Добрый день. Имеется SIP вызывная панель и SIP монитор (BAS IP) + мобильные софтфоны . Привязаны к астериску.
При звонке с 1003 (Android) на 1004 (SIP панель) видео идет (всек ОК)
При звонке с 1007 (IOS) на 1004 (SIP панель) видео нет.
При звонке с 1007 (IOS) на 1003 (Android) видео идет в обе стороны (Все ОК)
Кодеки g711 и h324
Трассировку приложил. Прошу помощи.
Спасибо
SIP домофон. Нет видео
Re: SIP домофон. Нет видео
Платный суппорт, двойной тариф.
Re: SIP домофон. Нет видео
Как? Кому? Сколько?
Re: SIP домофон. Нет видео
PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:
sterisk*CLI> sip set debug peer 1007
SIP Debugging Enabled for IP: 94.31.155.12
<--- SIP read from UDP:94.31.155.12:62646 --->
INVITE sip:1004@94.31.202.73 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.108:56495;branch=z9hG4bK.Ji9KrcAks;rport
From: <sip:1007@94.31.202.73>;tag=s0MPxGOy1
To: sip:1004@94.31.202.73
CSeq: 20 INVITE
Call-ID: HJMp6JeMc3
Max-Forwards: 70
Supported: outbound
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
Content-Type: application/sdp
Content-Length: 402
Contact: <sip:1007@94.31.155.12:62646>;+sip.instance="<urn:uuid:dbedf596-2bcd-4e39-8872-ff75aa2962d5>"
User-Agent: LinphoneIPhone/2.1.0 (belle-sip/1.4.0)
v=0
o=1007 3084 2486 IN IP4 192.168.1.108
s=Talk
c=IN IP4 192.168.1.108
b=AS:380
t=0 0
a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
m=audio 7076 RTP/AVP 0 8 101
a=rtpmap:101 telephone-event/8000
m=video 9078 RTP/AVP 96 97 98
a=rtpmap:96 VP8/90000
a=rtpmap:97 H264/90000
a=fmtp:97 profile-level-id=42801F
a=rtpmap:98 MP4V-ES/90000
a=fmtp:98 profile-level-id=3
<------------->
--- (13 headers 15 lines) ---
Sending to 94.31.155.12:62646 (NAT)
Sending to 94.31.155.12:62646 (NAT)
Using INVITE request as basis request - HJMp6JeMc3
Found peer '1007' for '1007' from 94.31.155.12:62646
<--- Reliably Transmitting (NAT) to 94.31.155.12:62646 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.108:56495;branch=z9hG4bK.Ji9KrcAks;received=94.31.155.12;rport=62646
From: <sip:1007@94.31.202.73>;tag=s0MPxGOy1
To: sip:1004@94.31.202.73;tag=as4d8f0a34
Call-ID: HJMp6JeMc3
CSeq: 20 INVITE
Server: FPBX-12.0.63(13.2.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1de76e3a"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'HJMp6JeMc3' in 6400 ms (Method: INVITE)
Retransmitting #1 (NAT) to 94.31.155.12:62646:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.108:56495;branch=z9hG4bK.Ji9KrcAks;received=94.31.155.12;rport=62646
From: <sip:1007@94.31.202.73>;tag=s0MPxGOy1
To: sip:1004@94.31.202.73;tag=as4d8f0a34
Call-ID: HJMp6JeMc3
CSeq: 20 INVITE
Server: FPBX-12.0.63(13.2.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1de76e3a"
Content-Length: 0
---
<--- SIP read from UDP:94.31.155.12:62646 --->
ACK sip:1004@94.31.202.73 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.108:56495;branch=z9hG4bK.Ji9KrcAks;rport
Call-ID: HJMp6JeMc3
From: <sip:1007@94.31.202.73>;tag=s0MPxGOy1
To: <sip:1004@94.31.202.73>;tag=as4d8f0a34
Contact: <sip:1007@94.31.155.12:62646>;+sip.instance="<urn:uuid:dbedf596-2bcd-4e39-8872-ff75aa2962d5>"
Max-Forwards: 70
CSeq: 20 ACK
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from UDP:94.31.155.12:62646 --->
INVITE sip:1004@94.31.202.73 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.108:56495;branch=z9hG4bK.iSLC6T4aV;rport
From: <sip:1007@94.31.202.73>;tag=s0MPxGOy1
To: sip:1004@94.31.202.73
CSeq: 21 INVITE
Call-ID: HJMp6JeMc3
Max-Forwards: 70
Supported: outbound
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
Content-Type: application/sdp
Content-Length: 402
Contact: <sip:1007@94.31.155.12:62646>;+sip.instance="<urn:uuid:dbedf596-2bcd-4e39-8872-ff75aa2962d5>"
User-Agent: LinphoneIPhone/2.1.0 (belle-sip/1.4.0)
Authorization: Digest realm="asterisk", nonce="1de76e3a", algorithm=MD5, username="1007", uri="sip:1004@94.31.202.73", response="70efe49423a60eb04ecf6d082e89ed13"
v=0
o=1007 3084 2486 IN IP4 192.168.1.108
s=Talk
c=IN IP4 192.168.1.108
b=AS:380
t=0 0
a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
m=audio 7076 RTP/AVP 0 8 101
a=rtpmap:101 telephone-event/8000
m=video 9078 RTP/AVP 96 97 98
a=rtpmap:96 VP8/90000
a=rtpmap:97 H264/90000
a=fmtp:97 profile-level-id=42801F
a=rtpmap:98 MP4V-ES/90000
a=fmtp:98 profile-level-id=3
<------------->
--- (14 headers 15 lines) ---
Sending to 94.31.155.12:62646 (NAT)
Using INVITE request as basis request - HJMp6JeMc3
Found peer '1007' for '1007' from 94.31.155.12:62646
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format telephone-event for ID 101
Found RTP video format 96
Found RTP video format 97
Found RTP video format 98
Found video description format VP8 for ID 96
Found video description format H264 for ID 97
Found video description format MP4V-ES for ID 98
Capabilities: us - (h263p|ulaw|alaw|h264), peer - audio=(ulaw|alaw)/video=(vp8|h264|mpeg4)/text=(nothing), combined - (ulaw|alaw|h264)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.108:7076
Peer video RTP is at port 192.168.1.108:9078
Looking for 1004 in from-internal (domain 94.31.202.73)
sip_route_dump: route/path hop: <sip:1007@94.31.155.12:62646>
<--- Transmitting (NAT) to 94.31.155.12:62646 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.108:56495;branch=z9hG4bK.iSLC6T4aV;received=94.31.155.12;rport=62646
From: <sip:1007@94.31.202.73>;tag=s0MPxGOy1
To: sip:1004@94.31.202.73
Call-ID: HJMp6JeMc3
CSeq: 21 INVITE
Server: FPBX-12.0.63(13.2.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:1004@94.31.202.73:5062>
Content-Length: 0
<------------>
<--- SIP read from UDP:94.31.155.12:62646 --->
ACK sip:1004@94.31.202.73 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.108:56495;branch=z9hG4bK.Ji9KrcAks;rport
Call-ID: HJMp6JeMc3
From: <sip:1007@94.31.202.73>;tag=s0MPxGOy1
To: <sip:1004@94.31.202.73>;tag=as4d8f0a34
Contact: <sip:1007@94.31.155.12:62646>;+sip.instance="<urn:uuid:dbedf596-2bcd-4e39-8872-ff75aa2962d5>"
Max-Forwards: 70
CSeq: 20 ACK
<------------->
--- (8 headers 0 lines) ---
[2015-05-22 13:11:58] WARNING[1223][C-0000008f]: ccss.c:1012 ast_set_cc_callback_macro: Usage of cc_callback_macro is deprecated. Please use cc_callback_sub instead.
[2015-05-22 13:11:58] WARNING[1223][C-0000008f]: ccss.c:1012 ast_set_cc_callback_macro: Usage of cc_callback_macro is deprecated. Please use cc_callback_sub instead.
