<--- SIP read from UDP:192.168.177.1:5060 --->INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.177.1:5060;branch=z9hG4bKE743BE6DBCF527B9
From: "Fabian" <sip:**[email protected]>;tag=1E90C6F62A933263
To: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 37 INVITE
Contact: <sip:[email protected]>
Max-Forwards: 70
Expires: 120
User-Agent: AVM FRITZ!Box Fon WLAN 7270 v3 74.06.05 (Apr 9 2014)
Supported: 100rel,replaces,timer
Allow-Events: telephone-event,refer
Allow: INVITE,ACK,OPTIONS,CANCEL,BYE,UPDATE,PRACK,INFO,SUBSCRIBE,NOTIFY,REFER,MESSAGE,PUBLISH
Content-Type: application/sdp
Accept: application/sdp, multipart/mixed
Accept-Encoding: identity
Content-Length: 417
v=0
o=user 457784 457784 IN IP4 192.168.177.1
s=call
c=IN IP4 192.168.177.1
t=0 0
m=audio 7090 RTP/AVP 9 8 0 2 102 100 99 97 120 121 101
a=sendrecv
a=rtpmap:2 G726-32/8000
a=rtpmap:102 G726-32/8000
a=rtpmap:100 G726-40/8000
a=rtpmap:99 G726-24/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:120 PCMA/16000
a=rtpmap:121 PCMU/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=rtcp:7091
<------------->
--- (17 headers 18 lines) ---
Sending to 192.168.177.1:5060 (no NAT)
Sending to 192.168.177.1:5060 (no NAT)
Using INVITE request as basis request - [email protected]
Found peer '62' for '**610' from 192.168.177.1:5060
Found RTP audio format 9
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 2
Found RTP audio format 102
Found RTP audio format 100
Found RTP audio format 99
Found RTP audio format 97
Found RTP audio format 120
Found RTP audio format 121
Found RTP audio format 101
Found audio description format G726-32 for ID 2
Found audio description format G726-32 for ID 102
Found unknown media description format G726-40 for ID 100
Found unknown media description format G726-24 for ID 99
Found audio description format iLBC for ID 97
Found unknown media description format PCMA for ID 120
Found unknown media description format PCMU for ID 121
Found audio description format telephone-event for ID 101
Capabilities: us - (gsm|ulaw|alaw|g726), peer - audio=(ulaw|alaw|g726|ilbc|g722)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|g726)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.177.1:7090
Looking for 620 in from-internal (domain 192.168.177.2)
list_route: hop: <sip:[email protected]>
<--- Transmitting (no NAT) to 192.168.177.1:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.177.1:5060;branch=z9hG4bKE743BE6DBCF527B9;received=192.168.177.1
From: "Fabian" <sip:**[email protected]>;tag=1E90C6F62A933263
To: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 37 INVITE
Server: FPBX-13.0.74(11.21.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:[email protected]:5060>
Content-Length: 0
<------------>
Audio is at 12570
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100011 (g726) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Transmitting (no NAT) to 192.168.177.1:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.177.1:5060;branch=z9hG4bKE743BE6DBCF527B9;received=192.168.177.1
From: "Fabian" <sip:**[email protected]>;tag=1E90C6F62A933263
To: <sip:[email protected]:5060>;tag=as2c0fc927
Call-ID: [email protected]
CSeq: 37 INVITE
Server: FPBX-13.0.74(11.21.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:[email protected]:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 289
v=0
o=root 2063065793 2063065793 IN IP4 192.168.177.2
s=Asterisk PBX 11.21.0
c=IN IP4 192.168.177.2
t=0 0
m=audio 12570 RTP/AVP 0 8 2 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------>
<--- Reliably Transmitting (no NAT) to 192.168.177.1:5060 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.177.1:5060;branch=z9hG4bKE743BE6DBCF527B9;received=192.168.177.1
