<--- SIP read from UDP:213.87.139.118:43456 --->
INVITE sip:
8083398@ast-ext.domain.ru:22606;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.211.62.42:34548;branch=z9hG4bK-524287-1---ef6980bb9aca468d;rport
Max-Forwards: 70
Contact: <sip:8083399@213.87.139.118:43456;transport=UDP>
To: <sip:
8083398@ast-ext.domain.ru:22606>
From: <sip:
8083399@ast-ext.domain.ru:22606;transport=UDP>;tag=79400958
Call-ID: QSVYEWS4zbI-4_D8YTfMQQ..
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
User-Agent: Zoiper rv2.10.2.0
Allow-Events: presence, kpml, talk
Content-Length: 187
v=0
o=Zoiper 1591085134507 1 IN IP4 213.87.139.118
s=Z
c=IN IP4 213.87.139.118
t=0 0
m=audio 63276 RTP/AVP 0 101 8 3
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
<------------->
--- (13 headers 9 lines) ---
Sending to 213.87.139.118:43456 (NAT)
Sending to 213.87.139.118:43456 (NAT)
Using INVITE request as basis request - QSVYEWS4zbI-4_D8YTfMQQ..
Found peer '8083399' for '8083399' from 213.87.139.118:43456
<--- Reliably Transmitting (NAT) to 213.87.139.118:43456 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.211.62.42:34548;branch=z9hG4bK-524287-1---ef6980bb9aca468d;received=213.87.139.118;rport=43456
From: <sip:
8083399@ast-ext.domain.ru:22606;transport=UDP>;tag=79400958
To: <sip:
8083398@ast-ext.domain.ru:22606>;tag=as60583aab
Call-ID: QSVYEWS4zbI-4_D8YTfMQQ..
CSeq: 1 INVITE
Server: Asterisk PBX 16.2.1~dfsg-2ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6c3c7c10"
Content-Length: 0
<--- SIP read from UDP:213.87.139.118:43456 --->
ACK sip:
8083398@ast-ext.domain.ru:22606;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.211.62.42:34548;branch=z9hG4bK-524287-1---ef6980bb9aca468d;rport
Max-Forwards: 70
To: <sip:
8083398@ast-ext.domain.ru:22606>;tag=as60583aab
From: <sip:
8083399@ast-ext.domain.ru:22606;transport=UDP>;tag=79400958
Call-ID: QSVYEWS4zbI-4_D8YTfMQQ..
CSeq: 1 ACK
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from UDP:213.87.139.118:43456 --->
INVITE sip:
8083398@ast-ext.domain.ru:22606;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 10.211.62.42:34548;branch=z9hG4bK-524287-1---a8a6f1165cce6c8f;rport
Max-Forwards: 70
Contact: <sip:8083399@213.87.139.118:43456;transport=UDP>
To: <sip:
8083398@ast-ext.domain.ru:22606>
From: <sip:
8083399@ast-ext.domain.ru:22606;transport=UDP>;tag=79400958
Call-ID: QSVYEWS4zbI-4_D8YTfMQQ..
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
User-Agent: Zoiper rv2.10.2.0
Authorization: Digest username="8083399",realm="asterisk",nonce="6c3c7c10",uri="sip:
8083398@ast-ext.domain.ru:22606;transport=UDP",response="250636663e5cc98d398b68c56b4fb21a",algorithm=MD5
Allow-Events: presence, kpml, talk
Content-Length: 187
v=0
o=Zoiper 1591085134507 1 IN IP4 213.87.139.118
s=Z
c=IN IP4 213.87.139.118
t=0 0
m=audio 63276 RTP/AVP 0 101 8 3
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
<------------->
--- (14 headers 9 lines) ---
Sending to 213.87.139.118:43456 (NAT)
Using INVITE request as basis request - QSVYEWS4zbI-4_D8YTfMQQ..
