hi,
let me explain in detail, what i have configured and what is happening now:

1st router w724v (Deutsche Telekom AG):
- port forwarding, everything to destination port 51000-55999 to device with ip 192.168.2.50 (interface of 2nd router)
2nd router Bintec RS353j):
- configured NAT, everything to port 51000-55999 to device 192.168.3.99 (same ports)

other direction is totally open.

I observed that all sip calls are closed exactly after 32s. call is disconnected on calling side as well... seems to be a timeout issue.

here i have some debug logs. I see lot of requests from asterisk to sipgate.de, which are not answered. but communication is going fine in both directions (otherwise registration would not be possible?):


<--- Received SIP request (1302 bytes) from UDP:217.10.79.9:5060 --->
INVITE sip:2636146e0@80.142.13.32:55060 SIP/2.0
Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK56ac.c519528374e3101a61791cf5a0ad1aae.0 Via: SIP/2.0/UDP 172.20.40.6;branch=z9hG4bK56ac.2ecd3532ae51c927dabcc6e27eaa4cbe.0 Via: SIP/2.0/UDP 217.10.68.137;branch=z9hG4bK56ac.73e224299594933979fdfb5b036e6563.0
Via: SIP/2.0/UDP 217.10.77.115:5060;branch=z9hG4bK7b31f031
Record-Route: <sip:217.10.79.9;lr;ftag=as02fa8fcc>
Record-Route: <sip:172.20.40.6;lr>
Record-Route: <sip:217.10.68.137;lr;ftag=as02fa8fcc>
From: "02363361779" <sip:02363361...@sipgate.de>;tag=as02fa8fcc
To: <sip:263614...@sipgate.de>
Call-ID: 370c0afa42c39f3d4ba96d7b0c1e7...@sipgate.de
CSeq: 103 INVITE
Contact: <sip:0xxxxxxxx9@217.10.77.115:5060>
max-forwards: 66
supported: replaces
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Content-Type: application/sdp
Content-Length:   394

v=0
o=root 15363811 15363812 IN IP4 192.168.2.1
s=sipgate VoIP GW
c=IN IP4 192.168.2.1
t=0 0
m=audio 7070 RTP/AVP 8 0 3 97 18 112 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/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:112 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<--- Transmitting SIP response (733 bytes) to UDP:217.10.79.9:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 217.10.79.9;rport=5060;received=217.10.79.9;branch=z9hG4bK56ac.c519528374e3101a61791cf5a0ad1aae.0 Via: SIP/2.0/UDP 172.20.40.6;branch=z9hG4bK56ac.2ecd3532ae51c927dabcc6e27eaa4cbe.0 Via: SIP/2.0/UDP 217.10.68.137;branch=z9hG4bK56ac.73e224299594933979fdfb5b036e6563.0
Via: SIP/2.0/UDP 217.10.77.115:5060;branch=z9hG4bK7b31f031
Record-Route: <sip:217.10.79.9;lr;ftag=as02fa8fcc>
Record-Route: <sip:172.20.40.6;lr>
Record-Route: <sip:217.10.68.137;lr;ftag=as02fa8fcc>
Call-ID: 370c0afa42c39f3d4ba96d7b0c1e7...@sipgate.de
From: "0xxxxxxxx9" <sip:0xxxxxx...@sipgate.de>;tag=as02fa8fcc
To: <sip:263614...@sipgate.de>
CSeq: 103 INVITE
Server: FPBX-13.0.188.8(13.11.2)
Content-Length:  0

