Проблема у меня такого характера...
Есть связка АТС NCP1000 и Asterisk (FreePBX). На Asterisk приходят внешние линии от провайдера телефонии по SIP. Все звонки в город клиенты Panasonic выполняют через Asterisk (без 9 через TIE линии). Так же есть некоторое количество клиентов Asterisk. Все sip-телефоны.
Столкнулся с такой проблемой... Сам понять не могу... квалификации маловато...
Если звонить на любые номера по шаблонам межгорода, т.е. 8<код><номер телефона>, то все работает отлично и через Panasonic и через Asterisk, но если позвонить через Panasonic по международке (например 81038ХХХХХХХХХХ) получаю 503ю ошибку и сообщение о том, что все линии заняты. При этом если позвонить на этот же номер с клиента Asterisk, то звонок проходит отлично.
Всю голову сломал, а разницы между логами удачного соединения и проваленного увидеть не могу. Вроде все, что нужно для удачного прохождения звонка присутствует, а все равно отлуп получаю...
<--- SIP read from UDP:10.59.0.101:53139 --->
INVITE sip:81038ХХХХХХХХХХ@<DNS имя сервера> SIP/2.0
Via: SIP/2.0/UDP 10.59.0.101:53139;branch=z9hG4bK14a9918c-9c1dde1c;rport
From: sip:2000@<DNS имя сервера>;tag=6f2ca2d5-114bc0da
To: sip:81038ХХХХХХХХХХ@<DNS имя сервера>
Call-ID: 9c76bfaa-90a69640-11f9d18f-9246b338
CSeq: 672 INVITE
Contact: <sip:2000@10.59.0.101:53139;x-reg=33DD1FC6297BA6FB>;audio
Content-Type: application/sdp
Content-Length: 363
User-Agent: Sipnetic/1.0.36 Android
Supported: 100rel,timer,replaces,tdialog
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,UPDATE,SUBSCRIBE,NOTIFY,REFER,PRACK,MESSAGE
Session-Expires: 300
Max-Forwards: 70
v=0
o=- 699186519 699186519 IN IP4 10.59.0.101
s=-
c=IN IP4 10.59.0.101
t=0 0
m=audio 26538 RTP/AVP 96 9 97 3 8 0 101
a=rtpmap:96 opus/48000/2
a=rtpmap:9 G722/8000
a=rtpmap:97 Speex/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
a=fmtp:96 useinbandfec=0
a=fmtp:101 0-15
a=rtcp-mux
<------------->
--- (14 headers 17 lines) ---
Sending to 10.59.0.101:53139 (NAT)
Sending to 10.59.0.101:53139 (NAT)
Using INVITE request as basis request - 9c76bfaa-90a69640-11f9d18f-9246b338
Found peer '2000' for '2000' from 10.59.0.101:53139
<--- Reliably Transmitting (NAT) to 10.59.0.101:53139 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.59.0.101:53139;branch=z9hG4bK14a9918c-9c1dde1c;received=10.59.0.101;rport=53139
From: sip:2000@<DNS имя сервера>;tag=6f2ca2d5-114bc0da
To: sip:81038ХХХХХХХХХХ@<DNS имя сервера>;tag=as20f0580c
Call-ID: 9c76bfaa-90a69640-11f9d18f-9246b338
CSeq: 672 INVITE
Server: FPBX-15.0.16.81(16.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="56535559"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '9c76bfaa-90a69640-11f9d18f-9246b338' in 6400 ms (Method: INVITE)
<--- SIP read from UDP:10.59.0.101:53139 --->
ACK sip:81038ХХХХХХХХХХ@<DNS имя сервера> SIP/2.0
Via: SIP/2.0/UDP 10.59.0.101:53139;branch=z9hG4bK14a9918c-9c1dde1c;rport
From: sip:2000@<DNS имя сервера>;tag=6f2ca2d5-114bc0da
To: sip:81038ХХХХХХХХХХ@<DNS имя сервера>;tag=as20f0580c
Call-ID: 9c76bfaa-90a69640-11f9d18f-9246b338
CSeq: 672 ACK
Max-Forwards: 70
<------------->
--- (7 headers 0 lines) ---
<--- SIP read from UDP:10.59.0.101:53139 --->
INVITE sip:81038ХХХХХХХХХХ@<DNS имя сервера> SIP/2.0
Via: SIP/2.0/UDP 10.59.0.101:53139;branch=z9hG4bKaff6df32-169f906e;rport
From: sip:2000@<DNS имя сервера>;tag=6f2ca2d5-114bc0da
To: sip:81038ХХХХХХХХХХ@<DNS имя сервера>
Call-ID: 9c76bfaa-90a69640-11f9d18f-9246b338
CSeq: 673 INVITE
Contact: <sip:2000@10.59.0.101:53139;x-reg=33DD1FC6297BA6FB>;audio
Content-Type: application/sdp
Content-Length: 363
User-Agent: Sipnetic/1.0.36 Android
Supported: 100rel,timer,replaces,tdialog
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,UPDATE,SUBSCRIBE,NOTIFY,REFER,PRACK,MESSAGE
Session-Expires: 300
Authorization: Digest username="2000",realm="asterisk",nonce="56535559",opaque="",uri="sip:81038ХХХХХХХХХХ@<DNS имя сервера>",algorithm=MD5,response="6ff2967b91ab53ba06266d034da06d24"
Max-Forwards: 70
v=0
o=- 699186519 699186519 IN IP4 10.59.0.101
s=-
c=IN IP4 10.59.0.101
t=0 0
m=audio 26538 RTP/AVP 96 9 97 3 8 0 101
a=rtpmap:96 opus/48000/2
a=rtpmap:9 G722/8000
a=rtpmap:97 Speex/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
a=fmtp:96 useinbandfec=0
a=fmtp:101 0-15
a=rtcp-mux
<------------->
--- (15 headers 17 lines) ---
Sending to 10.59.0.101:53139 (NAT)
Using INVITE request as basis request - 9c76bfaa-90a69640-11f9d18f-9246b338
Found peer '2000' for '2000' from 10.59.0.101:53139
Got SDP version 699186519 and unique parts [- 699186519 IN IP4 10.59.0.101]
Found RTP audio format 96
Found RTP audio format 9
Found RTP audio format 97
Found RTP audio format 3
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 101
Found audio description format opus for ID 96
Found audio description format G722 for ID 9
Found audio description format Speex for ID 97
Found audio description format GSM for ID 3
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw|gsm|g726|g722), peer - audio=(ulaw|gsm|alaw|g722|opus|speex)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|gsm|g722)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.59.0.101:26538
Looking for 81038ХХХХХХХХХХ in from-internal (domain <DNS имя сервера>)
sip_route_dump: route/path hop: <sip:2000@10.59.0.101:53139;x-reg=33DD1FC6297BA6FB>
<--- Transmitting (NAT) to 10.59.0.101:53139 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.59.0.101:53139;branch=z9hG4bKaff6df32-169f906e;received=10.59.0.101;rport=53139
From: sip:2000@<DNS имя сервера>;tag=6f2ca2d5-114bc0da
To: sip:81038ХХХХХХХХХХ@<DNS имя сервера>
Call-ID: 9c76bfaa-90a69640-11f9d18f-9246b338
CSeq: 673 INVITE
Server: FPBX-15.0.16.81(16.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 300;refresher=uas
Contact: <sip:81038ХХХХХХХХХХ@10.60.0.4:5060>
Content-Length: 0
<------------>
<--- SIP read from UDP:10.59.0.101:53139 --->
PUBLISH sip:2000@<DNS имя сервера> SIP/2.0
Via: SIP/2.0/UDP 10.59.0.101:53139;branch=z9hG4bK0675224b-2b3efba3;rport
From: sip:2000@<DNS имя сервера>;tag=aecf8486-c4ff2c95
To: sip:2000@<DNS имя сервера>
Call-ID: c2033fa0-af28ab87-1c4cbbb4-0dd9a5b5
CSeq: 57 PUBLISH
Content-Type: application/pidf+xml
Content-Length: 680
Event: presence
Expires: 600
User-Agent: Sipnetic/1.0.36 Android
Max-Forwards: 70
<?xml version="1.0" encoding="UTF-8"?><presence xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid" xmlns:sc="urn:ietf:params:xml:ns:pidf:caps" entity="sip:2000@<DNS имя сервера>"><tuple id="TC0wctkvRGeFGPK85pg8LF0EXvoK9GnUL"><status><basic>open</basic><rpid:activities><rpid:on-the-phone/></rpid:activities></status><sc:servcaps><sc:audio>true</sc:audio><sc:video>false</sc:video><sc:message>true</sc:message></sc:servcaps><timestamp>2020-12-05T11:14:02Z</timestamp></tuple><dm:person id="PKTUTBC4G0SQr963jHfLtQIWwjw1qV7D3"><rpid:activities><rpid:on-the-phone/></rpid:activities></dm:person></presence>
<------------->
--- (12 headers 1 lines) ---
Sending to 10.59.0.101:53139 (NAT)
<--- Transmitting (NAT) to 10.59.0.101:53139 --->
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP 10.59.0.101:53139;branch=z9hG4bK0675224b-2b3efba3;received=10.59.0.101;rport=53139
From: sip:2000@<DNS имя сервера>;tag=aecf8486-c4ff2c95
To: sip:2000@<DNS имя сервера>;tag=as21eb3601
Call-ID: c2033fa0-af28ab87-1c4cbbb4-0dd9a5b5
CSeq: 57 PUBLISH
Server: FPBX-15.0.16.81(16.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<------------>
Really destroying SIP dialog 'c2033fa0-af28ab87-1c4cbbb4-0dd9a5b5' Method: PUBLISH
Audio is at 10088
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to <PROVIDER SRV IP ADDRESS>:5060:
INVITE sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS> SIP/2.0
Via: SIP/2.0/UDP <EXT SRV IP ADDRESS>:5060;branch=z9hG4bK1b3d4ee0;rport
Max-Forwards: 70
From: <sip:495ХХХХХХХ@<EXT SRV IP ADDRESS>>;tag=as48c8fdc2
To: <sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS>>
Contact: <sip:495ХХХХХХХ@<EXT SRV IP ADDRESS>:5060>
Call-ID: 1df4d8c3167affd83fec830c1aa2d6c7@<EXT SRV IP ADDRESS>:5060
CSeq: 102 INVITE
User-Agent: FPBX-15.0.16.81(16.13.0)
Date: Sat, 05 Dec 2020 11:13:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 276
v=0
o=root 274566906 274566906 IN IP4 <EXT SRV IP ADDRESS>
s=Asterisk PBX 16.13.0
c=IN IP4 <EXT SRV IP ADDRESS>
t=0 0
m=audio 10088 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=ptime:20
a=maxptime:150
a=sendrecv
---
<--- SIP read from UDP:<PROVIDER SRV IP ADDRESS>:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP <EXT SRV IP ADDRESS>:5060;branch=z9hG4bK1b3d4ee0;received=<EXT SRV IP ADDRESS>;rport=5060
From: <sip:495ХХХХХХХ@<EXT SRV IP ADDRESS>>;tag=as48c8fdc2
To: <sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS>>;tag=as1173eb19
Call-ID: 1df4d8c3167affd83fec830c1aa2d6c7@<EXT SRV IP ADDRESS>:5060
CSeq: 102 INVITE
Server: FPBX-2.11.0(11.17.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="74885e74"
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
Transmitting (NAT) to <PROVIDER SRV IP ADDRESS>:5060:
ACK sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS> SIP/2.0
Via: SIP/2.0/UDP <EXT SRV IP ADDRESS>:5060;branch=z9hG4bK1b3d4ee0;rport
Max-Forwards: 70
From: <sip:495ХХХХХХХ@<EXT SRV IP ADDRESS>>;tag=as48c8fdc2
To: <sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS>>;tag=as1173eb19
Contact: <sip:495ХХХХХХХ@<EXT SRV IP ADDRESS>:5060>
Call-ID: 1df4d8c3167affd83fec830c1aa2d6c7@<EXT SRV IP ADDRESS>:5060
CSeq: 102 ACK
User-Agent: FPBX-15.0.16.81(16.13.0)
Content-Length: 0
---
Audio is at 10088
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to <PROVIDER SRV IP ADDRESS>:5060:
INVITE sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS> SIP/2.0
Via: SIP/2.0/UDP <EXT SRV IP ADDRESS>:5060;branch=z9hG4bK14b315af;rport
Max-Forwards: 70
From: <sip:495ХХХХХХХ@<EXT SRV IP ADDRESS>>;tag=as48c8fdc2