[2015-05-22 13:11:58] WARNING[1223][C-0000008f]: func_presencestate.c:133 presence_read: PRESENCE_STATE unknown
<--- Transmitting (NAT) to 94.31.155.12:62646 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.108:56495;branch=z9hG4bK.iSLC6T4aV;received=94.31.155.12;rport=62646
From: <sip:1007@94.31.202.73>;tag=s0MPxGOy1
To: sip:1004@94.31.202.73;tag=as13b54dfe
Call-ID: HJMp6JeMc3
CSeq: 21 INVITE
Server: FPBX-12.0.63(13.2.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:1004@94.31.202.73:5062>
Content-Length: 0
<------------>
<--- Transmitting (NAT) to 94.31.155.12:62646 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.108:56495;branch=z9hG4bK.iSLC6T4aV;received=94.31.155.12;rport=62646
From: <sip:1007@94.31.202.73>;tag=s0MPxGOy1
To: sip:1004@94.31.202.73;tag=as13b54dfe
Call-ID: HJMp6JeMc3
CSeq: 21 INVITE
Server: FPBX-12.0.63(13.2.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:1004@94.31.202.73:5062>
Content-Length: 0
<------------>
Audio is at 12598
Video is at 94.31.202.73:11830
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding video codec h264 to SDP
Adding video codec h263p to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (NAT) to 94.31.155.12:62646 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.108:56495;branch=z9hG4bK.iSLC6T4aV;received=94.31.155.12;rport=62646
From: <sip:1007@94.31.202.73>;tag=s0MPxGOy1
To: sip:1004@94.31.202.73;tag=as13b54dfe
Call-ID: HJMp6JeMc3
CSeq: 21 INVITE
Server: FPBX-12.0.63(13.2.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:1004@94.31.202.73:5062>
Content-Type: application/sdp
Content-Length: 401
v=0
o=root 126308876 126308876 IN IP4 94.31.202.73
s=Asterisk PBX 13.2.0
c=IN IP4 94.31.202.73
b=CT:1024
t=0 0
m=audio 12598 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv
m=video 11830 RTP/AVP 97 98
a=rtpmap:97 H264/90000
a=fmtp:97 profile-level-id=42801F
a=rtpmap:98 h263-1998/90000
a=sendrecv
<------------>
<--- SIP read from UDP:94.31.155.12:62646 --->
ACK sip:1004@94.31.202.73:5062 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.108:56495;rport;branch=z9hG4bK.ibRyrBRHd
From: <sip:1007@94.31.202.73>;tag=s0MPxGOy1
To: <sip:1004@94.31.202.73>;tag=as13b54dfe
CSeq: 21 ACK
Call-ID: HJMp6JeMc3
Max-Forwards: 70
Authorization: Digest realm="asterisk", nonce="1de76e3a", algorithm=MD5, username="1007", uri="sip:1004@94.31.202.73", response="70efe49423a60eb04ecf6d082e89ed13"
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from UDP:94.31.155.12:62646 --->
BYE sip:1004@94.31.202.73:5062 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.108:56495;branch=z9hG4bK.iC4yTtNmW;rport
From: <sip:1007@94.31.202.73>;tag=s0MPxGOy1
To: <sip:1004@94.31.202.73>;tag=as13b54dfe
CSeq: 22 BYE
Call-ID: HJMp6JeMc3
Max-Forwards: 70
User-Agent: LinphoneIPhone/2.1.0 (belle-sip/1.4.0)
<------------->
--- (8 headers 0 lines) ---
Sending to 94.31.155.12:62646 (NAT)
Scheduling destruction of SIP dialog 'HJMp6JeMc3' in 6400 ms (Method: BYE)
<--- Transmitting (NAT) to 94.31.155.12:62646 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.108:56495;branch=z9hG4bK.iC4yTtNmW;received=94.31.155.12;rport=62646
From: <sip:1007@94.31.202.73>;tag=s0MPxGOy1
To: <sip:1004@94.31.202.73>;tag=as13b54dfe
Call-ID: HJMp6JeMc3
CSeq: 22 BYE
Server: FPBX-12.0.63(13.2.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<------------>
<--- SIP read from UDP:94.31.155.12:62646 --->
<------------->
Asterisk*CLI> sip set debug off
Disconnected from Asterisk server
Asterisk cleanly ending (0).
Executing last minute cleanups
root@Asterisk:/home/huhkoooo#
SIP Debugging Enabled for IP: 94.31.155.12
<--- SIP read from UDP:94.31.155.12:62646 --->
INVITE sip:1004@94.31.202.73 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.108:56495;branch=z9hG4bK.Ji9KrcAks;rport
From: <sip:1007@94.31.202.73>;tag=s0MPxGOy1
To: sip:1004@94.31.202.73
CSeq: 20 INVITE
Call-ID: HJMp6JeMc3
Max-Forwards: 70
Supported: outbound
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
Content-Type: application/sdp
Content-Length: 402
Contact: <sip:1007@94.31.155.12:62646>;+sip.instance="<urn:uuid:dbedf596-2bcd-4e39-8872-ff75aa2962d5>"
User-Agent: LinphoneIPhone/2.1.0 (belle-sip/1.4.0)
v=0
o=1007 3084 2486 IN IP4 192.168.1.108
s=Talk
c=IN IP4 192.168.1.108
b=AS:380
t=0 0
a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
m=audio 7076 RTP/AVP 0 8 101
a=rtpmap:101 telephone-event/8000
m=video 9078 RTP/AVP 96 97 98
a=rtpmap:96 VP8/90000
a=rtpmap:97 H264/90000
a=fmtp:97 profile-level-id=42801F
a=rtpmap:98 MP4V-ES/90000
a=fmtp:98 profile-level-id=3
<------------->
--- (13 headers 15 lines) ---
Sending to 94.31.155.12:62646 (NAT)
Sending to 94.31.155.12:62646 (NAT)
Using INVITE request as basis request - HJMp6JeMc3
Found peer '1007' for '1007' from 94.31.155.12:62646
<--- Reliably Transmitting (NAT) to 94.31.155.12:62646 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.108:56495;branch=z9hG4bK.Ji9KrcAks;received=94.31.155.12;rport=62646
From: <sip:1007@94.31.202.73>;tag=s0MPxGOy1
To: sip:1004@94.31.202.73;tag=as4d8f0a34
Call-ID: HJMp6JeMc3
CSeq: 20 INVITE
Server: FPBX-12.0.63(13.2.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1de76e3a"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'HJMp6JeMc3' in 6400 ms (Method: INVITE)
Retransmitting #1 (NAT) to 94.31.155.12:62646:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.108:56495;branch=z9hG4bK.Ji9KrcAks;received=94.31.155.12;rport=62646
From: <sip:1007@94.31.202.73>;tag=s0MPxGOy1
To: sip:1004@94.31.202.73;tag=as4d8f0a34
Call-ID: HJMp6JeMc3
CSeq: 20 INVITE
Server: FPBX-12.0.63(13.2.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1de76e3a"
Content-Length: 0
---
<--- SIP read from UDP:94.31.155.12:62646 --->
ACK sip:1004@94.31.202.73 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.108:56495;branch=z9hG4bK.Ji9KrcAks;rport
Call-ID: HJMp6JeMc3
From: <sip:1007@94.31.202.73>;tag=s0MPxGOy1
To: <sip:1004@94.31.202.73>;tag=as4d8f0a34
Contact: <sip:1007@94.31.155.12:62646>;+sip.instance="<urn:uuid:dbedf596-2bcd-4e39-8872-ff75aa2962d5>"
Max-Forwards: 70
CSeq: 20 ACK
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from UDP:94.31.155.12:62646 --->
INVITE sip:1004@94.31.202.73 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.108:56495;branch=z9hG4bK.iSLC6T4aV;rport
From: <sip:1007@94.31.202.73>;tag=s0MPxGOy1
To: sip:1004@94.31.202.73
CSeq: 21 INVITE
Call-ID: HJMp6JeMc3
Max-Forwards: 70
Supported: outbound
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
Content-Type: application/sdp
Content-Length: 402
Contact: <sip:1007@94.31.155.12:62646>;+sip.instance="<urn:uuid:dbedf596-2bcd-4e39-8872-ff75aa2962d5>"
User-Agent: LinphoneIPhone/2.1.0 (belle-sip/1.4.0)
Authorization: Digest realm="asterisk", nonce="1de76e3a", algorithm=MD5, username="1007", uri="sip:1004@94.31.202.73", response="70efe49423a60eb04ecf6d082e89ed13"
v=0
o=1007 3084 2486 IN IP4 192.168.1.108
s=Talk
c=IN IP4 192.168.1.108
b=AS:380
t=0 0
a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
m=audio 7076 RTP/AVP 0 8 101
a=rtpmap:101 telephone-event/8000
m=video 9078 RTP/AVP 96 97 98
a=rtpmap:96 VP8/90000
a=rtpmap:97 H264/90000
a=fmtp:97 profile-level-id=42801F
a=rtpmap:98 MP4V-ES/90000
a=fmtp:98 profile-level-id=3
<------------->
--- (14 headers 15 lines) ---
Sending to 94.31.155.12:62646 (NAT)
Using INVITE request as basis request - HJMp6JeMc3
Found peer '1007' for '1007' from 94.31.155.12:62646
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format telephone-event for ID 101
Found RTP video format 96
Found RTP video format 97
Found RTP video format 98
Found video description format VP8 for ID 96
Found video description format H264 for ID 97
Found video description format MP4V-ES for ID 98
Capabilities: us - (h263p|ulaw|alaw|h264), peer - audio=(ulaw|alaw)/video=(vp8|h264|mpeg4)/text=(nothing), combined - (ulaw|alaw|h264)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.108:7076
Peer video RTP is at port 192.168.1.108:9078
Looking for 1004 in from-internal (domain 94.31.202.73)
sip_route_dump: route/path hop: <sip:1007@94.31.155.12:62646>
<--- Transmitting (NAT) to 94.31.155.12:62646 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.108:56495;branch=z9hG4bK.iSLC6T4aV;received=94.31.155.12;rport=62646
From: <sip:1007@94.31.202.73>;tag=s0MPxGOy1
To: sip:1004@94.31.202.73
Call-ID: HJMp6JeMc3
CSeq: 21 INVITE
Server: FPBX-12.0.63(13.2.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:1004@94.31.202.73:5062>
Content-Length: 0
<------------>
<--- SIP read from UDP:94.31.155.12:62646 --->
ACK sip:1004@94.31.202.73 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.108:56495;branch=z9hG4bK.Ji9KrcAks;rport
Call-ID: HJMp6JeMc3
From: <sip:1007@94.31.202.73>;tag=s0MPxGOy1
To: <sip:1004@94.31.202.73>;tag=as4d8f0a34
Contact: <sip:1007@94.31.155.12:62646>;+sip.instance="<urn:uuid:dbedf596-2bcd-4e39-8872-ff75aa2962d5>"
Max-Forwards: 70
CSeq: 20 ACK
<------------->
--- (8 headers 0 lines) ---
[2015-05-22 13:11:58] WARNING[1223][C-0000008f]: ccss.c:1012 ast_set_cc_callback_macro: Usage of cc_callback_macro is deprecated. Please use cc_callback_sub instead.
[2015-05-22 13:11:58] WARNING[1223][C-0000008f]: ccss.c:1012 ast_set_cc_callback_macro: Usage of cc_callback_macro is deprecated. Please use cc_callback_sub instead.