Found peer '8083399' for '8083399' from 213.87.139.118:43456
== Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 101
Found RTP audio format 8
Found RTP audio format 3
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw|gsm|h264|h263|h263p), peer - audio=(ulaw|gsm|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|gsm)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 213.87.139.118:63276
Looking for 8083398 in voip_ipp (domain ast-ext.domain.ru)
sip_route_dump: route/path hop: <sip:8083399@213.87.139.118:43456;transport=UDP>
<--- Transmitting (NAT) to 213.87.139.118:43456 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.211.62.42:34548;branch=z9hG4bK-524287-1---a8a6f1165cce6c8f;received=213.87.139.118;rport=43456
From: <sip:
8083399@ast-ext.domain.ru:22606;transport=UDP>;tag=79400958
To: <sip:
8083398@ast-ext.domain.ru:22606>
Call-ID: QSVYEWS4zbI-4_D8YTfMQQ..
CSeq: 2 INVITE
Server: Asterisk PBX 16.2.1~dfsg-2ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:8083398@1.2.3.4:22606>
Content-Length: 0
<------------>
-- Executing [8083398@voip_ipp:1] Dial("SIP/8083399-00000007", "SIP/8083398,5,") in new stack
== Using SIP RTP CoS mark 5
Audio is at 11524
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 10.101.35.131:5062:
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ - почему-то NAT...
INVITE sip:8083398@10.101.35.131:5062 SIP/2.0
Via: SIP/2.0/UDP 10.101.5.20:22606;branch=z9hG4bK3bba6c8e;rport
Max-Forwards: 70
From: "User-EXT" <sip:
8083399@domain.ru:22606>;tag=as4ca10d64
^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ - почему домен обрезало?
To: <sip:8083398@10.101.35.131:5062>
Contact: <sip:8083399@10.101.5.20:22606>
Call-ID:
5c916a4112a03c6d17cdab8241958824@domain.ru
CSeq: 102 INVITE
User-Agent: Asterisk PBX 16.2.1~dfsg-2ubuntu1
Date: Tue, 02 Jun 2020 08:05:36 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 298
v=0
o=root 1845138428 1845138428 IN IP4 10.101.5.20
s=Asterisk PBX 16.2.1~dfsg-2ubuntu1
c=IN IP4 10.101.5.20
t=0 0
m=audio 11524 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv
---
-- Called SIP/8083398
<--- SIP read from UDP:10.101.35.131:5062 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.101.5.20:22606;branch=z9hG4bK3bba6c8e;rport
From: "User-EXT" <sip:
8083399@domain.ru:22606>;tag=as4ca10d64
To: <sip:8083398@10.101.35.131:5062>
Call-ID:
5c916a4112a03c6d17cdab8241958824@domain.ru
CSeq: 102 INVITE
User-Agent: Yealink SIP-T32G 32.70.0.185
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from UDP:10.101.35.131:5062 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.101.5.20:22606;branch=z9hG4bK3bba6c8e;rport
From: "User-EXT" <sip:
8083399@domain.ru:22606>;tag=as4ca10d64
To: <sip:8083398@10.101.35.131:5062>;tag=1081700228
Call-ID:
5c916a4112a03c6d17cdab8241958824@domain.ru
CSeq: 102 INVITE
Contact: <sip:8083398@10.101.35.131:5062>
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
User-Agent: Yealink SIP-T32G 32.70.0.185
Allow-Events: talk,hold,conference,refer,check-sync
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
sip_route_dump: route/path hop: <sip:8083398@10.101.35.131:5062>
-- SIP/8083398-00000008 is ringing
<--- Transmitting (NAT) to 213.87.139.118:43456 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.211.62.42:34548;branch=z9hG4bK-524287-1---a8a6f1165cce6c8f;received=213.87.139.118;rport=43456
From: <sip:
8083399@ast-ext.domain.ru:22606;transport=UDP>;tag=79400958
To: <sip:
8083398@ast-ext.domain.ru:22606>;tag=as62caddd7
Call-ID: QSVYEWS4zbI-4_D8YTfMQQ..