To: <sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS>>
Contact: <sip:495ХХХХХХХ@<EXT SRV IP ADDRESS>:5060>
Call-ID: 1df4d8c3167affd83fec830c1aa2d6c7@<EXT SRV IP ADDRESS>:5060
CSeq: 103 INVITE
User-Agent: FPBX-15.0.16.81(16.13.0)
Authorization: Digest username="495ХХХХХХХ", realm="asterisk", algorithm=MD5, uri="sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS>", nonce="74885e74", response="bc5903222b7b8c2a8a0753f603d0259d"
Date: Sat, 05 Dec 2020 11:13:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 276
v=0
o=root 274566906 274566907 IN IP4 <EXT SRV IP ADDRESS>
s=Asterisk PBX 16.13.0
c=IN IP4 <EXT SRV IP ADDRESS>
t=0 0
m=audio 10088 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=ptime:20
a=maxptime:150
a=sendrecv
---
<--- SIP read from UDP:<PROVIDER SRV IP ADDRESS>:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP <EXT SRV IP ADDRESS>:5060;branch=z9hG4bK14b315af;received=<EXT SRV IP ADDRESS>;rport=5060
From: <sip:495ХХХХХХХ@<EXT SRV IP ADDRESS>>;tag=as48c8fdc2
To: <sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS>>
Call-ID: 1df4d8c3167affd83fec830c1aa2d6c7@<EXT SRV IP ADDRESS>:5060
CSeq: 103 INVITE
Server: FPBX-2.11.0(11.17.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS>:5060>
Content-Length: 0
<------------->
--- (12 headers 0 lines) ---
<--- SIP read from UDP:<PROVIDER SRV IP ADDRESS>:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP <EXT SRV IP ADDRESS>:5060;branch=z9hG4bK14b315af;received=<EXT SRV IP ADDRESS>;rport=5060
From: <sip:495ХХХХХХХ@<EXT SRV IP ADDRESS>>;tag=as48c8fdc2
To: <sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS>>;tag=as7b2845cb
Call-ID: 1df4d8c3167affd83fec830c1aa2d6c7@<EXT SRV IP ADDRESS>:5060
CSeq: 103 INVITE
Server: FPBX-2.11.0(11.17.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS>:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 260
v=0
o=root 962419659 962419659 IN IP4 <PROVIDER SRV IP ADDRESS>
s=Asterisk PBX 11.17.1
c=IN IP4 <PROVIDER SRV IP ADDRESS>
t=0 0
m=audio 16748 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 12 lines) ---
sip_route_dump: route/path hop: <sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS>:5060>
Got SDP version 962419659 and unique parts [root 962419659 IN IP4 <PROVIDER SRV IP ADDRESS>]
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - (alaw|ulaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (alaw|ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port <PROVIDER SRV IP ADDRESS>:16748
Audio is at 17346
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding codec g722 to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Transmitting (NAT) to 10.59.0.101:53139 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.59.0.101:53139;branch=z9hG4bKaff6df32-169f906e;received=10.59.0.101;rport=53139
From: sip:2000@<DNS имя сервера>;tag=6f2ca2d5-114bc0da
To: sip:81038ХХХХХХХХХХ@<DNS имя сервера>;tag=as054b4b7b
Call-ID: 9c76bfaa-90a69640-11f9d18f-9246b338
CSeq: 673 INVITE
Server: FPBX-15.0.16.81(16.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 300;refresher=uas
Contact: <sip:81038ХХХХХХХХХХ@10.60.0.4:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 317
v=0
o=root 1617966580 1617966580 IN IP4 10.60.0.4
s=Asterisk PBX 16.13.0
c=IN IP4 10.60.0.4
t=0 0
m=audio 17346 RTP/AVP 0 8 3 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
<------------>
<--- SIP read from UDP:10.59.0.101:53139 --->
CANCEL sip:81038ХХХХХХХХХХ@<DNS имя сервера> SIP/2.0
Via: SIP/2.0/UDP 10.59.0.101:53139;branch=z9hG4bKaff6df32-169f906e
From: sip:2000@<DNS имя сервера>;tag=6f2ca2d5-114bc0da
To: sip:81038ХХХХХХХХХХ@<DNS имя сервера>
Call-ID: 9c76bfaa-90a69640-11f9d18f-9246b338
CSeq: 673 CANCEL
Content-Length: 0
Authorization: Digest username="2000",realm="asterisk",nonce="56535559",opaque="",uri="sip:81038ХХХХХХХХХХ@<DNS имя сервера>",algorithm=MD5,response="6ff2967b91ab53ba06266d034da06d24"
Max-Forwards: 70
<------------->
--- (9 headers 0 lines) ---
Sending to 10.59.0.101:53139 (NAT)
<--- Reliably Transmitting (NAT) to 10.59.0.101:53139 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 10.59.0.101:53139;branch=z9hG4bKaff6df32-169f906e;received=10.59.0.101;rport=53139
From: sip:2000@<DNS имя сервера>;tag=6f2ca2d5-114bc0da
To: sip:81038ХХХХХХХХХХ@<DNS имя сервера>;tag=as054b4b7b
Call-ID: 9c76bfaa-90a69640-11f9d18f-9246b338
CSeq: 673 INVITE
Server: FPBX-15.0.16.81(16.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<------------>
<--- Transmitting (NAT) to 10.59.0.101:53139 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.59.0.101:53139;branch=z9hG4bKaff6df32-169f906e;received=10.59.0.101;rport=53139
From: sip:2000@<DNS имя сервера>;tag=6f2ca2d5-114bc0da
To: sip:81038ХХХХХХХХХХ@<DNS имя сервера>;tag=as054b4b7b
Call-ID: 9c76bfaa-90a69640-11f9d18f-9246b338
CSeq: 673 CANCEL
Server: FPBX-15.0.16.81(16.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '1df4d8c3167affd83fec830c1aa2d6c7@<EXT SRV IP ADDRESS>:5060' in 6400 ms (Method: INVITE)
Reliably Transmitting (NAT) to <PROVIDER SRV IP ADDRESS>:5060:
CANCEL sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS> SIP/2.0
Via: SIP/2.0/UDP <EXT SRV IP ADDRESS>:5060;branch=z9hG4bK14b315af;rport
Max-Forwards: 70
From: <sip:495ХХХХХХХ@<EXT SRV IP ADDRESS>>;tag=as48c8fdc2
To: <sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS>>
Call-ID: 1df4d8c3167affd83fec830c1aa2d6c7@<EXT SRV IP ADDRESS>:5060
CSeq: 103 CANCEL
User-Agent: FPBX-15.0.16.81(16.13.0)
Content-Length: 0
---
Scheduling destruction of SIP dialog '1df4d8c3167affd83fec830c1aa2d6c7@<EXT SRV IP ADDRESS>:5060' in 6400 ms (Method: INVITE)
<--- SIP read from UDP:<PROVIDER SRV IP ADDRESS>:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP <EXT SRV IP ADDRESS>:5060;branch=z9hG4bK14b315af;received=<EXT SRV IP ADDRESS>;rport=5060
From: <sip:495ХХХХХХХ@<EXT SRV IP ADDRESS>>;tag=as48c8fdc2
To: <sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS>>;tag=as7b2845cb
Call-ID: 1df4d8c3167affd83fec830c1aa2d6c7@<EXT SRV IP ADDRESS>:5060
CSeq: 103 INVITE
Server: FPBX-2.11.0(11.17.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Transmitting (NAT) to <PROVIDER SRV IP ADDRESS>:5060:
ACK sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS>:5060 SIP/2.0
Via: SIP/2.0/UDP <EXT SRV IP ADDRESS>:5060;branch=z9hG4bK14b315af;rport
Max-Forwards: 70
From: <sip:495ХХХХХХХ@<EXT SRV IP ADDRESS>>;tag=as48c8fdc2
To: <sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS>>;tag=as7b2845cb
Contact: <sip:495ХХХХХХХ@<EXT SRV IP ADDRESS>:5060>
Call-ID: 1df4d8c3167affd83fec830c1aa2d6c7@<EXT SRV IP ADDRESS>:5060
CSeq: 103 ACK
User-Agent: FPBX-15.0.16.81(16.13.0)
Content-Length: 0
---
Scheduling destruction of SIP dialog '1df4d8c3167affd83fec830c1aa2d6c7@<EXT SRV IP ADDRESS>:5060' in 6400 ms (Method: INVITE)
<--- SIP read from UDP:<PROVIDER SRV IP ADDRESS>:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP <EXT SRV IP ADDRESS>:5060;branch=z9hG4bK14b315af;received=<EXT SRV IP ADDRESS>;rport=5060
From: <sip:495ХХХХХХХ@<EXT SRV IP ADDRESS>>;tag=as48c8fdc2
To: <sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS>>;tag=as7b2845cb
Call-ID: 1df4d8c3167affd83fec830c1aa2d6c7@<EXT SRV IP ADDRESS>:5060
CSeq: 103 CANCEL
Server: FPBX-2.11.0(11.17.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
<--- SIP read from UDP:10.59.0.101:53139 --->
PUBLISH sip:2000@<DNS имя сервера> SIP/2.0
Via: SIP/2.0/UDP 10.59.0.101:53139;branch=z9hG4bKeede5c0a-027072f1;rport
From: sip:2000@<DNS имя сервера>;tag=79b78edc-f5277392
To: sip:2000@<DNS имя сервера>
Call-ID: 6eec6b06-3a8c1127-1d676e7a-7a10187c
CSeq: 627 PUBLISH
Content-Type: application/pidf+xml
Content-Length: 412
Event: presence
Expires: 600
User-Agent: Sipnetic/1.0.36 Android
Max-Forwards: 70
<?xml version="1.0" encoding="UTF-8"?><presence xmlns="urn:ietf:params:xml:ns:pidf" xmlns:sc="urn:ietf:params:xml:ns:pidf:caps" entity="sip:2000@<DNS имя сервера>"><tuple id="TC0wctkvRGeFGPK85pg8LF0EXvoK9GnUL"><status><basic>open</basic></status><sc:servcaps><sc:audio>true</sc:audio><sc:video>false</sc:video><sc:message>true</sc:message></sc:servcaps><timestamp>2020-12-05T11:14:05Z</timestamp></tuple></presence>
<------------->
--- (12 headers 1 lines) ---
Sending to 10.59.0.101:53139 (NAT)
<--- Transmitting (NAT) to 10.59.0.101:53139 --->
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP 10.59.0.101:53139;branch=z9hG4bKeede5c0a-027072f1;received=10.59.0.101;rport=53139
From: sip:2000@<DNS имя сервера>;tag=79b78edc-f5277392
To: sip:2000@<DNS имя сервера>;tag=as00f1df90
Call-ID: 6eec6b06-3a8c1127-1d676e7a-7a10187c
CSeq: 627 PUBLISH
Server: FPBX-15.0.16.81(16.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<------------>
Really destroying SIP dialog '6eec6b06-3a8c1127-1d676e7a-7a10187c' Method: PUBLISH
<--- SIP read from UDP:10.59.0.101:53139 --->
ACK sip:81038ХХХХХХХХХХ@<DNS имя сервера> SIP/2.0
Via: SIP/2.0/UDP 10.59.0.101:53139;branch=z9hG4bKaff6df32-169f906e;rport
From: sip:2000@<DNS имя сервера>;tag=6f2ca2d5-114bc0da
To: sip:81038ХХХХХХХХХХ@<DNS имя сервера>;tag=as054b4b7b
Call-ID: 9c76bfaa-90a69640-11f9d18f-9246b338
CSeq: 673 ACK
Max-Forwards: 70
<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '9c76bfaa-90a69640-11f9d18f-9246b338' Method: ACK
Really destroying SIP dialog '1df4d8c3167affd83fec830c1aa2d6c7@<EXT SRV IP ADDRESS>:5060' Method: INVITE
Really destroying SIP dialog '0325226f024027d62e8f307b2183f5b6@<PROVIDER SRV IP ADDRESS>:5060' Method: OPTIONS
Really destroying SIP dialog '7f559f597a7aef15151aa223769a945f@<PROVIDER SRV IP ADDRESS>:5060' Method: NOTIFY
Really destroying SIP dialog '1ade083132fd86e532da4ef5276c9989@<PROVIDER SRV IP ADDRESS>:5060' Method: OPTIONS
Really destroying SIP dialog '7ea297040caf98460767b9763fa34a4e@<PROVIDER SRV IP ADDRESS>:5060' Method: NOTIFY
Really destroying SIP dialog '65e5d19a496c1a911f79a9905d25b873@<PROVIDER SRV IP ADDRESS>:5060' Method: OPTIONS