<--- Transmitting SIP response (1280 bytes) to UDP:217.10.79.9:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.10.79.9;rport=5060;received=217.10.79.9;branch=z9hG4bK56ac.c519528374e3101a61791cf5a0ad1aae.0 Via: SIP/2.0/UDP 172.20.40.6;branch=z9hG4bK56ac.2ecd3532ae51c927dabcc6e27eaa4cbe.0 Via: SIP/2.0/UDP 217.10.68.137;branch=z9hG4bK56ac.73e224299594933979fdfb5b036e6563.0
Via: SIP/2.0/UDP 217.10.77.115:5060;branch=z9hG4bK7b31f031
Record-Route: <sip:217.10.79.9;lr;ftag=as02fa8fcc>
Record-Route: <sip:172.20.40.6;lr>
Record-Route: <sip:217.10.68.137;lr;ftag=as02fa8fcc>
Call-ID: 370c0afa42c39f3d4ba96d7b0c1e7...@sipgate.de
From: "0xxxxxxxx9" <sip:0xxxxxx...@sipgate.de>;tag=as02fa8fcc
To: <sip:263614...@sipgate.de>;tag=HvcIS2lEIQ9xPKihn9LFHjOtJ2YUNEXf
CSeq: 103 INVITE
Server: FPBX-13.0.188.8(13.11.2)
Contact: <sip:80.142.13.32:55060>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length:   286

v=0
o=- 15363811 15363814 IN IP4 192.168.3.99
s=Asterisk
c=IN IP4 80.142.13.32
t=0 0
m=audio 51822 RTP/AVP 8 3 112 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:112 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Received SIP request (1302 bytes) from UDP:217.10.79.9:5060 --->
INVITE sip:2636146e0@80.142.13.32:55060 SIP/2.0
Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK56ac.c519528374e3101a61791cf5a0ad1aae.0 Via: SIP/2.0/UDP 172.20.40.6;branch=z9hG4bK56ac.2ecd3532ae51c927dabcc6e27eaa4cbe.0 Via: SIP/2.0/UDP 217.10.68.137;branch=z9hG4bK56ac.73e224299594933979fdfb5b036e6563.0
Via: SIP/2.0/UDP 217.10.77.115:5060;branch=z9hG4bK7b31f031
Record-Route: <sip:217.10.79.9;lr;ftag=as02fa8fcc>
Record-Route: <sip:172.20.40.6;lr>
Record-Route: <sip:217.10.68.137;lr;ftag=as02fa8fcc>
From: "0xxxxxxxx9" <sip:0xxxxxx...@sipgate.de>;tag=as02fa8fcc
To: <sip:263614...@sipgate.de>
Call-ID: 370c0afa42c39f3d4ba96d7b0c1e7...@sipgate.de
CSeq: 103 INVITE
Contact: <sip:0xxxxxxxx9@217.10.77.115:5060>
max-forwards: 66
supported: replaces
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Content-Type: application/sdp
Content-Length:   394

v=0
o=root 15363811 15363812 IN IP4 192.168.2.1
s=sipgate VoIP GW
c=IN IP4 192.168.2.1
t=0 0
m=audio 7070 RTP/AVP 8 0 3 97 18 112 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/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:112 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<--- Transmitting SIP response (1280 bytes) to UDP:217.10.79.9:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.10.79.9;rport=5060;received=217.10.79.9;branch=z9hG4bK56ac.c519528374e3101a61791cf5a0ad1aae.0 Via: SIP/2.0/UDP 172.20.40.6;branch=z9hG4bK56ac.2ecd3532ae51c927dabcc6e27eaa4cbe.0 Via: SIP/2.0/UDP 217.10.68.137;branch=z9hG4bK56ac.73e224299594933979fdfb5b036e6563.0
Via: SIP/2.0/UDP 217.10.77.115:5060;branch=z9hG4bK7b31f031
Record-Route: <sip:217.10.79.9;lr;ftag=as02fa8fcc>
Record-Route: <sip:172.20.40.6;lr>
Record-Route: <sip:217.10.68.137;lr;ftag=as02fa8fcc>
Call-ID: 370c0afa42c39f3d4ba96d7b0c1e7...@sipgate.de
From: "0xxxxxxxx9" <sip:0xxxxxx...@sipgate.de>;tag=as02fa8fcc
To: <sip:263614...@sipgate.de>;tag=HvcIS2lEIQ9xPKihn9LFHjOtJ2YUNEXf
CSeq: 103 INVITE
Server: FPBX-13.0.188.8(13.11.2)
Contact: <sip:80.142.13.32:55060>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length:   286