CSeq: 2 INVITE
Server: Asterisk PBX 16.2.1~dfsg-2ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:8083398@1.2.3.4:22606>
Content-Length: 0
<------------>
<--- SIP read from UDP:83.220.237.183:57304 --->
<------------->
<--- SIP read from UDP:10.101.35.131:5062 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.101.5.20:22606;branch=z9hG4bK3bba6c8e;rport
From: "User-EXT" <sip:
8083399@domain.ru:60606>;tag=as4ca10d64
To: <sip:8083398@10.101.35.131:5062>;tag=1081700228
Call-ID:
5c916a4112a03c6d17cdab8241958824@domain.ru
CSeq: 102 INVITE
Contact: <sip:8083398@10.101.35.131:5062>
Content-Type: application/sdp
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
User-Agent: Yealink SIP-T32G 32.70.0.185
Content-Length: 201
v=0
o=- 20067 20067 IN IP4 10.101.35.131
s=SDP data
c=IN IP4 10.101.35.131
t=0 0
m=audio 11134 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=sendrecv
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
<------------->
--- (11 headers 10 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw|gsm|h264|h263|h263p), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.101.35.131:11134
sip_route_dump: route/path hop: <sip:8083398@10.101.35.131:5062>
Transmitting (NAT) to 10.101.35.131:5062:
ACK sip:8083398@10.101.35.131:5062 SIP/2.0
Via: SIP/2.0/UDP 10.101.5.20:22606;branch=z9hG4bK6c61c367;rport
Max-Forwards: 70
From: "User-EXT" <sip:
8083399@domain.ru:22606>;tag=as4ca10d64
To: <sip:8083398@10.101.35.131:5062>;tag=1081700228
Contact: <sip:8083399@10.101.5.20:22606>
Call-ID:
5c916a4112a03c6d17cdab8241958824@domain.ru
CSeq: 102 ACK
User-Agent: Asterisk PBX 16.2.1~dfsg-2ubuntu1
Content-Length: 0
---
-- SIP/8083398-00000008 answered SIP/8083399-00000007
Audio is at 11662
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (NAT) to 213.87.139.118:43456 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.211.62.42:34548;branch=z9hG4bK-524287-1---a8a6f1165cce6c8f;received=213.87.139.118;rport=43456
From: <sip:
8083399@ast-ext.quadra.ru:22606;transport=UDP>;tag=79400958
To: <sip:
8083398@ast-ext.quadra.ru:22606>;tag=as62caddd7
Call-ID: QSVYEWS4zbI-4_D8YTfMQQ..
CSeq: 2 INVITE
Server: Asterisk PBX 16.2.1~dfsg-2ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:8083398@1.2.3.4:22606>
Content-Type: application/sdp
Content-Length: 302
v=0
o=root 1447938726 1447938726 IN IP4 1.2.3.4
s=Asterisk PBX 16.2.1~dfsg-2ubuntu1
c=IN IP4 1.2.3.4
t=0 0
m=audio 11662 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv
<------------>
-- Channel SIP/8083398-00000008 joined 'simple_bridge' basic-bridge <ce0964d3-6012-4069-94c0-5ef2d8e3de92>
-- Channel SIP/8083399-00000007 joined 'simple_bridge' basic-bridge <ce0964d3-6012-4069-94c0-5ef2d8e3de92>
<--- SIP read from UDP:213.87.139.118:43456 --->
ACK sip:8083398@1.2.3.4:22606 SIP/2.0
Via: SIP/2.0/UDP 10.211.62.42:34548;branch=z9hG4bK-524287-1---261feb30a5caa034;rport
Max-Forwards: 70
Contact: <sip:8083399@213.87.139.118:43456;transport=UDP>
To: <sip:
8083398@ast-ext.domain.ru:22606>;tag=as62caddd7
From: <sip:
8083399@ast-ext.domain.ru:22606;transport=UDP>;tag=79400958
Call-ID: QSVYEWS4zbI-4_D8YTfMQQ..