[2015-05-22 13:11:58] WARNING[1223][C-0000008f]: func_presencestate.c:133 presence_read: PRESENCE_STATE unknown
<--- Transmitting (NAT) to 94.31.155.12:62646 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.108:56495;branch=z9hG4bK.iSLC6T4aV;received=94.31.155.12;rport=62646
From: <sip:1007@94.31.202.73>;tag=s0MPxGOy1
To: sip:1004@94.31.202.73;tag=as13b54dfe
Call-ID: HJMp6JeMc3
CSeq: 21 INVITE
Server: FPBX-12.0.63(13.2.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:1004@94.31.202.73:5062>
Content-Length: 0
<------------>
<--- Transmitting (NAT) to 94.31.155.12:62646 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.108:56495;branch=z9hG4bK.iSLC6T4aV;received=94.31.155.12;rport=62646
From: <sip:1007@94.31.202.73>;tag=s0MPxGOy1
To: sip:1004@94.31.202.73;tag=as13b54dfe
Call-ID: HJMp6JeMc3
CSeq: 21 INVITE
Server: FPBX-12.0.63(13.2.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:1004@94.31.202.73:5062>
Content-Length: 0
<------------>
Audio is at 12598
Video is at 94.31.202.73:11830
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding video codec h264 to SDP
Adding video codec h263p to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (NAT) to 94.31.155.12:62646 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.108:56495;branch=z9hG4bK.iSLC6T4aV;received=94.31.155.12;rport=62646
From: <sip:1007@94.31.202.73>;tag=s0MPxGOy1
To: sip:1004@94.31.202.73;tag=as13b54dfe
Call-ID: HJMp6JeMc3
CSeq: 21 INVITE
Server: FPBX-12.0.63(13.2.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:1004@94.31.202.73:5062>
Content-Type: application/sdp
Content-Length: 401
v=0
o=root 126308876 126308876 IN IP4 94.31.202.73
s=Asterisk PBX 13.2.0
c=IN IP4 94.31.202.73
b=CT:1024
t=0 0
m=audio 12598 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv
m=video 11830 RTP/AVP 97 98
a=rtpmap:97 H264/90000
a=fmtp:97 profile-level-id=42801F
a=rtpmap:98 h263-1998/90000
a=sendrecv
<------------>
<--- SIP read from UDP:94.31.155.12:62646 --->
ACK sip:1004@94.31.202.73:5062 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.108:56495;rport;branch=z9hG4bK.ibRyrBRHd
From: <sip:1007@94.31.202.73>;tag=s0MPxGOy1
To: <sip:1004@94.31.202.73>;tag=as13b54dfe
CSeq: 21 ACK
Call-ID: HJMp6JeMc3
Max-Forwards: 70
Authorization: Digest realm="asterisk", nonce="1de76e3a", algorithm=MD5, username="1007", uri="sip:1004@94.31.202.73", response="70efe49423a60eb04ecf6d082e89ed13"
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from UDP:94.31.155.12:62646 --->
BYE sip:1004@94.31.202.73:5062 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.108:56495;branch=z9hG4bK.iC4yTtNmW;rport
From: <sip:1007@94.31.202.73>;tag=s0MPxGOy1
To: <sip:1004@94.31.202.73>;tag=as13b54dfe
CSeq: 22 BYE
Call-ID: HJMp6JeMc3
Max-Forwards: 70
User-Agent: LinphoneIPhone/2.1.0 (belle-sip/1.4.0)
<------------->
--- (8 headers 0 lines) ---
Sending to 94.31.155.12:62646 (NAT)
Scheduling destruction of SIP dialog 'HJMp6JeMc3' in 6400 ms (Method: BYE)
<--- Transmitting (NAT) to 94.31.155.12:62646 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.108:56495;branch=z9hG4bK.iC4yTtNmW;received=94.31.155.12;rport=62646
From: <sip:1007@94.31.202.73>;tag=s0MPxGOy1
To: <sip:1004@94.31.202.73>;tag=as13b54dfe
Call-ID: HJMp6JeMc3
CSeq: 22 BYE
Server: FPBX-12.0.63(13.2.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<------------>
<--- SIP read from UDP:94.31.155.12:62646 --->
<------------->
Asterisk*CLI> sip set debug off
Disconnected from Asterisk server
Asterisk cleanly ending (0).
Executing last minute cleanups
root@Asterisk:/home/huhkoooo#
PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:
Asterisk*CLI> sip set debug peer 1003
SIP Debugging Enabled for IP: 94.31.155.12
<--- SIP read from UDP:94.31.155.12:61613 --->
INVITE sip:1004@94.31.202.73 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.92:39174;branch=z9hG4bK.w~sR55qSX;rport
From: <sip:1003@94.31.202.73>;tag=V~OrSU1~h
To: sip:1004@94.31.202.73
CSeq: 20 INVITE
Call-ID: UuGLTrmKxP
Max-Forwards: 70
Supported: outbound
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
Content-Type: application/sdp
Content-Length: 347
Contact: <sip:1003@94.31.155.12:61613>;+sip.instance="<urn:uuid:cfb8fd37-7195-4d46-b76b-a5c506bc1f78>"
User-Agent: CASTELSip/2.4.0 (belle-sip/1.3.3)
v=0
o=1003 877 3420 IN IP4 192.168.1.92
s=Talk
c=IN IP4 192.168.1.92
t=0 0
a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
m=audio 7076 RTP/AVP 0 9 8 3 101
a=fmtp:0 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
m=video 6200 RTP/AVP 102
a=rtpmap:102 H264/90000
a=fmtp:102 profile-level-id=42801F
<------------->
--- (13 headers 13 lines) ---
Sending to 94.31.155.12:61613 (NAT)
Sending to 94.31.155.12:61613 (NAT)
Using INVITE request as basis request - UuGLTrmKxP
Found peer '1003' for '1003' from 94.31.155.12:61613
<--- Reliably Transmitting (NAT) to 94.31.155.12:61613 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.92:39174;branch=z9hG4bK.w~sR55qSX;received=94.31.155.12;rport=61613
From: <sip:1003@94.31.202.73>;tag=V~OrSU1~h
To: sip:1004@94.31.202.73;tag=as23224214
Call-ID: UuGLTrmKxP
CSeq: 20 INVITE
Server: FPBX-12.0.63(13.2.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="23d4df7f"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'UuGLTrmKxP' in 6400 ms (Method: INVITE)
Retransmitting #1 (NAT) to 94.31.155.12:61613:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.92:39174;branch=z9hG4bK.w~sR55qSX;received=94.31.155.12;rport=61613
From: <sip:1003@94.31.202.73>;tag=V~OrSU1~h
To: sip:1004@94.31.202.73;tag=as23224214
Call-ID: UuGLTrmKxP
CSeq: 20 INVITE
Server: FPBX-12.0.63(13.2.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="23d4df7f"
Content-Length: 0
---
<--- SIP read from UDP:94.31.155.12:61613 --->
ACK sip:1004@94.31.202.73 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.92:39174;branch=z9hG4bK.w~sR55qSX;rport
Call-ID: UuGLTrmKxP
From: <sip:1003@94.31.202.73>;tag=V~OrSU1~h
To: <sip:1004@94.31.202.73>;tag=as23224214
Contact: <sip:1003@94.31.155.12:61613>;+sip.instance="<urn:uuid:cfb8fd37-7195-4d46-b76b-a5c506bc1f78>"
Max-Forwards: 70
CSeq: 20 ACK
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from UDP:94.31.155.12:61613 --->
INVITE sip:1004@94.31.202.73 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.92:39174;branch=z9hG4bK.ym24ewOjT;rport
From: <sip:1003@94.31.202.73>;tag=V~OrSU1~h
To: sip:1004@94.31.202.73
CSeq: 21 INVITE
Call-ID: UuGLTrmKxP
Max-Forwards: 70
Supported: outbound
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
Content-Type: application/sdp
Content-Length: 347
Contact: <sip:1003@94.31.155.12:61613>;+sip.instance="<urn:uuid:cfb8fd37-7195-4d46-b76b-a5c506bc1f78>"
User-Agent: CASTELSip/2.4.0 (belle-sip/1.3.3)
Authorization: Digest realm="asterisk", nonce="23d4df7f", algorithm=MD5, username="1003", uri="sip:1004@94.31.202.73", response="655e7a1d80963679cb28739d72fc9382"
v=0
o=1003 877 3420 IN IP4 192.168.1.92
s=Talk
c=IN IP4 192.168.1.92
t=0 0
a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
m=audio 7076 RTP/AVP 0 9 8 3 101
a=fmtp:0 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
m=video 6200 RTP/AVP 102
a=rtpmap:102 H264/90000
a=fmtp:102 profile-level-id=42801F
<------------->
--- (14 headers 13 lines) ---
Sending to 94.31.155.12:61613 (NAT)
Using INVITE request as basis request - UuGLTrmKxP
Found peer '1003' for '1003' from 94.31.155.12:61613
Found RTP audio format 0
Found RTP audio format 9
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 101
Found audio description format telephone-event for ID 101
Found RTP video format 102
Found video description format H264 for ID 102
Capabilities: us - (ulaw|alaw|h264|h263p), peer - audio=(ulaw|gsm|alaw|g722)/video=(h264)/text=(nothing), combined - (ulaw|alaw|h264)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.92:7076
Peer video RTP is at port 192.168.1.92:6200
Looking for 1004 in from-internal (domain 94.31.202.73)
sip_route_dump: route/path hop: <sip:1003@94.31.155.12:61613>
<--- Transmitting (NAT) to 94.31.155.12:61613 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.92:39174;branch=z9hG4bK.ym24ewOjT;received=94.31.155.12;rport=61613
From: <sip:1003@94.31.202.73>;tag=V~OrSU1~h
To: sip:1004@94.31.202.73
Call-ID: UuGLTrmKxP
CSeq: 21 INVITE
Server: FPBX-12.0.63(13.2.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:1004@94.31.202.73:5062>
Content-Length: 0
<------------>
[2015-05-22 13:39:01] WARNING[4250][C-00000097]: ccss.c:1012 ast_set_cc_callback_macro: Usage of cc_callback_macro is deprecated. Please use cc_callback_sub instead.