Really destroying SIP dialog '7421488d41acfcd91035ec497f90f413@<PROVIDER SRV IP ADDRESS>:5060' Method: NOTIFY
Really destroying SIP dialog 'c6e1a17f-30ef3f15-ad7a62bf-8700c696' Method: REGISTER
INVITE sip:81038ХХХХХХХХХХ@<DNS имя сервера> SIP/2.0
Via: SIP/2.0/UDP 10.59.0.101:53139;branch=z9hG4bK14a9918c-9c1dde1c;rport
From: sip:2000@<DNS имя сервера>;tag=6f2ca2d5-114bc0da
To: sip:81038ХХХХХХХХХХ@<DNS имя сервера>
Call-ID: 9c76bfaa-90a69640-11f9d18f-9246b338
CSeq: 672 INVITE
Contact: <sip:2000@10.59.0.101:53139;x-reg=33DD1FC6297BA6FB>;audio
Content-Type: application/sdp
Content-Length: 363
User-Agent: Sipnetic/1.0.36 Android
Supported: 100rel,timer,replaces,tdialog
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,UPDATE,SUBSCRIBE,NOTIFY,REFER,PRACK,MESSAGE
Session-Expires: 300
Max-Forwards: 70
v=0
o=- 699186519 699186519 IN IP4 10.59.0.101
s=-
c=IN IP4 10.59.0.101
t=0 0
m=audio 26538 RTP/AVP 96 9 97 3 8 0 101
a=rtpmap:96 opus/48000/2
a=rtpmap:9 G722/8000
a=rtpmap:97 Speex/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
a=fmtp:96 useinbandfec=0
a=fmtp:101 0-15
a=rtcp-mux
<------------->
--- (14 headers 17 lines) ---
Sending to 10.59.0.101:53139 (NAT)
Sending to 10.59.0.101:53139 (NAT)
Using INVITE request as basis request - 9c76bfaa-90a69640-11f9d18f-9246b338
Found peer '2000' for '2000' from 10.59.0.101:53139
<--- Reliably Transmitting (NAT) to 10.59.0.101:53139 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.59.0.101:53139;branch=z9hG4bK14a9918c-9c1dde1c;received=10.59.0.101;rport=53139
From: sip:2000@<DNS имя сервера>;tag=6f2ca2d5-114bc0da
To: sip:81038ХХХХХХХХХХ@<DNS имя сервера>;tag=as20f0580c
Call-ID: 9c76bfaa-90a69640-11f9d18f-9246b338
CSeq: 672 INVITE
Server: FPBX-15.0.16.81(16.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="56535559"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '9c76bfaa-90a69640-11f9d18f-9246b338' in 6400 ms (Method: INVITE)
<--- SIP read from UDP:10.59.0.101:53139 --->
ACK sip:81038ХХХХХХХХХХ@<DNS имя сервера> SIP/2.0
Via: SIP/2.0/UDP 10.59.0.101:53139;branch=z9hG4bK14a9918c-9c1dde1c;rport
From: sip:2000@<DNS имя сервера>;tag=6f2ca2d5-114bc0da
To: sip:81038ХХХХХХХХХХ@<DNS имя сервера>;tag=as20f0580c
Call-ID: 9c76bfaa-90a69640-11f9d18f-9246b338
CSeq: 672 ACK
Max-Forwards: 70
<------------->
--- (7 headers 0 lines) ---
<--- SIP read from UDP:10.59.0.101:53139 --->
INVITE sip:81038ХХХХХХХХХХ@<DNS имя сервера> SIP/2.0
Via: SIP/2.0/UDP 10.59.0.101:53139;branch=z9hG4bKaff6df32-169f906e;rport
From: sip:2000@<DNS имя сервера>;tag=6f2ca2d5-114bc0da
To: sip:81038ХХХХХХХХХХ@<DNS имя сервера>
Call-ID: 9c76bfaa-90a69640-11f9d18f-9246b338
CSeq: 673 INVITE
Contact: <sip:2000@10.59.0.101:53139;x-reg=33DD1FC6297BA6FB>;audio
Content-Type: application/sdp
Content-Length: 363
User-Agent: Sipnetic/1.0.36 Android
Supported: 100rel,timer,replaces,tdialog
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,UPDATE,SUBSCRIBE,NOTIFY,REFER,PRACK,MESSAGE
Session-Expires: 300
Authorization: Digest username="2000",realm="asterisk",nonce="56535559",opaque="",uri="sip:81038ХХХХХХХХХХ@<DNS имя сервера>",algorithm=MD5,response="6ff2967b91ab53ba06266d034da06d24"
Max-Forwards: 70
v=0
o=- 699186519 699186519 IN IP4 10.59.0.101
s=-
c=IN IP4 10.59.0.101
t=0 0
m=audio 26538 RTP/AVP 96 9 97 3 8 0 101
a=rtpmap:96 opus/48000/2
a=rtpmap:9 G722/8000
a=rtpmap:97 Speex/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
a=fmtp:96 useinbandfec=0
a=fmtp:101 0-15
a=rtcp-mux
<------------->
--- (15 headers 17 lines) ---
Sending to 10.59.0.101:53139 (NAT)
Using INVITE request as basis request - 9c76bfaa-90a69640-11f9d18f-9246b338
Found peer '2000' for '2000' from 10.59.0.101:53139
Got SDP version 699186519 and unique parts [- 699186519 IN IP4 10.59.0.101]
Found RTP audio format 96
Found RTP audio format 9
Found RTP audio format 97
Found RTP audio format 3
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 101
Found audio description format opus for ID 96
Found audio description format G722 for ID 9
Found audio description format Speex for ID 97
Found audio description format GSM for ID 3
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw|gsm|g726|g722), peer - audio=(ulaw|gsm|alaw|g722|opus|speex)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|gsm|g722)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.59.0.101:26538
Looking for 81038ХХХХХХХХХХ in from-internal (domain <DNS имя сервера>)
sip_route_dump: route/path hop: <sip:2000@10.59.0.101:53139;x-reg=33DD1FC6297BA6FB>
<--- Transmitting (NAT) to 10.59.0.101:53139 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.59.0.101:53139;branch=z9hG4bKaff6df32-169f906e;received=10.59.0.101;rport=53139
From: sip:2000@<DNS имя сервера>;tag=6f2ca2d5-114bc0da
To: sip:81038ХХХХХХХХХХ@<DNS имя сервера>
Call-ID: 9c76bfaa-90a69640-11f9d18f-9246b338
CSeq: 673 INVITE
Server: FPBX-15.0.16.81(16.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 300;refresher=uas
Contact: <sip:81038ХХХХХХХХХХ@10.60.0.4:5060>
Content-Length: 0
<------------>
<--- SIP read from UDP:10.59.0.101:53139 --->
PUBLISH sip:2000@<DNS имя сервера> SIP/2.0
Via: SIP/2.0/UDP 10.59.0.101:53139;branch=z9hG4bK0675224b-2b3efba3;rport
From: sip:2000@<DNS имя сервера>;tag=aecf8486-c4ff2c95
To: sip:2000@<DNS имя сервера>
Call-ID: c2033fa0-af28ab87-1c4cbbb4-0dd9a5b5
CSeq: 57 PUBLISH
Content-Type: application/pidf+xml
Content-Length: 680
Event: presence
Expires: 600
User-Agent: Sipnetic/1.0.36 Android
Max-Forwards: 70
<?xml version="1.0" encoding="UTF-8"?><presence xmlns="urn:ietf:params:xml:ns:pidf" xmlns:dm="urn:ietf:params:xml:ns:pidf:data-model" xmlns:rpid="urn:ietf:params:xml:ns:pidf:rpid" xmlns:sc="urn:ietf:params:xml:ns:pidf:caps" entity="sip:2000@<DNS имя сервера>"><tuple id="TC0wctkvRGeFGPK85pg8LF0EXvoK9GnUL"><status><basic>open</basic><rpid:activities><rpid:on-the-phone/></rpid:activities></status><sc:servcaps><sc:audio>true</sc:audio><sc:video>false</sc:video><sc:message>true</sc:message></sc:servcaps><timestamp>2020-12-05T11:14:02Z</timestamp></tuple><dm:person id="PKTUTBC4G0SQr963jHfLtQIWwjw1qV7D3"><rpid:activities><rpid:on-the-phone/></rpid:activities></dm:person></presence>
<------------->
--- (12 headers 1 lines) ---
Sending to 10.59.0.101:53139 (NAT)
<--- Transmitting (NAT) to 10.59.0.101:53139 --->
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP 10.59.0.101:53139;branch=z9hG4bK0675224b-2b3efba3;received=10.59.0.101;rport=53139
From: sip:2000@<DNS имя сервера>;tag=aecf8486-c4ff2c95
To: sip:2000@<DNS имя сервера>;tag=as21eb3601
Call-ID: c2033fa0-af28ab87-1c4cbbb4-0dd9a5b5
CSeq: 57 PUBLISH
Server: FPBX-15.0.16.81(16.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<------------>
Really destroying SIP dialog 'c2033fa0-af28ab87-1c4cbbb4-0dd9a5b5' Method: PUBLISH
Audio is at 10088
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to <PROVIDER SRV IP ADDRESS>:5060:
INVITE sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS> SIP/2.0
Via: SIP/2.0/UDP <EXT SRV IP ADDRESS>:5060;branch=z9hG4bK1b3d4ee0;rport
Max-Forwards: 70
From: <sip:495ХХХХХХХ@<EXT SRV IP ADDRESS>>;tag=as48c8fdc2
To: <sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS>>
Contact: <sip:495ХХХХХХХ@<EXT SRV IP ADDRESS>:5060>
Call-ID: 1df4d8c3167affd83fec830c1aa2d6c7@<EXT SRV IP ADDRESS>:5060
CSeq: 102 INVITE
User-Agent: FPBX-15.0.16.81(16.13.0)
Date: Sat, 05 Dec 2020 11:13:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 276
v=0
o=root 274566906 274566906 IN IP4 <EXT SRV IP ADDRESS>
s=Asterisk PBX 16.13.0
c=IN IP4 <EXT SRV IP ADDRESS>
t=0 0
m=audio 10088 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=ptime:20
a=maxptime:150
a=sendrecv
---
<--- SIP read from UDP:<PROVIDER SRV IP ADDRESS>:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP <EXT SRV IP ADDRESS>:5060;branch=z9hG4bK1b3d4ee0;received=<EXT SRV IP ADDRESS>;rport=5060
From: <sip:495ХХХХХХХ@<EXT SRV IP ADDRESS>>;tag=as48c8fdc2
To: <sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS>>;tag=as1173eb19
Call-ID: 1df4d8c3167affd83fec830c1aa2d6c7@<EXT SRV IP ADDRESS>:5060
CSeq: 102 INVITE
Server: FPBX-2.11.0(11.17.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="74885e74"
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
Transmitting (NAT) to <PROVIDER SRV IP ADDRESS>:5060:
ACK sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS> SIP/2.0
Via: SIP/2.0/UDP <EXT SRV IP ADDRESS>:5060;branch=z9hG4bK1b3d4ee0;rport
Max-Forwards: 70
From: <sip:495ХХХХХХХ@<EXT SRV IP ADDRESS>>;tag=as48c8fdc2
To: <sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS>>;tag=as1173eb19
Contact: <sip:495ХХХХХХХ@<EXT SRV IP ADDRESS>:5060>
Call-ID: 1df4d8c3167affd83fec830c1aa2d6c7@<EXT SRV IP ADDRESS>:5060
CSeq: 102 ACK
User-Agent: FPBX-15.0.16.81(16.13.0)
Content-Length: 0
---
Audio is at 10088
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to <PROVIDER SRV IP ADDRESS>:5060:
INVITE sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS> SIP/2.0
Via: SIP/2.0/UDP <EXT SRV IP ADDRESS>:5060;branch=z9hG4bK14b315af;rport
Max-Forwards: 70
From: <sip:495ХХХХХХХ@<EXT SRV IP ADDRESS>>;tag=as48c8fdc2
To: <sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS>>
Contact: <sip:495ХХХХХХХ@<EXT SRV IP ADDRESS>:5060>
Call-ID: 1df4d8c3167affd83fec830c1aa2d6c7@<EXT SRV IP ADDRESS>:5060
CSeq: 103 INVITE
User-Agent: FPBX-15.0.16.81(16.13.0)
Authorization: Digest username="495ХХХХХХХ", realm="asterisk", algorithm=MD5, uri="sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS>", nonce="74885e74", response="bc5903222b7b8c2a8a0753f603d0259d"
Date: Sat, 05 Dec 2020 11:13:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 276
v=0
o=root 274566906 274566907 IN IP4 <EXT SRV IP ADDRESS>
s=Asterisk PBX 16.13.0
c=IN IP4 <EXT SRV IP ADDRESS>
t=0 0
m=audio 10088 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=ptime:20