CSeq: 2 ACK
User-Agent: Zoiper rv2.10.2.0
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
<--- SIP read from UDP:213.87.139.118:43456 --->
<--- SIP read from UDP:10.101.35.131:5062 --->
BYE sip:8083399@10.101.5.20:22606 SIP/2.0
Via: SIP/2.0/UDP 10.101.35.131:5062;branch=z9hG4bK1743543320
From: <sip:8083398@10.101.35.131:5062>;tag=1081700228
To: "User-EXT" <sip:
8083399@domain.ru:22606>;tag=as4ca10d64
Call-ID:
5c916a4112a03c6d17cdab8241958824@domain.ru
CSeq: 103 BYE
Contact: <sip:8083398@10.101.35.131:5062>
Max-Forwards: 70
User-Agent: Yealink SIP-T32G 32.70.0.185
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Sending to 10.101.35.131:5062 (NAT)
Scheduling destruction of SIP dialog '
5c916a4112a03c6d17cdab8241958824@domain.ru' in 6400 ms (Method: BYE)
<--- Transmitting (NAT) to 10.101.35.131:5062 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.101.35.131:5062;branch=z9hG4bK1743543320;received=10.101.35.131;rport=5062
From: <sip:8083398@10.101.35.131:5062>;tag=1081700228
To: "User-EXT" <sip:
8083399@domain.ru:22606>;tag=as4ca10d64
Call-ID:
5c916a4112a03c6d17cdab8241958824@domain.ru
CSeq: 103 BYE
Server: Asterisk PBX 16.2.1~dfsg-2ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<------------>
-- Channel SIP/8083398-00000008 left 'native_rtp' basic-bridge <ce0964d3-6012-4069-94c0-5ef2d8e3de92>
-- Channel SIP/8083399-00000007 left 'native_rtp' basic-bridge <ce0964d3-6012-4069-94c0-5ef2d8e3de92>
== Spawn extension (voip_ipp, 8083398, 1) exited non-zero on 'SIP/8083399-00000007'
Scheduling destruction of SIP dialog 'QSVYEWS4zbI-4_D8YTfMQQ..' in 6400 ms (Method: ACK)
Reliably Transmitting (NAT) to 213.87.139.118:43456:
BYE sip:8083399@213.87.139.118:43456;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 1.2.3.4:22606;branch=z9hG4bK6e66e96f;rport
Max-Forwards: 70
From: <sip:
8083398@ast-ext.domain.ru:22606>;tag=as62caddd7
To: <sip:
8083399@ast-ext.domain.ru:22606;transport=UDP>;tag=79400958
Call-ID: QSVYEWS4zbI-4_D8YTfMQQ..
CSeq: 102 BYE
User-Agent: Asterisk PBX 16.2.1~dfsg-2ubuntu1
Proxy-Authorization: Digest username="8083399", realm="asterisk", algorithm=MD5, uri="sip:ast-ext.domain.ru", nonce="6c3c7c10", response="9b923e7dd0417c1c5a1d59b0414cc835"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
---
<--- SIP read from UDP:213.87.139.118:43456 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 1.2.3.4:22606;branch=z9hG4bK6e66e96f;rport=22606
Contact: <sip:8083399@213.87.139.118:43456;transport=UDP>
To: <sip:
8083399@ast-ext.domain.ru:22606;transport=UDP>;tag=79400958
From: <sip:
8083398@ast-ext.domain.ru:22606>;tag=as62caddd7
Call-ID: QSVYEWS4zbI-4_D8YTfMQQ..
CSeq: 102 BYE
User-Agent: Zoiper rv2.10.2.0
Content-Length: 0