v=0
o=- 15363811 15363814 IN IP4 192.168.3.99
s=Asterisk
c=IN IP4 80.142.13.32
t=0 0
m=audio 51822 RTP/AVP 8 3 112 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:112 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Transmitting SIP response (1280 bytes) to UDP:217.10.79.9:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.10.79.9;rport=5060;received=217.10.79.9;branch=z9hG4bK56ac.c519528374e3101a61791cf5a0ad1aae.0 Via: SIP/2.0/UDP 172.20.40.6;branch=z9hG4bK56ac.2ecd3532ae51c927dabcc6e27eaa4cbe.0 Via: SIP/2.0/UDP 217.10.68.137;branch=z9hG4bK56ac.73e224299594933979fdfb5b036e6563.0
Via: SIP/2.0/UDP 217.10.77.115:5060;branch=z9hG4bK7b31f031
Record-Route: <sip:217.10.79.9;lr;ftag=as02fa8fcc>
Record-Route: <sip:172.20.40.6;lr>
Record-Route: <sip:217.10.68.137;lr;ftag=as02fa8fcc>
Call-ID: 370c0afa42c39f3d4ba96d7b0c1e7...@sipgate.de
From: "0xxxxxxxx9" <sip:0xxxxxx...@sipgate.de>;tag=as02fa8fcc
To: <sip:263614...@sipgate.de>;tag=HvcIS2lEIQ9xPKihn9LFHjOtJ2YUNEXf
CSeq: 103 INVITE
Server: FPBX-13.0.188.8(13.11.2)
Contact: <sip:80.142.13.32:55060>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length:   286

v=0
o=- 15363811 15363814 IN IP4 192.168.3.99
s=Asterisk
c=IN IP4 80.142.13.32
t=0 0
m=audio 51822 RTP/AVP 8 3 112 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:112 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

-- Executing [s@app-announcement-1:3] Wait("PJSIP/pjsip_sipgate-00000003", "1") in new stack > 0x7f2ee8037810 -- Probation passed - setting RTP source address to 192.168.2.1:7070 -- Executing [s@app-announcement-1:4] NoOp("PJSIP/pjsip_sipgate-00000003", "Playing announcement ARAZ (Au├čerhalb Regelarbeitszeit)") in new stack -- Executing [s@app-announcement-1:5] Playback("PJSIP/pjsip_sipgate-00000003", "custom/araz01&custom/07-polly,noanswer") in new stack -- <PJSIP/pjsip_sipgate-00000003> Playing 'custom/araz01.alaw' (language 'en')
<--- Transmitting SIP response (1280 bytes) to UDP:217.10.79.9:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.10.79.9;rport=5060;received=217.10.79.9;branch=z9hG4bK56ac.c519528374e3101a61791cf5a0ad1aae.0 Via: SIP/2.0/UDP 172.20.40.6;branch=z9hG4bK56ac.2ecd3532ae51c927dabcc6e27eaa4cbe.0 Via: SIP/2.0/UDP 217.10.68.137;branch=z9hG4bK56ac.73e224299594933979fdfb5b036e6563.0
Via: SIP/2.0/UDP 217.10.77.115:5060;branch=z9hG4bK7b31f031
Record-Route: <sip:217.10.79.9;lr;ftag=as02fa8fcc>
Record-Route: <sip:172.20.40.6;lr>
Record-Route: <sip:217.10.68.137;lr;ftag=as02fa8fcc>
Call-ID: 370c0afa42c39f3d4ba96d7b0c1e7...@sipgate.de
From: "0xxxxxxxx9" <sip:0xxxxxx...@sipgate.de>;tag=as02fa8fcc
To: <sip:263614...@sipgate.de>;tag=HvcIS2lEIQ9xPKihn9LFHjOtJ2YUNEXf
CSeq: 103 INVITE
Server: FPBX-13.0.188.8(13.11.2)
Contact: <sip:80.142.13.32:55060>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length:   286