a=maxptime:150
a=sendrecv
---
<--- SIP read from UDP:<PROVIDER SRV IP ADDRESS>:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP <EXT SRV IP ADDRESS>:5060;branch=z9hG4bK14b315af;received=<EXT SRV IP ADDRESS>;rport=5060
From: <sip:495ХХХХХХХ@<EXT SRV IP ADDRESS>>;tag=as48c8fdc2
To: <sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS>>
Call-ID: 1df4d8c3167affd83fec830c1aa2d6c7@<EXT SRV IP ADDRESS>:5060
CSeq: 103 INVITE
Server: FPBX-2.11.0(11.17.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS>:5060>
Content-Length: 0
<------------->
--- (12 headers 0 lines) ---
<--- SIP read from UDP:<PROVIDER SRV IP ADDRESS>:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP <EXT SRV IP ADDRESS>:5060;branch=z9hG4bK14b315af;received=<EXT SRV IP ADDRESS>;rport=5060
From: <sip:495ХХХХХХХ@<EXT SRV IP ADDRESS>>;tag=as48c8fdc2
To: <sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS>>;tag=as7b2845cb
Call-ID: 1df4d8c3167affd83fec830c1aa2d6c7@<EXT SRV IP ADDRESS>:5060
CSeq: 103 INVITE
Server: FPBX-2.11.0(11.17.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS>:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 260
v=0
o=root 962419659 962419659 IN IP4 <PROVIDER SRV IP ADDRESS>
s=Asterisk PBX 11.17.1
c=IN IP4 <PROVIDER SRV IP ADDRESS>
t=0 0
m=audio 16748 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 12 lines) ---
sip_route_dump: route/path hop: <sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS>:5060>
Got SDP version 962419659 and unique parts [root 962419659 IN IP4 <PROVIDER SRV IP ADDRESS>]
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - (alaw|ulaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (alaw|ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port <PROVIDER SRV IP ADDRESS>:16748
Audio is at 17346
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding codec g722 to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Transmitting (NAT) to 10.59.0.101:53139 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.59.0.101:53139;branch=z9hG4bKaff6df32-169f906e;received=10.59.0.101;rport=53139
From: sip:2000@<DNS имя сервера>;tag=6f2ca2d5-114bc0da
To: sip:81038ХХХХХХХХХХ@<DNS имя сервера>;tag=as054b4b7b
Call-ID: 9c76bfaa-90a69640-11f9d18f-9246b338
CSeq: 673 INVITE
Server: FPBX-15.0.16.81(16.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 300;refresher=uas
Contact: <sip:81038ХХХХХХХХХХ@10.60.0.4:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 317
v=0
o=root 1617966580 1617966580 IN IP4 10.60.0.4
s=Asterisk PBX 16.13.0
c=IN IP4 10.60.0.4
t=0 0
m=audio 17346 RTP/AVP 0 8 3 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
<------------>
<--- SIP read from UDP:10.59.0.101:53139 --->
CANCEL sip:81038ХХХХХХХХХХ@<DNS имя сервера> SIP/2.0
Via: SIP/2.0/UDP 10.59.0.101:53139;branch=z9hG4bKaff6df32-169f906e
From: sip:2000@<DNS имя сервера>;tag=6f2ca2d5-114bc0da
To: sip:81038ХХХХХХХХХХ@<DNS имя сервера>
Call-ID: 9c76bfaa-90a69640-11f9d18f-9246b338
CSeq: 673 CANCEL
Content-Length: 0
Authorization: Digest username="2000",realm="asterisk",nonce="56535559",opaque="",uri="sip:81038ХХХХХХХХХХ@<DNS имя сервера>",algorithm=MD5,response="6ff2967b91ab53ba06266d034da06d24"
Max-Forwards: 70
<------------->
--- (9 headers 0 lines) ---
Sending to 10.59.0.101:53139 (NAT)
<--- Reliably Transmitting (NAT) to 10.59.0.101:53139 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 10.59.0.101:53139;branch=z9hG4bKaff6df32-169f906e;received=10.59.0.101;rport=53139
From: sip:2000@<DNS имя сервера>;tag=6f2ca2d5-114bc0da
To: sip:81038ХХХХХХХХХХ@<DNS имя сервера>;tag=as054b4b7b
Call-ID: 9c76bfaa-90a69640-11f9d18f-9246b338
CSeq: 673 INVITE
Server: FPBX-15.0.16.81(16.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<------------>
<--- Transmitting (NAT) to 10.59.0.101:53139 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.59.0.101:53139;branch=z9hG4bKaff6df32-169f906e;received=10.59.0.101;rport=53139
From: sip:2000@<DNS имя сервера>;tag=6f2ca2d5-114bc0da
To: sip:81038ХХХХХХХХХХ@<DNS имя сервера>;tag=as054b4b7b
Call-ID: 9c76bfaa-90a69640-11f9d18f-9246b338
CSeq: 673 CANCEL
Server: FPBX-15.0.16.81(16.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '1df4d8c3167affd83fec830c1aa2d6c7@<EXT SRV IP ADDRESS>:5060' in 6400 ms (Method: INVITE)
Reliably Transmitting (NAT) to <PROVIDER SRV IP ADDRESS>:5060:
CANCEL sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS> SIP/2.0
Via: SIP/2.0/UDP <EXT SRV IP ADDRESS>:5060;branch=z9hG4bK14b315af;rport
Max-Forwards: 70
From: <sip:495ХХХХХХХ@<EXT SRV IP ADDRESS>>;tag=as48c8fdc2
To: <sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS>>
Call-ID: 1df4d8c3167affd83fec830c1aa2d6c7@<EXT SRV IP ADDRESS>:5060
CSeq: 103 CANCEL
User-Agent: FPBX-15.0.16.81(16.13.0)
Content-Length: 0
---
Scheduling destruction of SIP dialog '1df4d8c3167affd83fec830c1aa2d6c7@<EXT SRV IP ADDRESS>:5060' in 6400 ms (Method: INVITE)
<--- SIP read from UDP:<PROVIDER SRV IP ADDRESS>:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP <EXT SRV IP ADDRESS>:5060;branch=z9hG4bK14b315af;received=<EXT SRV IP ADDRESS>;rport=5060
From: <sip:495ХХХХХХХ@<EXT SRV IP ADDRESS>>;tag=as48c8fdc2
To: <sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS>>;tag=as7b2845cb
Call-ID: 1df4d8c3167affd83fec830c1aa2d6c7@<EXT SRV IP ADDRESS>:5060
CSeq: 103 INVITE
Server: FPBX-2.11.0(11.17.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Transmitting (NAT) to <PROVIDER SRV IP ADDRESS>:5060:
ACK sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS>:5060 SIP/2.0
Via: SIP/2.0/UDP <EXT SRV IP ADDRESS>:5060;branch=z9hG4bK14b315af;rport
Max-Forwards: 70
From: <sip:495ХХХХХХХ@<EXT SRV IP ADDRESS>>;tag=as48c8fdc2
To: <sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS>>;tag=as7b2845cb
Contact: <sip:495ХХХХХХХ@<EXT SRV IP ADDRESS>:5060>
Call-ID: 1df4d8c3167affd83fec830c1aa2d6c7@<EXT SRV IP ADDRESS>:5060
CSeq: 103 ACK
User-Agent: FPBX-15.0.16.81(16.13.0)
Content-Length: 0
---
Scheduling destruction of SIP dialog '1df4d8c3167affd83fec830c1aa2d6c7@<EXT SRV IP ADDRESS>:5060' in 6400 ms (Method: INVITE)
<--- SIP read from UDP:<PROVIDER SRV IP ADDRESS>:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP <EXT SRV IP ADDRESS>:5060;branch=z9hG4bK14b315af;received=<EXT SRV IP ADDRESS>;rport=5060
From: <sip:495ХХХХХХХ@<EXT SRV IP ADDRESS>>;tag=as48c8fdc2
To: <sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS>>;tag=as7b2845cb
Call-ID: 1df4d8c3167affd83fec830c1aa2d6c7@<EXT SRV IP ADDRESS>:5060
CSeq: 103 CANCEL
Server: FPBX-2.11.0(11.17.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
<--- SIP read from UDP:10.59.0.101:53139 --->
PUBLISH sip:2000@<DNS имя сервера> SIP/2.0
Via: SIP/2.0/UDP 10.59.0.101:53139;branch=z9hG4bKeede5c0a-027072f1;rport
From: sip:2000@<DNS имя сервера>;tag=79b78edc-f5277392
To: sip:2000@<DNS имя сервера>
Call-ID: 6eec6b06-3a8c1127-1d676e7a-7a10187c
CSeq: 627 PUBLISH
Content-Type: application/pidf+xml
Content-Length: 412
Event: presence
Expires: 600
User-Agent: Sipnetic/1.0.36 Android
Max-Forwards: 70
<?xml version="1.0" encoding="UTF-8"?><presence xmlns="urn:ietf:params:xml:ns:pidf" xmlns:sc="urn:ietf:params:xml:ns:pidf:caps" entity="sip:2000@<DNS имя сервера>"><tuple id="TC0wctkvRGeFGPK85pg8LF0EXvoK9GnUL"><status><basic>open</basic></status><sc:servcaps><sc:audio>true</sc:audio><sc:video>false</sc:video><sc:message>true</sc:message></sc:servcaps><timestamp>2020-12-05T11:14:05Z</timestamp></tuple></presence>
<------------->
--- (12 headers 1 lines) ---
Sending to 10.59.0.101:53139 (NAT)
<--- Transmitting (NAT) to 10.59.0.101:53139 --->
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP 10.59.0.101:53139;branch=z9hG4bKeede5c0a-027072f1;received=10.59.0.101;rport=53139
From: sip:2000@<DNS имя сервера>;tag=79b78edc-f5277392
To: sip:2000@<DNS имя сервера>;tag=as00f1df90
Call-ID: 6eec6b06-3a8c1127-1d676e7a-7a10187c
CSeq: 627 PUBLISH
Server: FPBX-15.0.16.81(16.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<------------>
Really destroying SIP dialog '6eec6b06-3a8c1127-1d676e7a-7a10187c' Method: PUBLISH
<--- SIP read from UDP:10.59.0.101:53139 --->
ACK sip:81038ХХХХХХХХХХ@<DNS имя сервера> SIP/2.0
Via: SIP/2.0/UDP 10.59.0.101:53139;branch=z9hG4bKaff6df32-169f906e;rport
From: sip:2000@<DNS имя сервера>;tag=6f2ca2d5-114bc0da
To: sip:81038ХХХХХХХХХХ@<DNS имя сервера>;tag=as054b4b7b
Call-ID: 9c76bfaa-90a69640-11f9d18f-9246b338
CSeq: 673 ACK
Max-Forwards: 70
<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '9c76bfaa-90a69640-11f9d18f-9246b338' Method: ACK
Really destroying SIP dialog '1df4d8c3167affd83fec830c1aa2d6c7@<EXT SRV IP ADDRESS>:5060' Method: INVITE
Really destroying SIP dialog '0325226f024027d62e8f307b2183f5b6@<PROVIDER SRV IP ADDRESS>:5060' Method: OPTIONS
Really destroying SIP dialog '7f559f597a7aef15151aa223769a945f@<PROVIDER SRV IP ADDRESS>:5060' Method: NOTIFY
Really destroying SIP dialog '1ade083132fd86e532da4ef5276c9989@<PROVIDER SRV IP ADDRESS>:5060' Method: OPTIONS
Really destroying SIP dialog '7ea297040caf98460767b9763fa34a4e@<PROVIDER SRV IP ADDRESS>:5060' Method: NOTIFY
Really destroying SIP dialog '65e5d19a496c1a911f79a9905d25b873@<PROVIDER SRV IP ADDRESS>:5060' Method: OPTIONS
Really destroying SIP dialog '7421488d41acfcd91035ec497f90f413@<PROVIDER SRV IP ADDRESS>:5060' Method: NOTIFY
Really destroying SIP dialog 'c6e1a17f-30ef3f15-ad7a62bf-8700c696' Method: REGISTER
<--- SIP read from UDP:10.60.0.2:35060 --->
INVITE sip:81038ХХХХХХХХХХ@10.60.0.4 SIP/2.0
Via: SIP/2.0/UDP 10.60.0.2:35060;branch=z9hG4bK00005de0;rport
Max-Forwards: 70
To: sip:81038ХХХХХХХХХХ@10.60.0.4
From: "Sokolov S.V." <sip:189@10.60.0.4>;tag=9863
Call-ID: 000041e3-38e3985e3387100098f70080f0c171e8@10.60.0.2
CSeq: 1 INVITE
Contact: sip:sttc-ncp1000@10.60.0.2:35060
Supported: timer,100rel
Session-Expires: 180
Allow: INVITE,ACK,CANCEL,BYE,PRACK,OPTIONS,REGISTER,INFO,NOTIFY,UPDATE