<--- SIP read from UDP:94.31.155.12:61613 --->
ACK sip:1004@94.31.202.73 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.92:39174;branch=z9hG4bK.w~sR55qSX;rport
Call-ID: UuGLTrmKxP
From: <sip:1003@94.31.202.73>;tag=V~OrSU1~h
To: <sip:1004@94.31.202.73>;tag=as23224214
Contact: <sip:1003@94.31.155.12:61613>;+sip.instance="<urn:uuid:cfb8fd37-7195-4d46-b76b-a5c506bc1f78>"
Max-Forwards: 70
CSeq: 20 ACK
<------------->
--- (8 headers 0 lines) ---
[2015-05-22 13:39:01] WARNING[4250][C-00000097]: ccss.c:1012 ast_set_cc_callback_macro: Usage of cc_callback_macro is deprecated. Please use cc_callback_sub instead.
[2015-05-22 13:39:01] WARNING[4250][C-00000097]: func_presencestate.c:133 presence_read: PRESENCE_STATE unknown
<--- Transmitting (NAT) to 94.31.155.12:61613 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.92:39174;branch=z9hG4bK.ym24ewOjT;received=94.31.155.12;rport=61613
From: <sip:1003@94.31.202.73>;tag=V~OrSU1~h
To: sip:1004@94.31.202.73;tag=as2d9e6d52
Call-ID: UuGLTrmKxP
CSeq: 21 INVITE
Server: FPBX-12.0.63(13.2.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:1004@94.31.202.73:5062>
Content-Length: 0
<------------>
<--- Transmitting (NAT) to 94.31.155.12:61613 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.92:39174;branch=z9hG4bK.ym24ewOjT;received=94.31.155.12;rport=61613
From: <sip:1003@94.31.202.73>;tag=V~OrSU1~h
To: sip:1004@94.31.202.73;tag=as2d9e6d52
Call-ID: UuGLTrmKxP
CSeq: 21 INVITE
Server: FPBX-12.0.63(13.2.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:1004@94.31.202.73:5062>
Content-Length: 0
<------------>
Audio is at 11630
Video is at 94.31.202.73:13206
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding video codec h264 to SDP
Adding video codec h263p to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (NAT) to 94.31.155.12:61613 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.92:39174;branch=z9hG4bK.ym24ewOjT;received=94.31.155.12;rport=61613
From: <sip:1003@94.31.202.73>;tag=V~OrSU1~h
To: sip:1004@94.31.202.73;tag=as2d9e6d52
Call-ID: UuGLTrmKxP
CSeq: 21 INVITE
Server: FPBX-12.0.63(13.2.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:1004@94.31.202.73:5062>
Content-Type: application/sdp
Content-Length: 406
v=0
o=root 1149449988 1149449988 IN IP4 94.31.202.73
s=Asterisk PBX 13.2.0
c=IN IP4 94.31.202.73
b=CT:1024
t=0 0
m=audio 11630 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv
m=video 13206 RTP/AVP 102 98
a=rtpmap:102 H264/90000
a=fmtp:102 profile-level-id=42801F
a=rtpmap:98 h263-1998/90000
a=sendrecv
<------------>
Retransmitting #1 (NAT) to 94.31.155.12:61613:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.92:39174;branch=z9hG4bK.ym24ewOjT;received=94.31.155.12;rport=61613
From: <sip:1003@94.31.202.73>;tag=V~OrSU1~h
To: sip:1004@94.31.202.73;tag=as2d9e6d52
Call-ID: UuGLTrmKxP
CSeq: 21 INVITE
Server: FPBX-12.0.63(13.2.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:1004@94.31.202.73:5062>
Content-Type: application/sdp
Content-Length: 406
v=0
o=root 1149449988 1149449988 IN IP4 94.31.202.73
s=Asterisk PBX 13.2.0
c=IN IP4 94.31.202.73
b=CT:1024
t=0 0
m=audio 11630 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv
m=video 13206 RTP/AVP 102 98
a=rtpmap:102 H264/90000
a=fmtp:102 profile-level-id=42801F
a=rtpmap:98 h263-1998/90000
a=sendrecv
---
Retransmitting #2 (NAT) to 94.31.155.12:61613:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.92:39174;branch=z9hG4bK.ym24ewOjT;received=94.31.155.12;rport=61613
From: <sip:1003@94.31.202.73>;tag=V~OrSU1~h
To: sip:1004@94.31.202.73;tag=as2d9e6d52
Call-ID: UuGLTrmKxP
CSeq: 21 INVITE
Server: FPBX-12.0.63(13.2.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:1004@94.31.202.73:5062>
Content-Type: application/sdp
Content-Length: 406
v=0
o=root 1149449988 1149449988 IN IP4 94.31.202.73
s=Asterisk PBX 13.2.0
c=IN IP4 94.31.202.73
b=CT:1024
t=0 0
m=audio 11630 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv
m=video 13206 RTP/AVP 102 98
a=rtpmap:102 H264/90000
a=fmtp:102 profile-level-id=42801F
a=rtpmap:98 h263-1998/90000
a=sendrecv
---
<--- SIP read from UDP:94.31.155.12:61613 --->
ACK sip:1004@94.31.202.73:5062 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.92:39174;rport;branch=z9hG4bK.xgvDbaiGH
From: <sip:1003@94.31.202.73>;tag=V~OrSU1~h
To: <sip:1004@94.31.202.73>;tag=as2d9e6d52
CSeq: 21 ACK
Call-ID: UuGLTrmKxP
Max-Forwards: 70
Authorization: Digest realm="asterisk", nonce="23d4df7f", algorithm=MD5, username="1003", uri="sip:1004@94.31.202.73", response="655e7a1d80963679cb28739d72fc9382"
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from UDP:94.31.155.12:61613 --->
ACK sip:1004@94.31.202.73:5062 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.92:39174;branch=z9hG4bK.xgvDbaiGH;rport
From: <sip:1003@94.31.202.73>;tag=V~OrSU1~h
To: <sip:1004@94.31.202.73>;tag=as2d9e6d52
CSeq: 21 ACK
Call-ID: UuGLTrmKxP
Max-Forwards: 70
Authorization: Digest realm="asterisk", nonce="23d4df7f", algorithm=MD5, username="1003", uri="sip:1004@94.31.202.73", response="655e7a1d80963679cb28739d72fc9382"
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from UDP:94.31.155.12:61613 --->
ACK sip:1004@94.31.202.73:5062 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.92:39174;branch=z9hG4bK.xgvDbaiGH;rport
From: <sip:1003@94.31.202.73>;tag=V~OrSU1~h
To: <sip:1004@94.31.202.73>;tag=as2d9e6d52
CSeq: 21 ACK
Call-ID: UuGLTrmKxP
Max-Forwards: 70
Authorization: Digest realm="asterisk", nonce="23d4df7f", algorithm=MD5, username="1003", uri="sip:1004@94.31.202.73", response="655e7a1d80963679cb28739d72fc9382"
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from UDP:94.31.155.12:61613 --->
<------------->
<--- SIP read from UDP:94.31.155.12:61613 --->
INFO sip:1004@94.31.202.73:5062 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.92:39174;branch=z9hG4bK.APC13VA9w;rport
From: <sip:1003@94.31.202.73>;tag=V~OrSU1~h
To: <sip:1004@94.31.202.73>;tag=as2d9e6d52
CSeq: 22 INFO
Call-ID: UuGLTrmKxP
Max-Forwards: 70
Content-Type: application/media_control+xml
Content-Length: 185
User-Agent: CASTELSip/2.4.0 (belle-sip/1.3.3)
<?xml version="1.0" encoding="utf-8" ?><media_control> <vc_primitive> <to_encoder> <picture_fast_update></picture_fast_update> </to_encoder> </vc_primitive></media_control>
<------------->
--- (10 headers 1 lines) ---
Receiving INFO!