v=0
o=- 15363811 15363814 IN IP4 192.168.3.99
s=Asterisk
c=IN IP4 80.142.13.32
t=0 0
m=audio 51822 RTP/AVP 8 3 112 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:112 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Transmitting SIP response (1280 bytes) to UDP:217.10.79.9:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.10.79.9;rport=5060;received=217.10.79.9;branch=z9hG4bK56ac.c519528374e3101a61791cf5a0ad1aae.0 Via: SIP/2.0/UDP 172.20.40.6;branch=z9hG4bK56ac.2ecd3532ae51c927dabcc6e27eaa4cbe.0 Via: SIP/2.0/UDP 217.10.68.137;branch=z9hG4bK56ac.73e224299594933979fdfb5b036e6563.0
Via: SIP/2.0/UDP 217.10.77.115:5060;branch=z9hG4bK7b31f031
Record-Route: <sip:217.10.79.9;lr;ftag=as02fa8fcc>
Record-Route: <sip:172.20.40.6;lr>
Record-Route: <sip:217.10.68.137;lr;ftag=as02fa8fcc>
Call-ID: 370c0afa42c39f3d4ba96d7b0c1e7...@sipgate.de
From: "0xxxxxxxx9" <sip:0xxxxxx...@sipgate.de>;tag=as02fa8fcc
To: <sip:263614...@sipgate.de>;tag=HvcIS2lEIQ9xPKihn9LFHjOtJ2YUNEXf
CSeq: 103 INVITE
Server: FPBX-13.0.188.8(13.11.2)
Contact: <sip:80.142.13.32:55060>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length:   286

v=0
o=- 15363811 15363814 IN IP4 192.168.3.99
s=Asterisk
c=IN IP4 80.142.13.32
t=0 0
m=audio 51822 RTP/AVP 8 3 112 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:112 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

-- <PJSIP/pjsip_sipgate-00000003> Playing 'custom/07-polly.slin' (language 'en')
<--- Transmitting SIP response (1280 bytes) to UDP:217.10.79.9:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.10.79.9;rport=5060;received=217.10.79.9;branch=z9hG4bK56ac.c519528374e3101a61791cf5a0ad1aae.0 Via: SIP/2.0/UDP 172.20.40.6;branch=z9hG4bK56ac.2ecd3532ae51c927dabcc6e27eaa4cbe.0 Via: SIP/2.0/UDP 217.10.68.137;branch=z9hG4bK56ac.73e224299594933979fdfb5b036e6563.0
Via: SIP/2.0/UDP 217.10.77.115:5060;branch=z9hG4bK7b31f031
Record-Route: <sip:217.10.79.9;lr;ftag=as02fa8fcc>
Record-Route: <sip:172.20.40.6;lr>
Record-Route: <sip:217.10.68.137;lr;ftag=as02fa8fcc>
Call-ID: 370c0afa42c39f3d4ba96d7b0c1e7...@sipgate.de
From: "0xxxxxxxx9" <sip:0xxxxxx...@sipgate.de>;tag=as02fa8fcc
To: <sip:263614...@sipgate.de>;tag=HvcIS2lEIQ9xPKihn9LFHjOtJ2YUNEXf
CSeq: 103 INVITE
Server: FPBX-13.0.188.8(13.11.2)
Contact: <sip:80.142.13.32:55060>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length:   286

v=0
o=- 15363811 15363814 IN IP4 192.168.3.99
s=Asterisk
c=IN IP4 80.142.13.32
t=0 0
m=audio 51822 RTP/AVP 8 3 112 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:112 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Transmitting SIP request (429 bytes) to UDP:217.10.79.9:5060 --->
OPTIONS sip:263614...@sipgate.de:5060 SIP/2.0
Via: SIP/2.0/UDP 80.142.13.32:55060;rport;branch=z9hG4bKPj9pcvGusLP-xT4EfBJ4T9sYZ8jerfCb3E
From: <sip:263614...@sipgate.de>;tag=Ji-JiXBLWG1GmDEKXfwdQW0pVqiyOgOO
To: <sip:263614...@sipgate.de>
Contact: <sip:2636146e0@80.142.13.32:55060>
Call-ID: 0aV3SBgGaxKCUhyphLjZTZ3sc0-LvExV
CSeq: 43608 OPTIONS
Max-Forwards: 70
User-Agent: FPBX-13.0.188.8(13.11.2)
Content-Length:  0