From: "Fabian" <sip:**[email protected]>;tag=1E90C6F62A933263
To: <sip:[email protected]:5060>;tag=as2c0fc927
Call-ID: [email protected]
CSeq: 37 INVITE
Server: FPBX-13.0.74(11.21.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Content-Length: 0
<------------>
[2016-04-17 00:59:53] WARNING[2767][C-00000005]: channel.c:4861 ast_prod: Prodding channel 'SIP/62-00000005' failed
<--- SIP read from UDP:192.168.177.1:5060 --->
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.177.1:5060;branch=z9hG4bKE743BE6DBCF527B9
From: "Fabian" <sip:**[email protected]>;tag=1E90C6F62A933263
To: <sip:[email protected]:5060>;tag=as2c0fc927
Call-ID: [email protected]
CSeq: 37 ACK
User-Agent: AVM FRITZ!Box Fon WLAN 7270 v3 74.06.05 (Apr 9 2014)
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '[email protected]' Method: ACK
[2016-04-17 01:00:01] NOTICE[1868]: chan_sip.c:15144 sip_reregister: -- Re-registration for [email protected]
REGISTER 11 headers, 0 lines
Reliably Transmitting (no NAT) to 192.168.177.1:5060:
REGISTER sip:192.168.177.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.177.2:5060;branch=z9hG4bK539a311f
Max-Forwards: 70
From: <sip:[email protected]>;tag=as46fcc038
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 106 REGISTER
User-Agent: FPBX-13.0.74(11.21.0)
Authorization: Digest username="620", realm="fritz.box", algorithm=MD5, uri="sip:192.168.177.1", nonce="CCB8FB7DDC97466F", response="bc1cedd7e061bcc3bfca81d70be4863d"
Expires: 120
Contact: <sip:[email protected]:5060>
Content-Length: 0
---
<--- SIP read from UDP:192.168.177.1:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.177.2:5060;branch=z9hG4bK539a311f
From: <sip:[email protected]>;tag=as46fcc038
To: <sip:[email protected]>;tag=35BC66007E41BD56
Call-ID: [email protected]
CSeq: 106 REGISTER
WWW-Authenticate: Digest realm="fritz.box", nonce="EC5A1FFD19E3FF19"
User-Agent: FRITZ!OS
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Responding to challenge, registration to domain/host name 192.168.177.1
REGISTER 11 headers, 0 lines
Reliably Transmitting (no NAT) to 192.168.177.1:5060:
REGISTER sip:192.168.177.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.177.2:5060;branch=z9hG4bK2effe0c6
Max-Forwards: 70
From: <sip:[email protected]>;tag=as46fcc038
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 107 REGISTER
User-Agent: FPBX-13.0.74(11.21.0)
Authorization: Digest username="620", realm="fritz.box", algorithm=MD5, uri="sip:192.168.177.1", nonce="EC5A1FFD19E3FF19", response="b2b8a28a0d35eaea8a35a2c475813c03"
Expires: 120
Contact: <sip:[email protected]:5060>
Content-Length: 0
---
<--- SIP read from UDP:192.168.177.1:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.177.2:5060;branch=z9hG4bK2effe0c6
From: <sip:[email protected]>;tag=as46fcc038
To: <sip:[email protected]>;tag=6278B62F9F8CDDC8
Call-ID: [email protected]
CSeq: 107 REGISTER
Contact: <sip:[email protected]:5060>;expires=300
User-Agent: AVM FRITZ!Box Fon WLAN 7270 v3 74.06.05 (Apr 9 2014)
Supported: 100rel,replaces,timer
Allow-Events: telephone-event,refer,reg
Allow: INVITE,ACK,OPTIONS,CANCEL,BYE,UPDATE,PRACK,INFO,SUBSCRIBE,NOTIFY,REFER,MESSAGE,PUBLISH
Accept: application/sdp, multipart/mixed
Accept-Encoding: identity
Content-Length: 0
<------------->
--- (14 headers 0 lines) ---
[2016-04-17 01:00:01] NOTICE[1868]: chan_sip.c:23710 handle_response_register: Outbound Registration: Expiry for 192.168.177.1 is 300 sec (Scheduling reregistration in 285 s)
Really destroying SIP dialog '[email protected]' Method: REGISTER
raspbx*CLI> sip set debug off
SIP Debugging Disabled