<--- Transmitting (NAT) to 94.31.155.12:61613 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.92:39174;branch=z9hG4bK.APC13VA9w;received=94.31.155.12;rport=61613
From: <sip:1003@94.31.202.73>;tag=V~OrSU1~h
To: <sip:1004@94.31.202.73>;tag=as2d9e6d52
Call-ID: UuGLTrmKxP
CSeq: 22 INFO
Server: FPBX-12.0.63(13.2.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<------------>
Really destroying SIP dialog 'bEeZCgl-lN' Method: REGISTER
<--- SIP read from UDP:94.31.155.12:61613 --->
BYE sip:1004@94.31.202.73:5062 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.92:39174;branch=z9hG4bK.IAExuwmZr;rport
From: <sip:1003@94.31.202.73>;tag=V~OrSU1~h
To: <sip:1004@94.31.202.73>;tag=as2d9e6d52
CSeq: 23 BYE
Call-ID: UuGLTrmKxP
Max-Forwards: 70
User-Agent: CASTELSip/2.4.0 (belle-sip/1.3.3)
<------------->
--- (8 headers 0 lines) ---
Sending to 94.31.155.12:61613 (NAT)
Scheduling destruction of SIP dialog 'UuGLTrmKxP' in 6400 ms (Method: BYE)
<--- Transmitting (NAT) to 94.31.155.12:61613 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.92:39174;branch=z9hG4bK.IAExuwmZr;received=94.31.155.12;rport=61613
From: <sip:1003@94.31.202.73>;tag=V~OrSU1~h
To: <sip:1004@94.31.202.73>;tag=as2d9e6d52
Call-ID: UuGLTrmKxP
CSeq: 23 BYE
Server: FPBX-12.0.63(13.2.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<------------>
[2015-05-22 13:39:09] NOTICE[2543]: chan_sip.c:15274 sip_reregister: -- Re-registration for 001346@sip.siplink.pro
[2015-05-22 13:39:09] NOTICE[2543]: chan_sip.c:23839 handle_response_register: Outbound Registration: Expiry for sip.siplink.pro is 120 sec (Scheduling reregistration in 105 s)
Asterisk*CLI> sip set debug off
SIP Debugging Disabled
SIP Debugging Enabled for IP: 94.31.155.12
<--- SIP read from UDP:94.31.155.12:61613 --->
INVITE sip:1004@94.31.202.73 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.92:39174;branch=z9hG4bK.w~sR55qSX;rport
From: <sip:1003@94.31.202.73>;tag=V~OrSU1~h
To: sip:1004@94.31.202.73
CSeq: 20 INVITE
Call-ID: UuGLTrmKxP
Max-Forwards: 70
Supported: outbound
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
Content-Type: application/sdp
Content-Length: 347
Contact: <sip:1003@94.31.155.12:61613>;+sip.instance="<urn:uuid:cfb8fd37-7195-4d46-b76b-a5c506bc1f78>"
User-Agent: CASTELSip/2.4.0 (belle-sip/1.3.3)
v=0
o=1003 877 3420 IN IP4 192.168.1.92
s=Talk
c=IN IP4 192.168.1.92
t=0 0
a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
m=audio 7076 RTP/AVP 0 9 8 3 101
a=fmtp:0 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
m=video 6200 RTP/AVP 102
a=rtpmap:102 H264/90000
a=fmtp:102 profile-level-id=42801F
<------------->
--- (13 headers 13 lines) ---
Sending to 94.31.155.12:61613 (NAT)
Sending to 94.31.155.12:61613 (NAT)
Using INVITE request as basis request - UuGLTrmKxP
Found peer '1003' for '1003' from 94.31.155.12:61613
<--- Reliably Transmitting (NAT) to 94.31.155.12:61613 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.92:39174;branch=z9hG4bK.w~sR55qSX;received=94.31.155.12;rport=61613
From: <sip:1003@94.31.202.73>;tag=V~OrSU1~h
To: sip:1004@94.31.202.73;tag=as23224214
Call-ID: UuGLTrmKxP
CSeq: 20 INVITE
Server: FPBX-12.0.63(13.2.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="23d4df7f"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'UuGLTrmKxP' in 6400 ms (Method: INVITE)
Retransmitting #1 (NAT) to 94.31.155.12:61613:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.92:39174;branch=z9hG4bK.w~sR55qSX;received=94.31.155.12;rport=61613
From: <sip:1003@94.31.202.73>;tag=V~OrSU1~h
To: sip:1004@94.31.202.73;tag=as23224214
Call-ID: UuGLTrmKxP
CSeq: 20 INVITE
Server: FPBX-12.0.63(13.2.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="23d4df7f"
Content-Length: 0
---
<--- SIP read from UDP:94.31.155.12:61613 --->
ACK sip:1004@94.31.202.73 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.92:39174;branch=z9hG4bK.w~sR55qSX;rport
Call-ID: UuGLTrmKxP
From: <sip:1003@94.31.202.73>;tag=V~OrSU1~h
To: <sip:1004@94.31.202.73>;tag=as23224214
Contact: <sip:1003@94.31.155.12:61613>;+sip.instance="<urn:uuid:cfb8fd37-7195-4d46-b76b-a5c506bc1f78>"
Max-Forwards: 70
CSeq: 20 ACK
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from UDP:94.31.155.12:61613 --->
INVITE sip:1004@94.31.202.73 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.92:39174;branch=z9hG4bK.ym24ewOjT;rport
From: <sip:1003@94.31.202.73>;tag=V~OrSU1~h
To: sip:1004@94.31.202.73
CSeq: 21 INVITE
Call-ID: UuGLTrmKxP
Max-Forwards: 70
Supported: outbound
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
Content-Type: application/sdp
Content-Length: 347
Contact: <sip:1003@94.31.155.12:61613>;+sip.instance="<urn:uuid:cfb8fd37-7195-4d46-b76b-a5c506bc1f78>"
User-Agent: CASTELSip/2.4.0 (belle-sip/1.3.3)
Authorization: Digest realm="asterisk", nonce="23d4df7f", algorithm=MD5, username="1003", uri="sip:1004@94.31.202.73", response="655e7a1d80963679cb28739d72fc9382"
v=0
o=1003 877 3420 IN IP4 192.168.1.92
s=Talk
c=IN IP4 192.168.1.92
t=0 0
a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics
m=audio 7076 RTP/AVP 0 9 8 3 101
a=fmtp:0 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
m=video 6200 RTP/AVP 102
a=rtpmap:102 H264/90000
a=fmtp:102 profile-level-id=42801F
<------------->
--- (14 headers 13 lines) ---
Sending to 94.31.155.12:61613 (NAT)
Using INVITE request as basis request - UuGLTrmKxP
Found peer '1003' for '1003' from 94.31.155.12:61613
Found RTP audio format 0
Found RTP audio format 9
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 101
Found audio description format telephone-event for ID 101
Found RTP video format 102
Found video description format H264 for ID 102
Capabilities: us - (ulaw|alaw|h264|h263p), peer - audio=(ulaw|gsm|alaw|g722)/video=(h264)/text=(nothing), combined - (ulaw|alaw|h264)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.92:7076
Peer video RTP is at port 192.168.1.92:6200
Looking for 1004 in from-internal (domain 94.31.202.73)
sip_route_dump: route/path hop: <sip:1003@94.31.155.12:61613>
<--- Transmitting (NAT) to 94.31.155.12:61613 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.92:39174;branch=z9hG4bK.ym24ewOjT;received=94.31.155.12;rport=61613
From: <sip:1003@94.31.202.73>;tag=V~OrSU1~h
To: sip:1004@94.31.202.73
Call-ID: UuGLTrmKxP
CSeq: 21 INVITE
Server: FPBX-12.0.63(13.2.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:1004@94.31.202.73:5062>
Content-Length: 0
<------------>
[2015-05-22 13:39:01] WARNING[4250][C-00000097]: ccss.c:1012 ast_set_cc_callback_macro: Usage of cc_callback_macro is deprecated. Please use cc_callback_sub instead.
<--- SIP read from UDP:94.31.155.12:61613 --->
ACK sip:1004@94.31.202.73 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.92:39174;branch=z9hG4bK.w~sR55qSX;rport
Call-ID: UuGLTrmKxP
From: <sip:1003@94.31.202.73>;tag=V~OrSU1~h
To: <sip:1004@94.31.202.73>;tag=as23224214
Contact: <sip:1003@94.31.155.12:61613>;+sip.instance="<urn:uuid:cfb8fd37-7195-4d46-b76b-a5c506bc1f78>"
Max-Forwards: 70
CSeq: 20 ACK
<------------->
--- (8 headers 0 lines) ---
[2015-05-22 13:39:01] WARNING[4250][C-00000097]: ccss.c:1012 ast_set_cc_callback_macro: Usage of cc_callback_macro is deprecated. Please use cc_callback_sub instead.
[2015-05-22 13:39:01] WARNING[4250][C-00000097]: func_presencestate.c:133 presence_read: PRESENCE_STATE unknown
<--- Transmitting (NAT) to 94.31.155.12:61613 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.92:39174;branch=z9hG4bK.ym24ewOjT;received=94.31.155.12;rport=61613
From: <sip:1003@94.31.202.73>;tag=V~OrSU1~h
To: sip:1004@94.31.202.73;tag=as2d9e6d52
Call-ID: UuGLTrmKxP
CSeq: 21 INVITE
Server: FPBX-12.0.63(13.2.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:1004@94.31.202.73:5062>
Content-Length: 0
<------------>
<--- Transmitting (NAT) to 94.31.155.12:61613 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.92:39174;branch=z9hG4bK.ym24ewOjT;received=94.31.155.12;rport=61613
From: <sip:1003@94.31.202.73>;tag=V~OrSU1~h
To: sip:1004@94.31.202.73;tag=as2d9e6d52
Call-ID: UuGLTrmKxP
CSeq: 21 INVITE