<--- Transmitting SIP request (429 bytes) to UDP:217.10.79.9:5060 --->
OPTIONS sip:263614...@sipgate.de:5060 SIP/2.0
Via: SIP/2.0/UDP 80.142.13.32:55060;rport;branch=z9hG4bKPj9pcvGusLP-xT4EfBJ4T9sYZ8jerfCb3E
From: <sip:263614...@sipgate.de>;tag=Ji-JiXBLWG1GmDEKXfwdQW0pVqiyOgOO
To: <sip:263614...@sipgate.de>
Contact: <sip:2636146e0@80.142.13.32:55060>
Call-ID: 0aV3SBgGaxKCUhyphLjZTZ3sc0-LvExV
CSeq: 43608 OPTIONS
Max-Forwards: 70
User-Agent: FPBX-13.0.188.8(13.11.2)
Content-Length:  0


<--- Received SIP response (338 bytes) from UDP:217.10.79.9:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 80.142.13.32:55060;rport;branch=z9hG4bKPj9pcvGusLP-xT4EfBJ4T9sYZ8jerfCb3E
From: <sip:263614...@sipgate.de>;tag=Ji-JiXBLWG1GmDEKXfwdQW0pVqiyOgOO
To: <sip:263614...@sipgate.de>;tag=065a2aa3915c789dd1a0ab4f12b0002c.4434
Call-ID: 0aV3SBgGaxKCUhyphLjZTZ3sc0-LvExV
CSeq: 43608 OPTIONS
Content-Length: 0


<--- Received SIP response (338 bytes) from UDP:217.10.79.9:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 80.142.13.32:55060;rport;branch=z9hG4bKPj9pcvGusLP-xT4EfBJ4T9sYZ8jerfCb3E
From: <sip:263614...@sipgate.de>;tag=Ji-JiXBLWG1GmDEKXfwdQW0pVqiyOgOO
To: <sip:263614...@sipgate.de>;tag=065a2aa3915c789dd1a0ab4f12b0002c.4434
Call-ID: 0aV3SBgGaxKCUhyphLjZTZ3sc0-LvExV
CSeq: 43608 OPTIONS
Content-Length: 0


<--- Transmitting SIP response (1280 bytes) to UDP:217.10.79.9:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.10.79.9;rport=5060;received=217.10.79.9;branch=z9hG4bK56ac.c519528374e3101a61791cf5a0ad1aae.0 Via: SIP/2.0/UDP 172.20.40.6;branch=z9hG4bK56ac.2ecd3532ae51c927dabcc6e27eaa4cbe.0 Via: SIP/2.0/UDP 217.10.68.137;branch=z9hG4bK56ac.73e224299594933979fdfb5b036e6563.0
Via: SIP/2.0/UDP 217.10.77.115:5060;branch=z9hG4bK7b31f031
Record-Route: <sip:217.10.79.9;lr;ftag=as02fa8fcc>
Record-Route: <sip:172.20.40.6;lr>
Record-Route: <sip:217.10.68.137;lr;ftag=as02fa8fcc>
Call-ID: 370c0afa42c39f3d4ba96d7b0c1e7...@sipgate.de
From: "0xxxxxxxx9" <sip:0xxxxxx...@sipgate.de>;tag=as02fa8fcc
To: <sip:263614...@sipgate.de>;tag=HvcIS2lEIQ9xPKihn9LFHjOtJ2YUNEXf
CSeq: 103 INVITE
Server: FPBX-13.0.188.8(13.11.2)
Contact: <sip:80.142.13.32:55060>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length:   286

