-Agent: Dahua UAC/3.0 VTO2000A V1.200.1000.0
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
Sending to 192.168.0.166:5060 (no NAT)
Scheduling destruction of SIP dialog '[email protected]' in 6400 ms (Method: BYE)
<--- Transmitting (no NAT) to 192.168.0.166:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.166:5060;branch=z9hG4bK2010863511;received=192.168.0.166;rport=5060
From: <sip:[email protected]>;tag=392762144
To: <sip:[email protected]:5060>;tag=as119cc3c6
Call-ID: [email protected]
CSeq: 22 BYE
Server: FPBX-12.0.76.4(11.21.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
This one is from intercom
<--- Transmitting (no NAT) to 192.168.0.166:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.166:5060;branch=z9hG4bK1949793720;received=192.168.0.166;rport=5060
From: <sip:[email protected]>;tag=392762144
To: <sip:[email protected]:5060>;tag=as119cc3c6
Call-ID: [email protected]
CSeq: 21 INVITE
Server: FPBX-12.0.76.4(11.21.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:[email protected]:5060>
Content-Length: 0
<------------>
-- Connected line update to SIP/8001-00000050 prevented.
-- SIP/1-00000051 is ringing
<--- Transmitting (no NAT) to 192.168.0.166:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.166:5060;branch=z9hG4bK1949793720;received=192.168.0.166;rport=5060
From: <sip:[email protected]>;tag=392762144
To: <sip:[email protected]:5060>;tag=as119cc3c6
Call-ID: [email protected]
CSeq: 21 INVITE
Server: FPBX-12.0.76.4(11.21.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:[email protected]:5060>
Content-Length: 0
<------------>
-- Connected line update to SIP/8001-00000050 prevented.
-- SIP/1-00000051 answered SIP/8001-00000050
Audio is at 15324
Adding codec 100003 (ulaw) to SDP
<--- Reliably Transmitting (no NAT) to 192.168.0.166:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.166:5060;branch=z9hG4bK1949793720;received=192.168.0.166;rport=5060
From: <sip:[email protected]>;tag=392762144
To: <sip:[email protected]:5060>;tag=as119cc3c6
Call-ID: [email protected]
CSeq: 21 INVITE
Server: FPBX-12.0.76.4(11.21.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:[email protected]:5060>
Content-Type: application/sdp
Content-Length: 204
v=0
o=root 1821112262 1821112262 IN IP4 192.168.0.154
s=Asterisk PBX 11.21.0
c=IN IP4 192.168.0.154
t=0 0
m=video 0 RTP/AVP 96
m=audio 15324 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20
a=sendrecv
<------------>
<--- SIP read from UDP:192.168.0.166:5060 --->
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.166:5060;rport;branch=z9hG4bK929229632
From: <sip:[email protected]>;tag=392762144
To: <sip:[email protected]:5060>;tag=as119cc3c6
Call-ID: [email protected]
CSeq: 21 ACK
Contact: <sip:[email protected]:5060>
Max-Forwards: 70
User-Agent: Dahua UAC/3.0 VTO2000A V1.200.1000.0
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
> 0x21a8d10 -- Probation passed - setting RTP source address to 192.168.0.101:5004
> 0x7f10f4279f70 -- Probation passed - setting RTP source address to 192.168.0.166:20000
<--- SIP read from UDP:192.168.0.166:5060 --->
BYE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.166:5060;rport;branch=z9hG4bK2010863511
From: <sip:[email protected]>;tag=392762144
To: <sip:[email protected]:5060>;tag=as119cc3c6
Call-ID: [email protected]
CSeq: 22 BYE
Contact: <sip:[email protected]:5060>
Authorization: Digest username="8001", realm="asterisk", nonce="2e87e257", uri="sip:[email protected]:5060", response="55cda55e97e16028a57f1e1f8ee572e2", algorithm=MD5
Max-Forwards: 70
User-Agent: Dahua UAC/3.0 VTO2000A V1.200.1000.0
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
Sending to 192.168.0.166:5060 (no NAT)
Scheduling destruction of SIP dialog '[email protected]' in 6400 ms (Method: BYE)
<--- Transmitting (no NAT) to 192.168.0.166:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.166:5060;branch=z9hG4bK2010863511;received=192.168.0.166;rport=5060
From: <sip:[email protected]>;tag=392762144
To: <sip:[email protected]:5060>;tag=as119cc3c6
Call-ID: [email protected]
CSeq: 22 BYE
Server: FPBX-12.0.76.4(11.21.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0