Server: FPBX-12.0.63(13.2.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:1004@94.31.202.73:5062>
Content-Length: 0
<------------>
Audio is at 11630
Video is at 94.31.202.73:13206
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding video codec h264 to SDP
Adding video codec h263p to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (NAT) to 94.31.155.12:61613 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.92:39174;branch=z9hG4bK.ym24ewOjT;received=94.31.155.12;rport=61613
From: <sip:1003@94.31.202.73>;tag=V~OrSU1~h
To: sip:1004@94.31.202.73;tag=as2d9e6d52
Call-ID: UuGLTrmKxP
CSeq: 21 INVITE
Server: FPBX-12.0.63(13.2.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:1004@94.31.202.73:5062>
Content-Type: application/sdp
Content-Length: 406
v=0
o=root 1149449988 1149449988 IN IP4 94.31.202.73
s=Asterisk PBX 13.2.0
c=IN IP4 94.31.202.73
b=CT:1024
t=0 0
m=audio 11630 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv
m=video 13206 RTP/AVP 102 98
a=rtpmap:102 H264/90000
a=fmtp:102 profile-level-id=42801F
a=rtpmap:98 h263-1998/90000
a=sendrecv
<------------>
Retransmitting #1 (NAT) to 94.31.155.12:61613:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.92:39174;branch=z9hG4bK.ym24ewOjT;received=94.31.155.12;rport=61613
From: <sip:1003@94.31.202.73>;tag=V~OrSU1~h
To: sip:1004@94.31.202.73;tag=as2d9e6d52
Call-ID: UuGLTrmKxP
CSeq: 21 INVITE
Server: FPBX-12.0.63(13.2.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:1004@94.31.202.73:5062>
Content-Type: application/sdp
Content-Length: 406
v=0
o=root 1149449988 1149449988 IN IP4 94.31.202.73
s=Asterisk PBX 13.2.0
c=IN IP4 94.31.202.73
b=CT:1024
t=0 0
m=audio 11630 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv
m=video 13206 RTP/AVP 102 98
a=rtpmap:102 H264/90000
a=fmtp:102 profile-level-id=42801F
a=rtpmap:98 h263-1998/90000
a=sendrecv
---
Retransmitting #2 (NAT) to 94.31.155.12:61613:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.92:39174;branch=z9hG4bK.ym24ewOjT;received=94.31.155.12;rport=61613
From: <sip:1003@94.31.202.73>;tag=V~OrSU1~h
To: sip:1004@94.31.202.73;tag=as2d9e6d52
Call-ID: UuGLTrmKxP
CSeq: 21 INVITE
Server: FPBX-12.0.63(13.2.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:1004@94.31.202.73:5062>
Content-Type: application/sdp
Content-Length: 406
v=0
o=root 1149449988 1149449988 IN IP4 94.31.202.73
s=Asterisk PBX 13.2.0
c=IN IP4 94.31.202.73
b=CT:1024
t=0 0
m=audio 11630 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv
m=video 13206 RTP/AVP 102 98
a=rtpmap:102 H264/90000
a=fmtp:102 profile-level-id=42801F
a=rtpmap:98 h263-1998/90000
a=sendrecv
---
<--- SIP read from UDP:94.31.155.12:61613 --->
ACK sip:1004@94.31.202.73:5062 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.92:39174;rport;branch=z9hG4bK.xgvDbaiGH
From: <sip:1003@94.31.202.73>;tag=V~OrSU1~h
To: <sip:1004@94.31.202.73>;tag=as2d9e6d52
CSeq: 21 ACK
Call-ID: UuGLTrmKxP
Max-Forwards: 70
Authorization: Digest realm="asterisk", nonce="23d4df7f", algorithm=MD5, username="1003", uri="sip:1004@94.31.202.73", response="655e7a1d80963679cb28739d72fc9382"
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from UDP:94.31.155.12:61613 --->
ACK sip:1004@94.31.202.73:5062 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.92:39174;branch=z9hG4bK.xgvDbaiGH;rport
From: <sip:1003@94.31.202.73>;tag=V~OrSU1~h
To: <sip:1004@94.31.202.73>;tag=as2d9e6d52
CSeq: 21 ACK
Call-ID: UuGLTrmKxP
Max-Forwards: 70
Authorization: Digest realm="asterisk", nonce="23d4df7f", algorithm=MD5, username="1003", uri="sip:1004@94.31.202.73", response="655e7a1d80963679cb28739d72fc9382"
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from UDP:94.31.155.12:61613 --->
ACK sip:1004@94.31.202.73:5062 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.92:39174;branch=z9hG4bK.xgvDbaiGH;rport
From: <sip:1003@94.31.202.73>;tag=V~OrSU1~h
To: <sip:1004@94.31.202.73>;tag=as2d9e6d52
CSeq: 21 ACK
Call-ID: UuGLTrmKxP
Max-Forwards: 70
Authorization: Digest realm="asterisk", nonce="23d4df7f", algorithm=MD5, username="1003", uri="sip:1004@94.31.202.73", response="655e7a1d80963679cb28739d72fc9382"
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from UDP:94.31.155.12:61613 --->
<------------->
<--- SIP read from UDP:94.31.155.12:61613 --->
INFO sip:1004@94.31.202.73:5062 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.92:39174;branch=z9hG4bK.APC13VA9w;rport
From: <sip:1003@94.31.202.73>;tag=V~OrSU1~h
To: <sip:1004@94.31.202.73>;tag=as2d9e6d52
CSeq: 22 INFO
Call-ID: UuGLTrmKxP
Max-Forwards: 70
Content-Type: application/media_control+xml
Content-Length: 185
User-Agent: CASTELSip/2.4.0 (belle-sip/1.3.3)
<?xml version="1.0" encoding="utf-8" ?><media_control> <vc_primitive> <to_encoder> <picture_fast_update></picture_fast_update> </to_encoder> </vc_primitive></media_control>
<------------->
--- (10 headers 1 lines) ---
Receiving INFO!
<--- Transmitting (NAT) to 94.31.155.12:61613 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.92:39174;branch=z9hG4bK.APC13VA9w;received=94.31.155.12;rport=61613
From: <sip:1003@94.31.202.73>;tag=V~OrSU1~h
To: <sip:1004@94.31.202.73>;tag=as2d9e6d52
Call-ID: UuGLTrmKxP
CSeq: 22 INFO
Server: FPBX-12.0.63(13.2.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<------------>
Really destroying SIP dialog 'bEeZCgl-lN' Method: REGISTER
<--- SIP read from UDP:94.31.155.12:61613 --->
BYE sip:1004@94.31.202.73:5062 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.92:39174;branch=z9hG4bK.IAExuwmZr;rport
From: <sip:1003@94.31.202.73>;tag=V~OrSU1~h
To: <sip:1004@94.31.202.73>;tag=as2d9e6d52
CSeq: 23 BYE
Call-ID: UuGLTrmKxP
Max-Forwards: 70
User-Agent: CASTELSip/2.4.0 (belle-sip/1.3.3)
<------------->
--- (8 headers 0 lines) ---
Sending to 94.31.155.12:61613 (NAT)
Scheduling destruction of SIP dialog 'UuGLTrmKxP' in 6400 ms (Method: BYE)
<--- Transmitting (NAT) to 94.31.155.12:61613 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.92:39174;branch=z9hG4bK.IAExuwmZr;received=94.31.155.12;rport=61613
From: <sip:1003@94.31.202.73>;tag=V~OrSU1~h
To: <sip:1004@94.31.202.73>;tag=as2d9e6d52
Call-ID: UuGLTrmKxP
CSeq: 23 BYE
Server: FPBX-12.0.63(13.2.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<------------>
[2015-05-22 13:39:09] NOTICE[2543]: chan_sip.c:15274 sip_reregister: -- Re-registration for 001346@sip.siplink.pro
[2015-05-22 13:39:09] NOTICE[2543]: chan_sip.c:23839 handle_response_register: Outbound Registration: Expiry for sip.siplink.pro is 120 sec (Scheduling reregistration in 105 s)
Asterisk*CLI> sip set debug off
SIP Debugging Disabled
Re: SIP домофон. Нет видео
а нельзя на 1003 и 1007 поставить одинаковые клиенты ?
1003 - CASTELSip/2.4.0 (belle-sip/1.3.3)
1007 - LinphoneIPhone/2.1.0 (belle-sip/1.4.0)
1003 - CASTELSip/2.4.0 (belle-sip/1.3.3)
1007 - LinphoneIPhone/2.1.0 (belle-sip/1.4.0)
платный суппорт по мере возможностей
Re: SIP домофон. Нет видео
1007 - Iphone. Linphone единственное бесплатное приложение с 264, которое более менее работает
Re: SIP домофон. Нет видео
Вы хотите сказать что нет linphone под Андроид?
Re: SIP домофон. Нет видео
Дак на android все ОК
Re: SIP домофон. Нет видео
Трассировка 1005 (SIP монитор)
Звонок с 1007 на 1005 видео нет в обе стороны
Звонок с 1007 на 1005 видео нет в обе стороны
PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:
SIP Debugging Enabled for IP: 37.29.44.126
[2015-05-22 16:56:02] NOTICE[2574]: chan_sip.c:16703 check_auth: Correct auth, but based on stale nonce received from '<sip:1007@94.31.202.73>;tag=bvQcrHGJ8'
[2015-05-22 16:56:02] NOTICE[2574]: chan_sip.c:29389 sip_poke_peer: Still have a QUALIFY dialog active, deleting
[2015-05-22 16:56:02] NOTICE[2574]: chan_sip.c:23890 handle_response_peerpoke: Peer '1007' is now Reachable. (17ms / 2000ms)
[2015-05-22 16:56:10] WARNING[10194][C-00000011]: ccss.c:1012 ast_set_cc_callback_macro: Usage of cc_callback_macro is deprecated. Please use cc_callback_sub instead.
[2015-05-22 16:56:10] WARNING[10194][C-00000011]: ccss.c:1012 ast_set_cc_callback_macro: Usage of cc_callback_macro is deprecated. Please use cc_callback_sub instead.