v=0
o=- 15363811 15363814 IN IP4 192.168.3.99
s=Asterisk
c=IN IP4 80.142.13.32
t=0 0
m=audio 51822 RTP/AVP 8 3 112 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:112 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Transmitting SIP response (1280 bytes) to UDP:217.10.79.9:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.10.79.9;rport=5060;received=217.10.79.9;branch=z9hG4bK56ac.c519528374e3101a61791cf5a0ad1aae.0 Via: SIP/2.0/UDP 172.20.40.6;branch=z9hG4bK56ac.2ecd3532ae51c927dabcc6e27eaa4cbe.0 Via: SIP/2.0/UDP 217.10.68.137;branch=z9hG4bK56ac.73e224299594933979fdfb5b036e6563.0
Via: SIP/2.0/UDP 217.10.77.115:5060;branch=z9hG4bK7b31f031
Record-Route: <sip:217.10.79.9;lr;ftag=as02fa8fcc>
Record-Route: <sip:172.20.40.6;lr>
Record-Route: <sip:217.10.68.137;lr;ftag=as02fa8fcc>
Call-ID: 370c0afa42c39f3d4ba96d7b0c1e7...@sipgate.de
From: "0xxxxxxxx9" <sip:0xxxxxx...@sipgate.de>;tag=as02fa8fcc
To: <sip:263614...@sipgate.de>;tag=HvcIS2lEIQ9xPKihn9LFHjOtJ2YUNEXf
CSeq: 103 INVITE
Server: FPBX-13.0.188.8(13.11.2)
Contact: <sip:80.142.13.32:55060>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length:   286

v=0
o=- 15363811 15363814 IN IP4 192.168.3.99
s=Asterisk
c=IN IP4 80.142.13.32
t=0 0
m=audio 51822 RTP/AVP 8 3 112 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:112 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Transmitting SIP response (1280 bytes) to UDP:217.10.79.9:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.10.79.9;rport=5060;received=217.10.79.9;branch=z9hG4bK56ac.c519528374e3101a61791cf5a0ad1aae.0 Via: SIP/2.0/UDP 172.20.40.6;branch=z9hG4bK56ac.2ecd3532ae51c927dabcc6e27eaa4cbe.0 Via: SIP/2.0/UDP 217.10.68.137;branch=z9hG4bK56ac.73e224299594933979fdfb5b036e6563.0
Via: SIP/2.0/UDP 217.10.77.115:5060;branch=z9hG4bK7b31f031
Record-Route: <sip:217.10.79.9;lr;ftag=as02fa8fcc>
Record-Route: <sip:172.20.40.6;lr>
Record-Route: <sip:217.10.68.137;lr;ftag=as02fa8fcc>
Call-ID: 370c0afa42c39f3d4ba96d7b0c1e7...@sipgate.de
From: "0xxxxxxxx9" <sip:0xxxxxx...@sipgate.de>;tag=as02fa8fcc
To: <sip:263614...@sipgate.de>;tag=HvcIS2lEIQ9xPKihn9LFHjOtJ2YUNEXf
CSeq: 103 INVITE
Server: FPBX-13.0.188.8(13.11.2)
Contact: <sip:80.142.13.32:55060>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length:   286

v=0
o=- 15363811 15363814 IN IP4 192.168.3.99
s=Asterisk
c=IN IP4 80.142.13.32
t=0 0
m=audio 51822 RTP/AVP 8 3 112 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:112 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Transmitting SIP response (1280 bytes) to UDP:217.10.79.9:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.10.79.9;rport=5060;received=217.10.79.9;branch=z9hG4bK56ac.c519528374e3101a61791cf5a0ad1aae.0 Via: SIP/2.0/UDP 172.20.40.6;branch=z9hG4bK56ac.2ecd3532ae51c927dabcc6e27eaa4cbe.0 Via: SIP/2.0/UDP 217.10.68.137;branch=z9hG4bK56ac.73e224299594933979fdfb5b036e6563.0
Via: SIP/2.0/UDP 217.10.77.115:5060;branch=z9hG4bK7b31f031
Record-Route: <sip:217.10.79.9;lr;ftag=as02fa8fcc>
Record-Route: <sip:172.20.40.6;lr>
Record-Route: <sip:217.10.68.137;lr;ftag=as02fa8fcc>
Call-ID: 370c0afa42c39f3d4ba96d7b0c1e7...@sipgate.de
From: "0xxxxxxxx9" <sip:0xxxxxx...@sipgate.de>;tag=as02fa8fcc
To: <sip:263614...@sipgate.de>;tag=HvcIS2lEIQ9xPKihn9LFHjOtJ2YUNEXf
CSeq: 103 INVITE
Server: FPBX-13.0.188.8(13.11.2)
Contact: <sip:80.142.13.32:55060>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length:   286