Content-Type: application/sdp
User-Agent: Panasonic-MPR11-V8.0102/VSIPGW-V2.3002
Content-Length: 260
v=0
o=- 1 1 IN IP4 10.60.0.3
s=-
c=IN IP4 10.60.0.3
t=0 0
m=audio 12170 RTP/AVP 8 0 18 101
a=rtpmap:8 PCMA/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:18 G729/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=ptime:20
a=rtcp:12171
<------------->
--- (14 headers 14 lines) ---
Sending to 10.60.0.2:35060 (NAT)
Sending to 10.60.0.2:35060 (NAT)
Using INVITE request as basis request - 000041e3-38e3985e3387100098f70080f0c171e8@10.60.0.2
Found peer 'sttc-ncp1000' for '189' from 10.60.0.2:35060
<--- Reliably Transmitting (no NAT) to 10.60.0.2:35060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.60.0.2:35060;branch=z9hG4bK00005de0;received=10.60.0.2;rport=35060
From: "Sokolov S.V." <sip:189@10.60.0.4>;tag=9863
To: sip:81038ХХХХХХХХХХ@10.60.0.4;tag=as418283a6
Call-ID: 000041e3-38e3985e3387100098f70080f0c171e8@10.60.0.2
CSeq: 1 INVITE
Server: FPBX-15.0.16.81(16.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="76aa5b39"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '000041e3-38e3985e3387100098f70080f0c171e8@10.60.0.2' in 6400 ms (Method: INVITE)
<--- SIP read from UDP:10.60.0.2:35060 --->
ACK sip:81038ХХХХХХХХХХ@10.60.0.4 SIP/2.0
Via: SIP/2.0/UDP 10.60.0.2:35060;branch=z9hG4bK00005de0;rport
Max-Forwards: 70
To: sip:81038ХХХХХХХХХХ@10.60.0.4;tag=as418283a6
From: "Sokolov S.V." <sip:189@10.60.0.4>;tag=9863
Call-ID: 000041e3-38e3985e3387100098f70080f0c171e8@10.60.0.2
CSeq: 1 ACK
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from UDP:10.60.0.2:35060 --->
INVITE sip:81038ХХХХХХХХХХ@10.60.0.4 SIP/2.0
Via: SIP/2.0/UDP 10.60.0.2:35060;branch=z9hG4bK0000697d;rport
Max-Forwards: 70
To: sip:81038ХХХХХХХХХХ@10.60.0.4
From: "Sokolov S.V." <sip:189@10.60.0.4>;tag=9863
Call-ID: 000041e3-38e3985e3387100098f70080f0c171e8@10.60.0.2
CSeq: 2 INVITE
Contact: sip:sttc-ncp1000@10.60.0.2:35060
Supported: timer,100rel
Authorization: Digest realm="asterisk", nonce="76aa5b39", algorithm=MD5, uri="sip:81038ХХХХХХХХХХ@10.60.0.4", username="sttc-ncp1000", response="fff1bbbc8b5e7edc02b56fbef2885f3f"
Session-Expires: 180
Allow: INVITE,ACK,CANCEL,BYE,PRACK,OPTIONS,REGISTER,INFO,NOTIFY,UPDATE
Content-Type: application/sdp
User-Agent: Panasonic-MPR11-V8.0102/VSIPGW-V2.3002
Content-Length: 260
v=0
o=- 1 1 IN IP4 10.60.0.3
s=-
c=IN IP4 10.60.0.3
t=0 0
m=audio 12170 RTP/AVP 8 0 18 101
a=rtpmap:8 PCMA/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:18 G729/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=ptime:20
a=rtcp:12171
<------------->
--- (15 headers 14 lines) ---
Sending to 10.60.0.2:35060 (no NAT)
Using INVITE request as basis request - 000041e3-38e3985e3387100098f70080f0c171e8@10.60.0.2
Found peer 'sttc-ncp1000' for '189' from 10.60.0.2:35060
Got SDP version 1 and unique parts [- 1 IN IP4 10.60.0.3]
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - (alaw|ulaw|g729), peer - audio=(ulaw|alaw|g729)/video=(nothing)/text=(nothing), combined - (alaw|ulaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.60.0.3:12170
Looking for 81038ХХХХХХХХХХ in from-internal (domain 10.60.0.4)
sip_route_dump: route/path hop: <sip:sttc-ncp1000@10.60.0.2:35060>
<--- Transmitting (no NAT) to 10.60.0.2:35060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.60.0.2:35060;branch=z9hG4bK0000697d;received=10.60.0.2;rport=35060
From: "Sokolov S.V." <sip:189@10.60.0.4>;tag=9863
To: sip:81038ХХХХХХХХХХ@10.60.0.4
Call-ID: 000041e3-38e3985e3387100098f70080f0c171e8@10.60.0.2
CSeq: 2 INVITE
Server: FPBX-15.0.16.81(16.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 180;refresher=uas
Contact: <sip:81038ХХХХХХХХХХ@10.60.0.4:5060>
Content-Length: 0
<------------>
Audio is at 18354
Adding codec alaw to SDP
Adding codec ulaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to <PROVIDER SRV IP ADDRESS>:5060:
INVITE sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS> SIP/2.0
Via: SIP/2.0/UDP <EXT SRV IP ADDRESS>:5060;branch=z9hG4bK6153286a;rport
Max-Forwards: 70
From: <sip:495ХХХХХХХ@<EXT SRV IP ADDRESS>>;tag=as0b354112
To: <sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS>>
Contact: <sip:495ХХХХХХХ@<EXT SRV IP ADDRESS>:5060>
Call-ID: 3cacfe5530b1605f066eb0776b491bf0@<EXT SRV IP ADDRESS>:5060
CSeq: 102 INVITE
User-Agent: FPBX-15.0.16.81(16.13.0)
Date: Sat, 05 Dec 2020 11:06:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 276
v=0
o=root 144511809 144511809 IN IP4 <EXT SRV IP ADDRESS>
s=Asterisk PBX 16.13.0
c=IN IP4 <EXT SRV IP ADDRESS>
t=0 0
m=audio 18354 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
---
<--- SIP read from UDP:<PROVIDER SRV IP ADDRESS>:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP <EXT SRV IP ADDRESS>:5060;branch=z9hG4bK6153286a;received=<EXT SRV IP ADDRESS>;rport=5060
From: <sip:495ХХХХХХХ@<EXT SRV IP ADDRESS>>;tag=as0b354112
To: <sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS>>;tag=as299baccc
Call-ID: 3cacfe5530b1605f066eb0776b491bf0@<EXT SRV IP ADDRESS>:5060
CSeq: 102 INVITE
Server: FPBX-2.11.0(11.17.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="65982f24"
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
Transmitting (NAT) to <PROVIDER SRV IP ADDRESS>:5060:
ACK sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS> SIP/2.0
Via: SIP/2.0/UDP <EXT SRV IP ADDRESS>:5060;branch=z9hG4bK6153286a;rport
Max-Forwards: 70
From: <sip:495ХХХХХХХ@<EXT SRV IP ADDRESS>>;tag=as0b354112
To: <sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS>>;tag=as299baccc
Contact: <sip:495ХХХХХХХ@<EXT SRV IP ADDRESS>:5060>
Call-ID: 3cacfe5530b1605f066eb0776b491bf0@<EXT SRV IP ADDRESS>:5060
CSeq: 102 ACK
User-Agent: FPBX-15.0.16.81(16.13.0)
Content-Length: 0
---
Audio is at 18354
Adding codec alaw to SDP
Adding codec ulaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to <PROVIDER SRV IP ADDRESS>:5060:
INVITE sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS> SIP/2.0
Via: SIP/2.0/UDP <EXT SRV IP ADDRESS>:5060;branch=z9hG4bK0277acd4;rport
Max-Forwards: 70
From: <sip:495ХХХХХХХ@<EXT SRV IP ADDRESS>>;tag=as0b354112
To: <sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS>>
Contact: <sip:495ХХХХХХХ@<EXT SRV IP ADDRESS>:5060>
Call-ID: 3cacfe5530b1605f066eb0776b491bf0@<EXT SRV IP ADDRESS>:5060
CSeq: 103 INVITE
User-Agent: FPBX-15.0.16.81(16.13.0)
Authorization: Digest username="495ХХХХХХХ", realm="asterisk", algorithm=MD5, uri="sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS>", nonce="65982f24", response="619cf336f045a45a59e3aa0a6d218dbc"
Date: Sat, 05 Dec 2020 11:06:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 276
v=0
o=root 144511809 144511810 IN IP4 <EXT SRV IP ADDRESS>
s=Asterisk PBX 16.13.0
c=IN IP4 <EXT SRV IP ADDRESS>
t=0 0
m=audio 18354 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
---
<--- SIP read from UDP:<PROVIDER SRV IP ADDRESS>:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP <EXT SRV IP ADDRESS>:5060;branch=z9hG4bK0277acd4;received=<EXT SRV IP ADDRESS>;rport=5060
From: <sip:495ХХХХХХХ@<EXT SRV IP ADDRESS>>;tag=as0b354112
To: <sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS>>
Call-ID: 3cacfe5530b1605f066eb0776b491bf0@<EXT SRV IP ADDRESS>:5060
CSeq: 103 INVITE
Server: FPBX-2.11.0(11.17.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS>:5060>
Content-Length: 0
<------------->
--- (12 headers 0 lines) ---
<--- SIP read from UDP:<PROVIDER SRV IP ADDRESS>:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP <EXT SRV IP ADDRESS>:5060;branch=z9hG4bK0277acd4;received=<EXT SRV IP ADDRESS>;rport=5060
From: <sip:495ХХХХХХХ@<EXT SRV IP ADDRESS>>;tag=as0b354112
To: <sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS>>;tag=as4d5bd64c
Call-ID: 3cacfe5530b1605f066eb0776b491bf0@<EXT SRV IP ADDRESS>:5060
CSeq: 103 INVITE
Server: FPBX-2.11.0(11.17.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS>:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 262
v=0
o=root 1929802605 1929802605 IN IP4 <PROVIDER SRV IP ADDRESS>
s=Asterisk PBX 11.17.1
c=IN IP4 <PROVIDER SRV IP ADDRESS>
t=0 0
m=audio 15712 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 12 lines) ---
sip_route_dump: route/path hop: <sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS>:5060>
Got SDP version 1929802605 and unique parts [root 1929802605 IN IP4 <PROVIDER SRV IP ADDRESS>]
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - (alaw|ulaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (alaw|ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port <PROVIDER SRV IP ADDRESS>:15712
Audio is at 18494
Adding codec alaw to SDP
Adding codec ulaw to SDP
Adding codec g729 to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Transmitting (no NAT) to 10.60.0.2:35060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.60.0.2:35060;branch=z9hG4bK0000697d;received=10.60.0.2;rport=35060
From: "Sokolov S.V." <sip:189@10.60.0.4>;tag=9863
To: sip:81038ХХХХХХХХХХ@10.60.0.4;tag=as5a78e927
Call-ID: 000041e3-38e3985e3387100098f70080f0c171e8@10.60.0.2
CSeq: 2 INVITE
Server: FPBX-15.0.16.81(16.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 180;refresher=uas
Contact: <sip:81038ХХХХХХХХХХ@10.60.0.4:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 315
v=0
o=root 225066610 225066610 IN IP4 10.60.0.4
s=Asterisk PBX 16.13.0
c=IN IP4 10.60.0.4
t=0 0
m=audio 18494 RTP/AVP 8 0 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
<------------>
<--- SIP read from UDP:<PROVIDER SRV IP ADDRESS>:5060 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP <EXT SRV IP ADDRESS>:5060;branch=z9hG4bK0277acd4;received=<EXT SRV IP ADDRESS>;rport=5060
From: <sip:495ХХХХХХХ@<EXT SRV IP ADDRESS>>;tag=as0b354112
To: <sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS>>;tag=as4d5bd64c
Call-ID: 3cacfe5530b1605f066eb0776b491bf0@<EXT SRV IP ADDRESS>:5060