[2015-05-22 16:56:10] WARNING[10194][C-00000011]: func_presencestate.c:133 presence_read: PRESENCE_STATE unknown
Audio is at 16838
Video is at 94.31.202.73:19964
Adding codec ulaw to SDP
Adding video codec h264 to SDP
Adding codec alaw to SDP
Adding video codec h263p to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 37.29.44.126
INVITE sip:1005@37.29.44.126:1038;line=75788dcccd620a9 SIP/2.0
Via: SIP/2.0/UDP 94.31.202.73:5062;branch=z9hG4bK0fd4057e;rport
Max-Forwards: 70
From: "1007" <sip:1007@94.31.202.73:5062>;tag=as0f101ba3
To: <sip:1005@37.29.44.126:1038;line=75788dcccd620a9>
Contact: <sip:1007@94.31.202.73:5062>
Call-ID: 327279fc25f62e10073e57634ff3d1e7@94.31.202.73:5062
CSeq: 102 INVITE
User-Agent: FPBX-12.0.63(13.2.0)
Date: Fri, 22 May 2015 11:56:10 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 401
v=0
o=root 187695433 187695433 IN IP4 94.31.202.73
s=Asterisk PBX 13.2.0
c=IN IP4 94.31.202.73
b=CT:1024
t=0 0
m=audio 16838 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv
m=video 19964 RTP/AVP 99 98
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=42801F
a=rtpmap:98 h263-1998/90000
a=sendrecv
---
<--- SIP read from UDP:37.29.44.126:1038 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 94.31.202.73:5062;branch=z9hG4bK0fd4057e;rport=5062
From: "1007" <sip:1007@94.31.202.73:5062>;tag=as0f101ba3
To: <sip:1005@37.29.44.126:1038;line=75788dcccd620a9>
Call-ID: 327279fc25f62e10073e57634ff3d1e7@94.31.202.73:5062
CSeq: 102 INVITE
User-Agent: DnakeVoip v1.0
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from UDP:37.29.44.126:1038 --->
SIP/2.0 101 Dialog Establishement
Via: SIP/2.0/UDP 94.31.202.73:5062;branch=z9hG4bK0fd4057e;rport=5062
From: "1007" <sip:1007@94.31.202.73:5062>;tag=as0f101ba3
To: <sip:1005@37.29.44.126:1038;line=75788dcccd620a9>;tag=1032122909
Call-ID: 327279fc25f62e10073e57634ff3d1e7@94.31.202.73:5062
CSeq: 102 INVITE
Contact: <sip:1005@192.168.1.202:5060>
User-Agent: DnakeVoip v1.0
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
<--- SIP read from UDP:37.29.44.126:1038 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 94.31.202.73:5062;branch=z9hG4bK0fd4057e;rport=5062
From: "1007" <sip:1007@94.31.202.73:5062>;tag=as0f101ba3
To: <sip:1005@37.29.44.126:1038;line=75788dcccd620a9>;tag=1032122909
Call-ID: 327279fc25f62e10073e57634ff3d1e7@94.31.202.73:5062
CSeq: 102 INVITE
Contact: <sip:1005@192.168.1.202:5060>
Content-Type: application/sdp
User-Agent: DnakeVoip v1.0
Content-Length: 384
v=0
o=dnake 2036557841 2036557841 IN IP4 192.168.1.202
s=dnake
c=IN IP4 192.168.1.202
t=0 0
m=audio 6000 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-11
a=sendrecv
m=video 6200 RTP/AVP 102
a=rtpmap:102 H264/90000
a=fmtp:102 profile-level-id=4D0029; packetization-mode=1
a=ex_fmtp:102 2CIF=1
a=sendrecv
<------------->
--- (10 headers 16 lines) ---
sip_route_dump: route/path hop: <sip:1005@192.168.1.202:5060>
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Found RTP video format 102
Found video description format H264 for ID 102
Capabilities: us - (ulaw|alaw|h263p|h264), peer - audio=(ulaw|alaw)/video=(h264)/text=(nothing), combined - (ulaw|alaw|h264)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.202:6000
Peer video RTP is at port 192.168.1.202:6200
<--- SIP read from UDP:37.29.44.126:1038 --->
MESSAGE sip:1007@asterisk SIP/2.0
Via: SIP/2.0/UDP 192.168.1.202:5060;rport;branch=z9hG4bK336624442
From: <sip:1005@asterisk>;tag=1804080153
To: <sip:1007@94.31.202.73:5062>
Call-ID: 989403753
CSeq: 20 MESSAGE
Content-Type: text/plain
Max-Forwards: 70
User-Agent: DnakeVoip v1.0
Content-Length: 197
<params> <app>talk</app> <event>host2id</event> <event_url>/talk/host2id</event_url> <host>1005</host> <building>1</building> <unit>1</unit> <floor>11</floor> <family>11</family>
</params>
<------------->
--- (10 headers 2 lines) ---
Sending to 37.29.44.126:1038 (NAT)
Receiving message!
Looking for 1007 in dpma_message_context (domain asterisk)
<--- Transmitting (NAT) to 37.29.44.126:1038 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.1.202:5060;branch=z9hG4bK336624442;received=37.29.44.126;rport=1038
From: <sip:1005@asterisk>;tag=1804080153
To: <sip:1007@94.31.202.73:5062>;tag=as44f2cfcf
Call-ID: 989403753
CSeq: 20 MESSAGE
Server: FPBX-12.0.63(13.2.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '989403753' in 32000 ms (Method: MESSAGE)
<--- SIP read from UDP:37.29.44.126:1038 --->
jaK
<------------->
<--- SIP read from UDP:37.29.44.126:1038 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 94.31.202.73:5062;branch=z9hG4bK0fd4057e;rport=5062
From: "1007" <sip:1007@94.31.202.73:5062>;tag=as0f101ba3
To: <sip:1005@37.29.44.126:1038;line=75788dcccd620a9>;tag=1032122909
Call-ID: 327279fc25f62e10073e57634ff3d1e7@94.31.202.73:5062
CSeq: 102 INVITE
Contact: <sip:1005@37.29.44.126:1038;line=75788dcccd620a9>
Content-Type: application/sdp
User-Agent: DnakeVoip v1.0
Content-Length: 358
v=0
o=dnake 2036557841 2036557841 IN IP4 192.168.1.202
s=dnake
c=IN IP4 192.168.1.202
t=0 0
m=audio 6000 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-11
a=sendrecv
m=video 6200 RTP/AVP 102
a=rtpmap:102 H264/90000
a=fmtp:102 profile-level-id=4D0029; packetization-mode=1
a=ex_fmtp:102 2CIF=1
a=sendrecv
<------------->
--- (10 headers 15 lines) ---
sip_route_dump: route/path hop: <sip:1005@37.29.44.126:1038;line=75788dcccd620a9>
Transmitting (NAT) to 37.29.44.126
ACK sip:1005@37.29.44.126:1038;line=75788dcccd620a9 SIP/2.0
Via: SIP/2.0/UDP 94.31.202.73:5062;branch=z9hG4bK724de6b8;rport
Max-Forwards: 70
From: "1007" <sip:1007@94.31.202.73:5062>;tag=as0f101ba3
To: <sip:1005@37.29.44.126:1038;line=75788dcccd620a9>;tag=1032122909
Contact: <sip:1007@94.31.202.73:5062>
Call-ID: 327279fc25f62e10073e57634ff3d1e7@94.31.202.73:5062
CSeq: 102 ACK
User-Agent: FPBX-12.0.63(13.2.0)
Content-Length: 0
---
Reliably Transmitting (NAT) to 37.29.44.126
OPTIONS sip:1005@37.29.44.126:1038;line=75788dcccd620a9 SIP/2.0
Via: SIP/2.0/UDP 94.31.202.73:5062;branch=z9hG4bK1b859b8f;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@94.31.202.73:5062>;tag=as12ef1f68
To: <sip:1005@37.29.44.126:1038;line=75788dcccd620a9>
Contact: <sip:Unknown@94.31.202.73:5062>
Call-ID: 3673dee8349cff1a3a5922ca78ab4f28@94.31.202.73:5062
CSeq: 102 OPTIONS
User-Agent: FPBX-12.0.63(13.2.0)
Date: Fri, 22 May 2015 11:56:28 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
---
<--- SIP read from UDP:37.29.44.126:1038 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 94.31.202.73:5062;branch=z9hG4bK1b859b8f;rport=5062
From: "Unknown" <sip:Unknown@94.31.202.73:5062>;tag=as12ef1f68
To: <sip:1005@37.29.44.126:1038;line=75788dcccd620a9>;tag=2143242032
Call-ID: 3673dee8349cff1a3a5922ca78ab4f28@94.31.202.73:5062
CSeq: 102 OPTIONS
User-Agent: DnakeVoip v1.0
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '3673dee8349cff1a3a5922ca78ab4f28@94.31.202.73:5062' Method: OPTIONS
Scheduling destruction of SIP dialog '327279fc25f62e10073e57634ff3d1e7@94.31.202.73:5062' in 7744 ms (Method: INVITE)
Reliably Transmitting (NAT) to 37.29.44.126
BYE sip:1005@37.29.44.126:1038;line=75788dcccd620a9 SIP/2.0
Via: SIP/2.0/UDP 94.31.202.73:5062;branch=z9hG4bK1c450a3d;rport
Max-Forwards: 70
From: "1007" <sip:1007@94.31.202.73:5062>;tag=as0f101ba3
To: <sip:1005@37.29.44.126:1038;line=75788dcccd620a9>;tag=1032122909
Call-ID: 327279fc25f62e10073e57634ff3d1e7@94.31.202.73:5062
CSeq: 103 BYE
User-Agent: FPBX-12.0.63(13.2.0)
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
---
<--- SIP read from UDP:37.29.44.126:1038 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 94.31.202.73:5062;branch=z9hG4bK1c450a3d;rport=5062
From: "1007" <sip:1007@94.31.202.73:5062>;tag=as0f101ba3
To: <sip:1005@37.29.44.126:1038;line=75788dcccd620a9>;tag=1032122909
Call-ID: 327279fc25f62e10073e57634ff3d1e7@94.31.202.73:5062
CSeq: 103 BYE
User-Agent: DnakeVoip v1.0
Content-Length: 0
[2015-05-22 16:56:02] NOTICE[2574]: chan_sip.c:16703 check_auth: Correct auth, but based on stale nonce received from '<sip:1007@94.31.202.73>;tag=bvQcrHGJ8'
[2015-05-22 16:56:02] NOTICE[2574]: chan_sip.c:29389 sip_poke_peer: Still have a QUALIFY dialog active, deleting
[2015-05-22 16:56:02] NOTICE[2574]: chan_sip.c:23890 handle_response_peerpoke: Peer '1007' is now Reachable. (17ms / 2000ms)
[2015-05-22 16:56:10] WARNING[10194][C-00000011]: ccss.c:1012 ast_set_cc_callback_macro: Usage of cc_callback_macro is deprecated. Please use cc_callback_sub instead.
[2015-05-22 16:56:10] WARNING[10194][C-00000011]: ccss.c:1012 ast_set_cc_callback_macro: Usage of cc_callback_macro is deprecated. Please use cc_callback_sub instead.