v=0
o=- 15363811 15363814 IN IP4 192.168.3.99
s=Asterisk
c=IN IP4 80.142.13.32
t=0 0
m=audio 51822 RTP/AVP 8 3 112 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:112 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Transmitting SIP response (1280 bytes) to UDP:217.10.79.9:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.10.79.9;rport=5060;received=217.10.79.9;branch=z9hG4bK56ac.c519528374e3101a61791cf5a0ad1aae.0 Via: SIP/2.0/UDP 172.20.40.6;branch=z9hG4bK56ac.2ecd3532ae51c927dabcc6e27eaa4cbe.0 Via: SIP/2.0/UDP 217.10.68.137;branch=z9hG4bK56ac.73e224299594933979fdfb5b036e6563.0
Via: SIP/2.0/UDP 217.10.77.115:5060;branch=z9hG4bK7b31f031
Record-Route: <sip:217.10.79.9;lr;ftag=as02fa8fcc>
Record-Route: <sip:172.20.40.6;lr>
Record-Route: <sip:217.10.68.137;lr;ftag=as02fa8fcc>
Call-ID: 370c0afa42c39f3d4ba96d7b0c1e7...@sipgate.de
From: "0xxxxxxxx9" <sip:0xxxxxx...@sipgate.de>;tag=as02fa8fcc
To: <sip:263614...@sipgate.de>;tag=HvcIS2lEIQ9xPKihn9LFHjOtJ2YUNEXf
CSeq: 103 INVITE
Server: FPBX-13.0.188.8(13.11.2)
Contact: <sip:80.142.13.32:55060>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length:   286

v=0
o=- 15363811 15363814 IN IP4 192.168.3.99
s=Asterisk
c=IN IP4 80.142.13.32
t=0 0
m=audio 51822 RTP/AVP 8 3 112 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:112 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

freepbx*CLI>
freepbx*CLI>
freepbx*CLI>
freepbx*CLI>
freepbx*CLI>
<--- Transmitting SIP request (543 bytes) to UDP:217.10.79.9:5060 --->
BYE sip:0xxxxxxxx9@217.10.77.115:5060 SIP/2.0
Via: SIP/2.0/UDP 80.142.13.32:55060;rport;branch=z9hG4bKPjmhpnZAJdsqBV9w-4WA.1DjZHqFpj6-au
From: <sip:263614...@sipgate.de>;tag=HvcIS2lEIQ9xPKihn9LFHjOtJ2YUNEXf
To: "0xxxxxxxx9" <sip:0xxxxxx...@sipgate.de>;tag=as02fa8fcc
Call-ID: 370c0afa42c39f3d4ba96d7b0c1e7...@sipgate.de
CSeq: 5732 BYE
Route: <sip:217.10.79.9;lr;ftag=as02fa8fcc>
Route: <sip:172.20.40.6;lr>
Route: <sip:217.10.68.137;lr;ftag=as02fa8fcc>
Max-Forwards: 70
User-Agent: FPBX-13.0.188.8(13.11.2)
Content-Length:  0


freepbx*CLI>
<--- Received SIP response (446 bytes) from UDP:217.10.79.9:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 80.142.13.32:55060;rport;branch=z9hG4bKPjmhpnZAJdsqBV9w-4WA.1DjZHqFpj6-au
From: <sip:263614...@sipgate.de>;tag=HvcIS2lEIQ9xPKihn9LFHjOtJ2YUNEXf
To: "0xxxxxxxx9" <sip:0xxxxxx...@sipgate.de>;tag=as02fa8fcc
Call-ID: 370c0afa42c39f3d4ba96d7b0c1e7...@sipgate.de
CSeq: 5732 BYE
supported: replaces
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Content-Length: 0