CSeq: 103 INVITE
Server: FPBX-2.11.0(11.17.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
X-Asterisk-HangupCause: Circuit/channel congestion
X-Asterisk-HangupCauseCode: 34
Content-Length: 0
<------------->
--- (13 headers 0 lines) ---
Transmitting (NAT) to <PROVIDER SRV IP ADDRESS>:5060:
ACK sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS>:5060 SIP/2.0
Via: SIP/2.0/UDP <EXT SRV IP ADDRESS>:5060;branch=z9hG4bK0277acd4;rport
Max-Forwards: 70
From: <sip:495ХХХХХХХ@<EXT SRV IP ADDRESS>>;tag=as0b354112
To: <sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS>>;tag=as4d5bd64c
Contact: <sip:495ХХХХХХХ@<EXT SRV IP ADDRESS>:5060>
Call-ID: 3cacfe5530b1605f066eb0776b491bf0@<EXT SRV IP ADDRESS>:5060
CSeq: 103 ACK
User-Agent: FPBX-15.0.16.81(16.13.0)
Content-Length: 0
---
Really destroying SIP dialog '3cacfe5530b1605f066eb0776b491bf0@<EXT SRV IP ADDRESS>:5060' Method: INVITE
Audio is at 12850
Adding codec alaw to SDP
Adding codec ulaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to <PROVIDER SRV IP ADDRESS>:5060:
INVITE sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS> SIP/2.0
Via: SIP/2.0/UDP <EXT SRV IP ADDRESS>:5060;branch=z9hG4bK23ad8ef2;rport
Max-Forwards: 70
From: <sip:495XXXXXXX@<EXT SRV IP ADDRESS>>;tag=as0ac50218
To: <sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS>>
Contact: <sip:495XXXXXXX@<EXT SRV IP ADDRESS>:5060>
Call-ID: 1f4619432cdaad63687c777d4cff2f7f@<EXT SRV IP ADDRESS>:5060
CSeq: 102 INVITE
User-Agent: FPBX-15.0.16.81(16.13.0)
Date: Sat, 05 Dec 2020 11:06:59 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 276
v=0
o=root 351939896 351939896 IN IP4 <EXT SRV IP ADDRESS>
s=Asterisk PBX 16.13.0
c=IN IP4 <EXT SRV IP ADDRESS>
t=0 0
m=audio 12850 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
---
<--- SIP read from UDP:<PROVIDER SRV IP ADDRESS>:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP <EXT SRV IP ADDRESS>:5060;branch=z9hG4bK23ad8ef2;received=<EXT SRV IP ADDRESS>;rport=5060
From: <sip:495XXXXXXX@<EXT SRV IP ADDRESS>>;tag=as0ac50218
To: <sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS>>;tag=as556c0c28
Call-ID: 1f4619432cdaad63687c777d4cff2f7f@<EXT SRV IP ADDRESS>:5060
CSeq: 102 INVITE
Server: FPBX-2.11.0(11.17.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="09e97248"
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
Transmitting (NAT) to <PROVIDER SRV IP ADDRESS>:5060:
ACK sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS> SIP/2.0
Via: SIP/2.0/UDP <EXT SRV IP ADDRESS>:5060;branch=z9hG4bK23ad8ef2;rport
Max-Forwards: 70
From: <sip:495XXXXXXX@<EXT SRV IP ADDRESS>>;tag=as0ac50218
To: <sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS>>;tag=as556c0c28
Contact: <sip:495XXXXXXX@<EXT SRV IP ADDRESS>:5060>
Call-ID: 1f4619432cdaad63687c777d4cff2f7f@<EXT SRV IP ADDRESS>:5060
CSeq: 102 ACK
User-Agent: FPBX-15.0.16.81(16.13.0)
Content-Length: 0
---
Audio is at 12850
Adding codec alaw to SDP
Adding codec ulaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to <PROVIDER SRV IP ADDRESS>:5060:
INVITE sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS> SIP/2.0
Via: SIP/2.0/UDP <EXT SRV IP ADDRESS>:5060;branch=z9hG4bK1245c14b;rport
Max-Forwards: 70
From: <sip:495XXXXXXX@<EXT SRV IP ADDRESS>>;tag=as0ac50218
To: <sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS>>
Contact: <sip:495XXXXXXX@<EXT SRV IP ADDRESS>:5060>
Call-ID: 1f4619432cdaad63687c777d4cff2f7f@<EXT SRV IP ADDRESS>:5060
CSeq: 103 INVITE
User-Agent: FPBX-15.0.16.81(16.13.0)
Authorization: Digest username="495XXXXXXX", realm="asterisk", algorithm=MD5, uri="sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS>", nonce="09e97248", response="417ae4a7196a840462491d4de968539f"
Date: Sat, 05 Dec 2020 11:06:59 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 276
v=0
o=root 351939896 351939897 IN IP4 <EXT SRV IP ADDRESS>
s=Asterisk PBX 16.13.0
c=IN IP4 <EXT SRV IP ADDRESS>
t=0 0
m=audio 12850 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
---
<--- SIP read from UDP:<PROVIDER SRV IP ADDRESS>:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP <EXT SRV IP ADDRESS>:5060;branch=z9hG4bK1245c14b;received=<EXT SRV IP ADDRESS>;rport=5060
From: <sip:495XXXXXXX@<EXT SRV IP ADDRESS>>;tag=as0ac50218
To: <sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS>>
Call-ID: 1f4619432cdaad63687c777d4cff2f7f@<EXT SRV IP ADDRESS>:5060
CSeq: 103 INVITE
Server: FPBX-2.11.0(11.17.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS>:5060>
Content-Length: 0
<------------->
--- (12 headers 0 lines) ---
<--- SIP read from UDP:<PROVIDER SRV IP ADDRESS>:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP <EXT SRV IP ADDRESS>:5060;branch=z9hG4bK1245c14b;received=<EXT SRV IP ADDRESS>;rport=5060
From: <sip:495XXXXXXX@<EXT SRV IP ADDRESS>>;tag=as0ac50218
To: <sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS>>;tag=as1b67c728
Call-ID: 1f4619432cdaad63687c777d4cff2f7f@<EXT SRV IP ADDRESS>:5060
CSeq: 103 INVITE
Server: FPBX-2.11.0(11.17.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS>:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 262
v=0
o=root 2140215254 2140215254 IN IP4 <PROVIDER SRV IP ADDRESS>
s=Asterisk PBX 11.17.1
c=IN IP4 <PROVIDER SRV IP ADDRESS>
t=0 0
m=audio 17992 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 12 lines) ---
sip_route_dump: route/path hop: <sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS>:5060>
Got SDP version 2140215254 and unique parts [root 2140215254 IN IP4 <PROVIDER SRV IP ADDRESS>]
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - (alaw|ulaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (alaw|ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port <PROVIDER SRV IP ADDRESS>:17992
<--- SIP read from UDP:10.60.0.2:35060 --->
CANCEL sip:81038ХХХХХХХХХХ@10.60.0.4 SIP/2.0
Via: SIP/2.0/UDP 10.60.0.2:35060;branch=z9hG4bK0000697d;rport
Max-Forwards: 70
To: sip:81038ХХХХХХХХХХ@10.60.0.4
From: "Sokolov S.V." <sip:189@10.60.0.4>;tag=9863
Call-ID: 000041e3-38e3985e3387100098f70080f0c171e8@10.60.0.2
CSeq: 2 CANCEL
Authorization: Digest realm="asterisk", nonce="76aa5b39", algorithm=MD5, uri="sip:81038ХХХХХХХХХХ@10.60.0.4", username="sttc-ncp1000", response="69dfd10328646aea35eb8f979ebc97cb"
User-Agent: Panasonic-MPR11-V8.0102/VSIPGW-V2.3002
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Sending to 10.60.0.2:35060 (no NAT)
<--- Reliably Transmitting (no NAT) to 10.60.0.2:35060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 10.60.0.2:35060;branch=z9hG4bK0000697d;received=10.60.0.2;rport=35060
From: "Sokolov S.V." <sip:189@10.60.0.4>;tag=9863
To: sip:81038ХХХХХХХХХХ@10.60.0.4;tag=as5a78e927
Call-ID: 000041e3-38e3985e3387100098f70080f0c171e8@10.60.0.2
CSeq: 2 INVITE
Server: FPBX-15.0.16.81(16.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<------------>
<--- Transmitting (no NAT) to 10.60.0.2:35060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.60.0.2:35060;branch=z9hG4bK0000697d;received=10.60.0.2;rport=35060
From: "Sokolov S.V." <sip:189@10.60.0.4>;tag=9863
To: sip:81038ХХХХХХХХХХ@10.60.0.4;tag=as5a78e927
Call-ID: 000041e3-38e3985e3387100098f70080f0c171e8@10.60.0.2
CSeq: 2 CANCEL
Server: FPBX-15.0.16.81(16.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '1f4619432cdaad63687c777d4cff2f7f@<EXT SRV IP ADDRESS>:5060' in 6400 ms (Method: INVITE)
Reliably Transmitting (NAT) to <PROVIDER SRV IP ADDRESS>:5060:
CANCEL sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS> SIP/2.0
Via: SIP/2.0/UDP <EXT SRV IP ADDRESS>:5060;branch=z9hG4bK1245c14b;rport
Max-Forwards: 70
From: <sip:495XXXXXXX@<EXT SRV IP ADDRESS>>;tag=as0ac50218
To: <sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS>>
Call-ID: 1f4619432cdaad63687c777d4cff2f7f@<EXT SRV IP ADDRESS>:5060
CSeq: 103 CANCEL
User-Agent: FPBX-15.0.16.81(16.13.0)
Content-Length: 0
---
Scheduling destruction of SIP dialog '1f4619432cdaad63687c777d4cff2f7f@<EXT SRV IP ADDRESS>:5060' in 6400 ms (Method: INVITE)
<--- SIP read from UDP:<PROVIDER SRV IP ADDRESS>:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP <EXT SRV IP ADDRESS>:5060;branch=z9hG4bK1245c14b;received=<EXT SRV IP ADDRESS>;rport=5060
From: <sip:495XXXXXXX@<EXT SRV IP ADDRESS>>;tag=as0ac50218
To: <sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS>>;tag=as1b67c728
Call-ID: 1f4619432cdaad63687c777d4cff2f7f@<EXT SRV IP ADDRESS>:5060
CSeq: 103 INVITE
Server: FPBX-2.11.0(11.17.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Transmitting (NAT) to <PROVIDER SRV IP ADDRESS>:5060:
ACK sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS>:5060 SIP/2.0
Via: SIP/2.0/UDP <EXT SRV IP ADDRESS>:5060;branch=z9hG4bK1245c14b;rport
Max-Forwards: 70
From: <sip:495XXXXXXX@<EXT SRV IP ADDRESS>>;tag=as0ac50218
To: <sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS>>;tag=as1b67c728
Contact: <sip:495XXXXXXX@<EXT SRV IP ADDRESS>:5060>
Call-ID: 1f4619432cdaad63687c777d4cff2f7f@<EXT SRV IP ADDRESS>:5060
CSeq: 103 ACK
User-Agent: FPBX-15.0.16.81(16.13.0)
Content-Length: 0
---
Scheduling destruction of SIP dialog '1f4619432cdaad63687c777d4cff2f7f@<EXT SRV IP ADDRESS>:5060' in 6400 ms (Method: INVITE)
<--- SIP read from UDP:<PROVIDER SRV IP ADDRESS>:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP <EXT SRV IP ADDRESS>:5060;branch=z9hG4bK1245c14b;received=<EXT SRV IP ADDRESS>;rport=5060
From: <sip:495XXXXXXX@<EXT SRV IP ADDRESS>>;tag=as0ac50218
To: <sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS>>;tag=as1b67c728
Call-ID: 1f4619432cdaad63687c777d4cff2f7f@<EXT SRV IP ADDRESS>:5060
CSeq: 103 CANCEL
Server: FPBX-2.11.0(11.17.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
<--- SIP read from UDP:10.60.0.2:35060 --->
ACK sip:81038ХХХХХХХХХХ@10.60.0.4 SIP/2.0
Via: SIP/2.0/UDP 10.60.0.2:35060;branch=z9hG4bK0000697d;rport
Max-Forwards: 70
To: sip:81038ХХХХХХХХХХ@10.60.0.4;tag=as5a78e927