[2015-05-22 16:56:10] WARNING[10194][C-00000011]: func_presencestate.c:133 presence_read: PRESENCE_STATE unknown
Audio is at 16838
Video is at 94.31.202.73:19964
Adding codec ulaw to SDP
Adding video codec h264 to SDP
Adding codec alaw to SDP
Adding video codec h263p to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 37.29.44.126
INVITE sip:1005@37.29.44.126:1038;line=75788dcccd620a9 SIP/2.0
Via: SIP/2.0/UDP 94.31.202.73:5062;branch=z9hG4bK0fd4057e;rport
Max-Forwards: 70
From: "1007" <sip:1007@94.31.202.73:5062>;tag=as0f101ba3
To: <sip:1005@37.29.44.126:1038;line=75788dcccd620a9>
Contact: <sip:1007@94.31.202.73:5062>
Call-ID: 327279fc25f62e10073e57634ff3d1e7@94.31.202.73:5062
CSeq: 102 INVITE
User-Agent: FPBX-12.0.63(13.2.0)
Date: Fri, 22 May 2015 11:56:10 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 401
v=0
o=root 187695433 187695433 IN IP4 94.31.202.73
s=Asterisk PBX 13.2.0
c=IN IP4 94.31.202.73
b=CT:1024
t=0 0
m=audio 16838 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv
m=video 19964 RTP/AVP 99 98
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=42801F
a=rtpmap:98 h263-1998/90000
a=sendrecv
---
<--- SIP read from UDP:37.29.44.126:1038 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 94.31.202.73:5062;branch=z9hG4bK0fd4057e;rport=5062
From: "1007" <sip:1007@94.31.202.73:5062>;tag=as0f101ba3
To: <sip:1005@37.29.44.126:1038;line=75788dcccd620a9>
Call-ID: 327279fc25f62e10073e57634ff3d1e7@94.31.202.73:5062
CSeq: 102 INVITE
User-Agent: DnakeVoip v1.0
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from UDP:37.29.44.126:1038 --->
SIP/2.0 101 Dialog Establishement
Via: SIP/2.0/UDP 94.31.202.73:5062;branch=z9hG4bK0fd4057e;rport=5062
From: "1007" <sip:1007@94.31.202.73:5062>;tag=as0f101ba3
To: <sip:1005@37.29.44.126:1038;line=75788dcccd620a9>;tag=1032122909
Call-ID: 327279fc25f62e10073e57634ff3d1e7@94.31.202.73:5062
CSeq: 102 INVITE
Contact: <sip:1005@192.168.1.202:5060>
User-Agent: DnakeVoip v1.0
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
<--- SIP read from UDP:37.29.44.126:1038 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 94.31.202.73:5062;branch=z9hG4bK0fd4057e;rport=5062
From: "1007" <sip:1007@94.31.202.73:5062>;tag=as0f101ba3
To: <sip:1005@37.29.44.126:1038;line=75788dcccd620a9>;tag=1032122909
Call-ID: 327279fc25f62e10073e57634ff3d1e7@94.31.202.73:5062
CSeq: 102 INVITE
Contact: <sip:1005@192.168.1.202:5060>
Content-Type: application/sdp
User-Agent: DnakeVoip v1.0
Content-Length: 384
v=0
o=dnake 2036557841 2036557841 IN IP4 192.168.1.202
s=dnake
c=IN IP4 192.168.1.202
t=0 0
m=audio 6000 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-11
a=sendrecv
m=video 6200 RTP/AVP 102
a=rtpmap:102 H264/90000
a=fmtp:102 profile-level-id=4D0029; packetization-mode=1
a=ex_fmtp:102 2CIF=1
a=sendrecv
<------------->
--- (10 headers 16 lines) ---
sip_route_dump: route/path hop: <sip:1005@192.168.1.202:5060>
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Found RTP video format 102
Found video description format H264 for ID 102
Capabilities: us - (ulaw|alaw|h263p|h264), peer - audio=(ulaw|alaw)/video=(h264)/text=(nothing), combined - (ulaw|alaw|h264)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.202:6000
Peer video RTP is at port 192.168.1.202:6200
<--- SIP read from UDP:37.29.44.126:1038 --->
MESSAGE sip:1007@asterisk SIP/2.0
Via: SIP/2.0/UDP 192.168.1.202:5060;rport;branch=z9hG4bK336624442
From: <sip:1005@asterisk>;tag=1804080153
To: <sip:1007@94.31.202.73:5062>
Call-ID: 989403753
CSeq: 20 MESSAGE
Content-Type: text/plain
Max-Forwards: 70
User-Agent: DnakeVoip v1.0
Content-Length: 197
<params> <app>talk</app> <event>host2id</event> <event_url>/talk/host2id</event_url> <host>1005</host> <building>1</building> <unit>1</unit> <floor>11</floor> <family>11</family>
</params>
<------------->
--- (10 headers 2 lines) ---
Sending to 37.29.44.126:1038 (NAT)
Receiving message!
Looking for 1007 in dpma_message_context (domain asterisk)
<--- Transmitting (NAT) to 37.29.44.126:1038 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.1.202:5060;branch=z9hG4bK336624442;received=37.29.44.126;rport=1038
From: <sip:1005@asterisk>;tag=1804080153
To: <sip:1007@94.31.202.73:5062>;tag=as44f2cfcf
Call-ID: 989403753
CSeq: 20 MESSAGE
Server: FPBX-12.0.63(13.2.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '989403753' in 32000 ms (Method: MESSAGE)
<--- SIP read from UDP:37.29.44.126:1038 --->
jaK
<------------->
<--- SIP read from UDP:37.29.44.126:1038 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 94.31.202.73:5062;branch=z9hG4bK0fd4057e;rport=5062
From: "1007" <sip:1007@94.31.202.73:5062>;tag=as0f101ba3
To: <sip:1005@37.29.44.126:1038;line=75788dcccd620a9>;tag=1032122909
Call-ID: 327279fc25f62e10073e57634ff3d1e7@94.31.202.73:5062
CSeq: 102 INVITE
Contact: <sip:1005@37.29.44.126:1038;line=75788dcccd620a9>
Content-Type: application/sdp
User-Agent: DnakeVoip v1.0
Content-Length: 358
v=0
o=dnake 2036557841 2036557841 IN IP4 192.168.1.202
s=dnake
c=IN IP4 192.168.1.202
t=0 0
m=audio 6000 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-11
a=sendrecv
m=video 6200 RTP/AVP 102
a=rtpmap:102 H264/90000
a=fmtp:102 profile-level-id=4D0029; packetization-mode=1
a=ex_fmtp:102 2CIF=1
a=sendrecv
<------------->
--- (10 headers 15 lines) ---
sip_route_dump: route/path hop: <sip:1005@37.29.44.126:1038;line=75788dcccd620a9>
Transmitting (NAT) to 37.29.44.126
ACK sip:1005@37.29.44.126:1038;line=75788dcccd620a9 SIP/2.0
Via: SIP/2.0/UDP 94.31.202.73:5062;branch=z9hG4bK724de6b8;rport
Max-Forwards: 70
From: "1007" <sip:1007@94.31.202.73:5062>;tag=as0f101ba3
To: <sip:1005@37.29.44.126:1038;line=75788dcccd620a9>;tag=1032122909
Contact: <sip:1007@94.31.202.73:5062>
Call-ID: 327279fc25f62e10073e57634ff3d1e7@94.31.202.73:5062
CSeq: 102 ACK
User-Agent: FPBX-12.0.63(13.2.0)
Content-Length: 0
---
Reliably Transmitting (NAT) to 37.29.44.126
OPTIONS sip:1005@37.29.44.126:1038;line=75788dcccd620a9 SIP/2.0
Via: SIP/2.0/UDP 94.31.202.73:5062;branch=z9hG4bK1b859b8f;rport
Max-Forwards: 70
From: "Unknown" <sip:Unknown@94.31.202.73:5062>;tag=as12ef1f68
To: <sip:1005@37.29.44.126:1038;line=75788dcccd620a9>
Contact: <sip:Unknown@94.31.202.73:5062>
Call-ID: 3673dee8349cff1a3a5922ca78ab4f28@94.31.202.73:5062
CSeq: 102 OPTIONS
User-Agent: FPBX-12.0.63(13.2.0)
Date: Fri, 22 May 2015 11:56:28 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
---
<--- SIP read from UDP:37.29.44.126:1038 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 94.31.202.73:5062;branch=z9hG4bK1b859b8f;rport=5062
From: "Unknown" <sip:Unknown@94.31.202.73:5062>;tag=as12ef1f68
To: <sip:1005@37.29.44.126:1038;line=75788dcccd620a9>;tag=2143242032
Call-ID: 3673dee8349cff1a3a5922ca78ab4f28@94.31.202.73:5062
CSeq: 102 OPTIONS
User-Agent: DnakeVoip v1.0
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '3673dee8349cff1a3a5922ca78ab4f28@94.31.202.73:5062' Method: OPTIONS
Scheduling destruction of SIP dialog '327279fc25f62e10073e57634ff3d1e7@94.31.202.73:5062' in 7744 ms (Method: INVITE)
Reliably Transmitting (NAT) to 37.29.44.126
BYE sip:1005@37.29.44.126:1038;line=75788dcccd620a9 SIP/2.0
Via: SIP/2.0/UDP 94.31.202.73:5062;branch=z9hG4bK1c450a3d;rport
Max-Forwards: 70
From: "1007" <sip:1007@94.31.202.73:5062>;tag=as0f101ba3
To: <sip:1005@37.29.44.126:1038;line=75788dcccd620a9>;tag=1032122909
Call-ID: 327279fc25f62e10073e57634ff3d1e7@94.31.202.73:5062
CSeq: 103 BYE
User-Agent: FPBX-12.0.63(13.2.0)
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
---
<--- SIP read from UDP:37.29.44.126:1038 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 94.31.202.73:5062;branch=z9hG4bK1c450a3d;rport=5062
From: "1007" <sip:1007@94.31.202.73:5062>;tag=as0f101ba3
To: <sip:1005@37.29.44.126:1038;line=75788dcccd620a9>;tag=1032122909
Call-ID: 327279fc25f62e10073e57634ff3d1e7@94.31.202.73:5062
CSeq: 103 BYE
User-Agent: DnakeVoip v1.0
Content-Length: 0