<--- Transmitting SIP request (545 bytes) to UDP:217.10.79.9:5060 --->
REGISTER sip:sipgate.de:5060 SIP/2.0
Via: SIP/2.0/UDP 80.142.13.32:55060;rport;branch=z9hG4bKPjShmmpRhUENHI8CUFtNiZttd1lZohqw6p
From: <sip:263614...@sipgate.de>;tag=PzFp-J0wxtCAh4UikI2rYw0agSBQB7c3
To: <sip:263614...@sipgate.de>
Call-ID: bFLef6CKy1KlGt-YYkjqV7ja3BmyYyCu
CSeq: 55530 REGISTER
Contact: <sip:2636146e0@80.142.13.32:55060>
Expires: 60
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, REFER, MESSAGE
Max-Forwards: 70
User-Agent: FPBX-13.0.188.8(13.11.2)
Content-Length:  0


<--- Received SIP response (436 bytes) from UDP:217.10.79.9:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 80.142.13.32:55060;rport;branch=z9hG4bKPjShmmpRhUENHI8CUFtNiZttd1lZohqw6p
From: <sip:263614...@sipgate.de>;tag=PzFp-J0wxtCAh4UikI2rYw0agSBQB7c3
To: <sip:263614...@sipgate.de>;tag=86e53dd608d1c001e0b8060625977563.c38e
Call-ID: bFLef6CKy1KlGt-YYkjqV7ja3BmyYyCu
CSeq: 55530 REGISTER
WWW-Authenticate: Digest realm="sipgate.de", nonce="WAIJSVgCCB1kfjXwrwmT7mfxLr/nkdQO"
Content-Length: 0


<--- Transmitting SIP request (723 bytes) to UDP:217.10.79.9:5060 --->
REGISTER sip:sipgate.de:5060 SIP/2.0
Via: SIP/2.0/UDP 80.142.13.32:55060;rport;branch=z9hG4bKPjkC0dwtjcOsKzwskJq2gE2RelAFlFm7cw
From: <sip:263614...@sipgate.de>;tag=PzFp-J0wxtCAh4UikI2rYw0agSBQB7c3
To: <sip:263614...@sipgate.de>
Call-ID: bFLef6CKy1KlGt-YYkjqV7ja3BmyYyCu
CSeq: 55531 REGISTER
Contact: <sip:2636146e0@80.142.13.32:55060>
Expires: 60
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, REFER, MESSAGE
Max-Forwards: 70
User-Agent: FPBX-13.0.188.8(13.11.2)
Authorization: Digest username="2636146e0", realm="sipgate.de", nonce="WAIJSVgCCB1kfjXwrwmT7mfxLr/nkdQO", uri="sip:sipgate.de:5060", response="514fd5c1b4aa1b951400836d2b5a0b10"
Content-Length:  0


<--- Received SIP response (395 bytes) from UDP:217.10.79.9:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 80.142.13.32:55060;rport;branch=z9hG4bKPjkC0dwtjcOsKzwskJq2gE2RelAFlFm7cw
From: <sip:263614...@sipgate.de>;tag=PzFp-J0wxtCAh4UikI2rYw0agSBQB7c3
To: <sip:263614...@sipgate.de>;tag=86e53dd608d1c001e0b8060625977563.2957
Call-ID: bFLef6CKy1KlGt-YYkjqV7ja3BmyYyCu
CSeq: 55531 REGISTER
Contact: <sip:2636146e0@80.142.13.32:55060>;expires=60
Content-Length: 0







kind regards,
andre



--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Join the Asterisk Community at the 13th AstriCon, September 27-29, 2016
     http://www.asterisk.org/community/astricon-user-conference

New to Asterisk? Start here:
     https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Reply via email to