From: "Sokolov S.V." <sip:189@10.60.0.4>;tag=9863
Call-ID: 000041e3-38e3985e3387100098f70080f0c171e8@10.60.0.2
CSeq: 2 ACK
Authorization: Digest realm="asterisk", nonce="76aa5b39", algorithm=MD5, uri="sip:81038ХХХХХХХХХХ@10.60.0.4", username="sttc-ncp1000", response="5919fa45d25dbc1fc8145e193ece7d60"
Content-Length: 0
INVITE sip:81038ХХХХХХХХХХ@10.60.0.4 SIP/2.0
Via: SIP/2.0/UDP 10.60.0.2:35060;branch=z9hG4bK00005de0;rport
Max-Forwards: 70
To: sip:81038ХХХХХХХХХХ@10.60.0.4
From: "Sokolov S.V." <sip:189@10.60.0.4>;tag=9863
Call-ID: 000041e3-38e3985e3387100098f70080f0c171e8@10.60.0.2
CSeq: 1 INVITE
Contact: sip:sttc-ncp1000@10.60.0.2:35060
Supported: timer,100rel
Session-Expires: 180
Allow: INVITE,ACK,CANCEL,BYE,PRACK,OPTIONS,REGISTER,INFO,NOTIFY,UPDATE
Content-Type: application/sdp
User-Agent: Panasonic-MPR11-V8.0102/VSIPGW-V2.3002
Content-Length: 260
v=0
o=- 1 1 IN IP4 10.60.0.3
s=-
c=IN IP4 10.60.0.3
t=0 0
m=audio 12170 RTP/AVP 8 0 18 101
a=rtpmap:8 PCMA/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:18 G729/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=ptime:20
a=rtcp:12171
<------------->
--- (14 headers 14 lines) ---
Sending to 10.60.0.2:35060 (NAT)
Sending to 10.60.0.2:35060 (NAT)
Using INVITE request as basis request - 000041e3-38e3985e3387100098f70080f0c171e8@10.60.0.2
Found peer 'sttc-ncp1000' for '189' from 10.60.0.2:35060
<--- Reliably Transmitting (no NAT) to 10.60.0.2:35060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.60.0.2:35060;branch=z9hG4bK00005de0;received=10.60.0.2;rport=35060
From: "Sokolov S.V." <sip:189@10.60.0.4>;tag=9863
To: sip:81038ХХХХХХХХХХ@10.60.0.4;tag=as418283a6
Call-ID: 000041e3-38e3985e3387100098f70080f0c171e8@10.60.0.2
CSeq: 1 INVITE
Server: FPBX-15.0.16.81(16.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="76aa5b39"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '000041e3-38e3985e3387100098f70080f0c171e8@10.60.0.2' in 6400 ms (Method: INVITE)
<--- SIP read from UDP:10.60.0.2:35060 --->
ACK sip:81038ХХХХХХХХХХ@10.60.0.4 SIP/2.0
Via: SIP/2.0/UDP 10.60.0.2:35060;branch=z9hG4bK00005de0;rport
Max-Forwards: 70
To: sip:81038ХХХХХХХХХХ@10.60.0.4;tag=as418283a6
From: "Sokolov S.V." <sip:189@10.60.0.4>;tag=9863
Call-ID: 000041e3-38e3985e3387100098f70080f0c171e8@10.60.0.2
CSeq: 1 ACK
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from UDP:10.60.0.2:35060 --->
INVITE sip:81038ХХХХХХХХХХ@10.60.0.4 SIP/2.0
Via: SIP/2.0/UDP 10.60.0.2:35060;branch=z9hG4bK0000697d;rport
Max-Forwards: 70
To: sip:81038ХХХХХХХХХХ@10.60.0.4
From: "Sokolov S.V." <sip:189@10.60.0.4>;tag=9863
Call-ID: 000041e3-38e3985e3387100098f70080f0c171e8@10.60.0.2
CSeq: 2 INVITE
Contact: sip:sttc-ncp1000@10.60.0.2:35060
Supported: timer,100rel
Authorization: Digest realm="asterisk", nonce="76aa5b39", algorithm=MD5, uri="sip:81038ХХХХХХХХХХ@10.60.0.4", username="sttc-ncp1000", response="fff1bbbc8b5e7edc02b56fbef2885f3f"
Session-Expires: 180
Allow: INVITE,ACK,CANCEL,BYE,PRACK,OPTIONS,REGISTER,INFO,NOTIFY,UPDATE
Content-Type: application/sdp
User-Agent: Panasonic-MPR11-V8.0102/VSIPGW-V2.3002
Content-Length: 260
v=0
o=- 1 1 IN IP4 10.60.0.3
s=-
c=IN IP4 10.60.0.3
t=0 0
m=audio 12170 RTP/AVP 8 0 18 101
a=rtpmap:8 PCMA/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:18 G729/8000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=ptime:20
a=rtcp:12171
<------------->
--- (15 headers 14 lines) ---
Sending to 10.60.0.2:35060 (no NAT)
Using INVITE request as basis request - 000041e3-38e3985e3387100098f70080f0c171e8@10.60.0.2
Found peer 'sttc-ncp1000' for '189' from 10.60.0.2:35060
Got SDP version 1 and unique parts [- 1 IN IP4 10.60.0.3]
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - (alaw|ulaw|g729), peer - audio=(ulaw|alaw|g729)/video=(nothing)/text=(nothing), combined - (alaw|ulaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 10.60.0.3:12170
Looking for 81038ХХХХХХХХХХ in from-internal (domain 10.60.0.4)
sip_route_dump: route/path hop: <sip:sttc-ncp1000@10.60.0.2:35060>
<--- Transmitting (no NAT) to 10.60.0.2:35060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.60.0.2:35060;branch=z9hG4bK0000697d;received=10.60.0.2;rport=35060
From: "Sokolov S.V." <sip:189@10.60.0.4>;tag=9863
To: sip:81038ХХХХХХХХХХ@10.60.0.4
Call-ID: 000041e3-38e3985e3387100098f70080f0c171e8@10.60.0.2
CSeq: 2 INVITE
Server: FPBX-15.0.16.81(16.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 180;refresher=uas
Contact: <sip:81038ХХХХХХХХХХ@10.60.0.4:5060>
Content-Length: 0
<------------>
Audio is at 18354
Adding codec alaw to SDP
Adding codec ulaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to <PROVIDER SRV IP ADDRESS>:5060:
INVITE sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS> SIP/2.0
Via: SIP/2.0/UDP <EXT SRV IP ADDRESS>:5060;branch=z9hG4bK6153286a;rport
Max-Forwards: 70
From: <sip:495ХХХХХХХ@<EXT SRV IP ADDRESS>>;tag=as0b354112
To: <sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS>>
Contact: <sip:495ХХХХХХХ@<EXT SRV IP ADDRESS>:5060>
Call-ID: 3cacfe5530b1605f066eb0776b491bf0@<EXT SRV IP ADDRESS>:5060
CSeq: 102 INVITE
User-Agent: FPBX-15.0.16.81(16.13.0)
Date: Sat, 05 Dec 2020 11:06:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 276
v=0
o=root 144511809 144511809 IN IP4 <EXT SRV IP ADDRESS>
s=Asterisk PBX 16.13.0
c=IN IP4 <EXT SRV IP ADDRESS>
t=0 0
m=audio 18354 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
---
<--- SIP read from UDP:<PROVIDER SRV IP ADDRESS>:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP <EXT SRV IP ADDRESS>:5060;branch=z9hG4bK6153286a;received=<EXT SRV IP ADDRESS>;rport=5060
From: <sip:495ХХХХХХХ@<EXT SRV IP ADDRESS>>;tag=as0b354112
To: <sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS>>;tag=as299baccc
Call-ID: 3cacfe5530b1605f066eb0776b491bf0@<EXT SRV IP ADDRESS>:5060
CSeq: 102 INVITE
Server: FPBX-2.11.0(11.17.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="65982f24"
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
Transmitting (NAT) to <PROVIDER SRV IP ADDRESS>:5060:
ACK sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS> SIP/2.0
Via: SIP/2.0/UDP <EXT SRV IP ADDRESS>:5060;branch=z9hG4bK6153286a;rport
Max-Forwards: 70
From: <sip:495ХХХХХХХ@<EXT SRV IP ADDRESS>>;tag=as0b354112
To: <sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS>>;tag=as299baccc
Contact: <sip:495ХХХХХХХ@<EXT SRV IP ADDRESS>:5060>
Call-ID: 3cacfe5530b1605f066eb0776b491bf0@<EXT SRV IP ADDRESS>:5060
CSeq: 102 ACK
User-Agent: FPBX-15.0.16.81(16.13.0)
Content-Length: 0
---
Audio is at 18354
Adding codec alaw to SDP
Adding codec ulaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to <PROVIDER SRV IP ADDRESS>:5060:
INVITE sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS> SIP/2.0
Via: SIP/2.0/UDP <EXT SRV IP ADDRESS>:5060;branch=z9hG4bK0277acd4;rport
Max-Forwards: 70
From: <sip:495ХХХХХХХ@<EXT SRV IP ADDRESS>>;tag=as0b354112
To: <sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS>>
Contact: <sip:495ХХХХХХХ@<EXT SRV IP ADDRESS>:5060>
Call-ID: 3cacfe5530b1605f066eb0776b491bf0@<EXT SRV IP ADDRESS>:5060
CSeq: 103 INVITE
User-Agent: FPBX-15.0.16.81(16.13.0)
Authorization: Digest username="495ХХХХХХХ", realm="asterisk", algorithm=MD5, uri="sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS>", nonce="65982f24", response="619cf336f045a45a59e3aa0a6d218dbc"
Date: Sat, 05 Dec 2020 11:06:47 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 276
v=0
o=root 144511809 144511810 IN IP4 <EXT SRV IP ADDRESS>
s=Asterisk PBX 16.13.0
c=IN IP4 <EXT SRV IP ADDRESS>
t=0 0
m=audio 18354 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
---
<--- SIP read from UDP:<PROVIDER SRV IP ADDRESS>:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP <EXT SRV IP ADDRESS>:5060;branch=z9hG4bK0277acd4;received=<EXT SRV IP ADDRESS>;rport=5060
From: <sip:495ХХХХХХХ@<EXT SRV IP ADDRESS>>;tag=as0b354112
To: <sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS>>
Call-ID: 3cacfe5530b1605f066eb0776b491bf0@<EXT SRV IP ADDRESS>:5060
CSeq: 103 INVITE
Server: FPBX-2.11.0(11.17.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS>:5060>
Content-Length: 0
<------------->
--- (12 headers 0 lines) ---
<--- SIP read from UDP:<PROVIDER SRV IP ADDRESS>:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP <EXT SRV IP ADDRESS>:5060;branch=z9hG4bK0277acd4;received=<EXT SRV IP ADDRESS>;rport=5060
From: <sip:495ХХХХХХХ@<EXT SRV IP ADDRESS>>;tag=as0b354112
To: <sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS>>;tag=as4d5bd64c
Call-ID: 3cacfe5530b1605f066eb0776b491bf0@<EXT SRV IP ADDRESS>:5060
CSeq: 103 INVITE
Server: FPBX-2.11.0(11.17.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS>:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 262
v=0
o=root 1929802605 1929802605 IN IP4 <PROVIDER SRV IP ADDRESS>
s=Asterisk PBX 11.17.1
c=IN IP4 <PROVIDER SRV IP ADDRESS>
t=0 0
m=audio 15712 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 12 lines) ---
sip_route_dump: route/path hop: <sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS>:5060>
Got SDP version 1929802605 and unique parts [root 1929802605 IN IP4 <PROVIDER SRV IP ADDRESS>]
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - (alaw|ulaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (alaw|ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port <PROVIDER SRV IP ADDRESS>:15712
Audio is at 18494
Adding codec alaw to SDP
Adding codec ulaw to SDP
Adding codec g729 to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Transmitting (no NAT) to 10.60.0.2:35060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.60.0.2:35060;branch=z9hG4bK0000697d;received=10.60.0.2;rport=35060
From: "Sokolov S.V." <sip:189@10.60.0.4>;tag=9863
To: sip:81038ХХХХХХХХХХ@10.60.0.4;tag=as5a78e927
Call-ID: 000041e3-38e3985e3387100098f70080f0c171e8@10.60.0.2
CSeq: 2 INVITE
Server: FPBX-15.0.16.81(16.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 180;refresher=uas
Contact: <sip:81038ХХХХХХХХХХ@10.60.0.4:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 315
v=0
o=root 225066610 225066610 IN IP4 10.60.0.4
s=Asterisk PBX 16.13.0
c=IN IP4 10.60.0.4
t=0 0
m=audio 18494 RTP/AVP 8 0 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
<------------>
<--- SIP read from UDP:<PROVIDER SRV IP ADDRESS>:5060 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP <EXT SRV IP ADDRESS>:5060;branch=z9hG4bK0277acd4;received=<EXT SRV IP ADDRESS>;rport=5060
From: <sip:495ХХХХХХХ@<EXT SRV IP ADDRESS>>;tag=as0b354112
To: <sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS>>;tag=as4d5bd64c
Call-ID: 3cacfe5530b1605f066eb0776b491bf0@<EXT SRV IP ADDRESS>:5060
CSeq: 103 INVITE
Server: FPBX-2.11.0(11.17.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
X-Asterisk-HangupCause: Circuit/channel congestion
X-Asterisk-HangupCauseCode: 34
Content-Length: 0
<------------->
--- (13 headers 0 lines) ---
Transmitting (NAT) to <PROVIDER SRV IP ADDRESS>:5060:
ACK sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS>:5060 SIP/2.0
Via: SIP/2.0/UDP <EXT SRV IP ADDRESS>:5060;branch=z9hG4bK0277acd4;rport
Max-Forwards: 70
From: <sip:495ХХХХХХХ@<EXT SRV IP ADDRESS>>;tag=as0b354112
To: <sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS>>;tag=as4d5bd64c
Contact: <sip:495ХХХХХХХ@<EXT SRV IP ADDRESS>:5060>
Call-ID: 3cacfe5530b1605f066eb0776b491bf0@<EXT SRV IP ADDRESS>:5060
CSeq: 103 ACK
User-Agent: FPBX-15.0.16.81(16.13.0)
Content-Length: 0
---
Really destroying SIP dialog '3cacfe5530b1605f066eb0776b491bf0@<EXT SRV IP ADDRESS>:5060' Method: INVITE
Audio is at 12850
Adding codec alaw to SDP
Adding codec ulaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to <PROVIDER SRV IP ADDRESS>:5060:
INVITE sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS> SIP/2.0
Via: SIP/2.0/UDP <EXT SRV IP ADDRESS>:5060;branch=z9hG4bK23ad8ef2;rport
Max-Forwards: 70
From: <sip:495XXXXXXX@<EXT SRV IP ADDRESS>>;tag=as0ac50218
To: <sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS>>
Contact: <sip:495XXXXXXX@<EXT SRV IP ADDRESS>:5060>
Call-ID: 1f4619432cdaad63687c777d4cff2f7f@<EXT SRV IP ADDRESS>:5060
CSeq: 102 INVITE
User-Agent: FPBX-15.0.16.81(16.13.0)
Date: Sat, 05 Dec 2020 11:06:59 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 276
v=0
o=root 351939896 351939896 IN IP4 <EXT SRV IP ADDRESS>
s=Asterisk PBX 16.13.0
c=IN IP4 <EXT SRV IP ADDRESS>
t=0 0
m=audio 12850 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
---
<--- SIP read from UDP:<PROVIDER SRV IP ADDRESS>:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP <EXT SRV IP ADDRESS>:5060;branch=z9hG4bK23ad8ef2;received=<EXT SRV IP ADDRESS>;rport=5060
From: <sip:495XXXXXXX@<EXT SRV IP ADDRESS>>;tag=as0ac50218
To: <sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS>>;tag=as556c0c28
Call-ID: 1f4619432cdaad63687c777d4cff2f7f@<EXT SRV IP ADDRESS>:5060
CSeq: 102 INVITE
Server: FPBX-2.11.0(11.17.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="09e97248"
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
Transmitting (NAT) to <PROVIDER SRV IP ADDRESS>:5060:
ACK sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS> SIP/2.0
Via: SIP/2.0/UDP <EXT SRV IP ADDRESS>:5060;branch=z9hG4bK23ad8ef2;rport
Max-Forwards: 70
From: <sip:495XXXXXXX@<EXT SRV IP ADDRESS>>;tag=as0ac50218
To: <sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS>>;tag=as556c0c28
Contact: <sip:495XXXXXXX@<EXT SRV IP ADDRESS>:5060>
Call-ID: 1f4619432cdaad63687c777d4cff2f7f@<EXT SRV IP ADDRESS>:5060
CSeq: 102 ACK
User-Agent: FPBX-15.0.16.81(16.13.0)
Content-Length: 0
---
Audio is at 12850
Adding codec alaw to SDP
Adding codec ulaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to <PROVIDER SRV IP ADDRESS>:5060:
INVITE sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS> SIP/2.0
Via: SIP/2.0/UDP <EXT SRV IP ADDRESS>:5060;branch=z9hG4bK1245c14b;rport
Max-Forwards: 70
From: <sip:495XXXXXXX@<EXT SRV IP ADDRESS>>;tag=as0ac50218
To: <sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS>>
Contact: <sip:495XXXXXXX@<EXT SRV IP ADDRESS>:5060>
Call-ID: 1f4619432cdaad63687c777d4cff2f7f@<EXT SRV IP ADDRESS>:5060
CSeq: 103 INVITE
User-Agent: FPBX-15.0.16.81(16.13.0)
Authorization: Digest username="495XXXXXXX", realm="asterisk", algorithm=MD5, uri="sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS>", nonce="09e97248", response="417ae4a7196a840462491d4de968539f"
Date: Sat, 05 Dec 2020 11:06:59 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 276
v=0
o=root 351939896 351939897 IN IP4 <EXT SRV IP ADDRESS>
s=Asterisk PBX 16.13.0
c=IN IP4 <EXT SRV IP ADDRESS>
t=0 0
m=audio 12850 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
---
<--- SIP read from UDP:<PROVIDER SRV IP ADDRESS>:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP <EXT SRV IP ADDRESS>:5060;branch=z9hG4bK1245c14b;received=<EXT SRV IP ADDRESS>;rport=5060
From: <sip:495XXXXXXX@<EXT SRV IP ADDRESS>>;tag=as0ac50218
To: <sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS>>
Call-ID: 1f4619432cdaad63687c777d4cff2f7f@<EXT SRV IP ADDRESS>:5060
CSeq: 103 INVITE
Server: FPBX-2.11.0(11.17.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS>:5060>
Content-Length: 0
<------------->
--- (12 headers 0 lines) ---
<--- SIP read from UDP:<PROVIDER SRV IP ADDRESS>:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP <EXT SRV IP ADDRESS>:5060;branch=z9hG4bK1245c14b;received=<EXT SRV IP ADDRESS>;rport=5060
From: <sip:495XXXXXXX@<EXT SRV IP ADDRESS>>;tag=as0ac50218
To: <sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS>>;tag=as1b67c728
Call-ID: 1f4619432cdaad63687c777d4cff2f7f@<EXT SRV IP ADDRESS>:5060
CSeq: 103 INVITE
Server: FPBX-2.11.0(11.17.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS>:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 262
v=0
o=root 2140215254 2140215254 IN IP4 <PROVIDER SRV IP ADDRESS>
s=Asterisk PBX 11.17.1
c=IN IP4 <PROVIDER SRV IP ADDRESS>
t=0 0
m=audio 17992 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------->
--- (14 headers 12 lines) ---
sip_route_dump: route/path hop: <sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS>:5060>
Got SDP version 2140215254 and unique parts [root 2140215254 IN IP4 <PROVIDER SRV IP ADDRESS>]
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - (alaw|ulaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (alaw|ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port <PROVIDER SRV IP ADDRESS>:17992
<--- SIP read from UDP:10.60.0.2:35060 --->
CANCEL sip:81038ХХХХХХХХХХ@10.60.0.4 SIP/2.0
Via: SIP/2.0/UDP 10.60.0.2:35060;branch=z9hG4bK0000697d;rport
Max-Forwards: 70
To: sip:81038ХХХХХХХХХХ@10.60.0.4
From: "Sokolov S.V." <sip:189@10.60.0.4>;tag=9863
Call-ID: 000041e3-38e3985e3387100098f70080f0c171e8@10.60.0.2
CSeq: 2 CANCEL
Authorization: Digest realm="asterisk", nonce="76aa5b39", algorithm=MD5, uri="sip:81038ХХХХХХХХХХ@10.60.0.4", username="sttc-ncp1000", response="69dfd10328646aea35eb8f979ebc97cb"
User-Agent: Panasonic-MPR11-V8.0102/VSIPGW-V2.3002
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Sending to 10.60.0.2:35060 (no NAT)
<--- Reliably Transmitting (no NAT) to 10.60.0.2:35060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 10.60.0.2:35060;branch=z9hG4bK0000697d;received=10.60.0.2;rport=35060
From: "Sokolov S.V." <sip:189@10.60.0.4>;tag=9863
To: sip:81038ХХХХХХХХХХ@10.60.0.4;tag=as5a78e927
Call-ID: 000041e3-38e3985e3387100098f70080f0c171e8@10.60.0.2
CSeq: 2 INVITE
Server: FPBX-15.0.16.81(16.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<------------>
<--- Transmitting (no NAT) to 10.60.0.2:35060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.60.0.2:35060;branch=z9hG4bK0000697d;received=10.60.0.2;rport=35060
From: "Sokolov S.V." <sip:189@10.60.0.4>;tag=9863
To: sip:81038ХХХХХХХХХХ@10.60.0.4;tag=as5a78e927
Call-ID: 000041e3-38e3985e3387100098f70080f0c171e8@10.60.0.2
CSeq: 2 CANCEL
Server: FPBX-15.0.16.81(16.13.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '1f4619432cdaad63687c777d4cff2f7f@<EXT SRV IP ADDRESS>:5060' in 6400 ms (Method: INVITE)
Reliably Transmitting (NAT) to <PROVIDER SRV IP ADDRESS>:5060:
CANCEL sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS> SIP/2.0
Via: SIP/2.0/UDP <EXT SRV IP ADDRESS>:5060;branch=z9hG4bK1245c14b;rport
Max-Forwards: 70
From: <sip:495XXXXXXX@<EXT SRV IP ADDRESS>>;tag=as0ac50218
To: <sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS>>
Call-ID: 1f4619432cdaad63687c777d4cff2f7f@<EXT SRV IP ADDRESS>:5060
CSeq: 103 CANCEL
User-Agent: FPBX-15.0.16.81(16.13.0)
Content-Length: 0
---
Scheduling destruction of SIP dialog '1f4619432cdaad63687c777d4cff2f7f@<EXT SRV IP ADDRESS>:5060' in 6400 ms (Method: INVITE)
<--- SIP read from UDP:<PROVIDER SRV IP ADDRESS>:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP <EXT SRV IP ADDRESS>:5060;branch=z9hG4bK1245c14b;received=<EXT SRV IP ADDRESS>;rport=5060
From: <sip:495XXXXXXX@<EXT SRV IP ADDRESS>>;tag=as0ac50218
To: <sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS>>;tag=as1b67c728
Call-ID: 1f4619432cdaad63687c777d4cff2f7f@<EXT SRV IP ADDRESS>:5060
CSeq: 103 INVITE
Server: FPBX-2.11.0(11.17.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Transmitting (NAT) to <PROVIDER SRV IP ADDRESS>:5060:
ACK sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS>:5060 SIP/2.0
Via: SIP/2.0/UDP <EXT SRV IP ADDRESS>:5060;branch=z9hG4bK1245c14b;rport
Max-Forwards: 70
From: <sip:495XXXXXXX@<EXT SRV IP ADDRESS>>;tag=as0ac50218
To: <sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS>>;tag=as1b67c728
Contact: <sip:495XXXXXXX@<EXT SRV IP ADDRESS>:5060>
Call-ID: 1f4619432cdaad63687c777d4cff2f7f@<EXT SRV IP ADDRESS>:5060
CSeq: 103 ACK
User-Agent: FPBX-15.0.16.81(16.13.0)
Content-Length: 0
---
Scheduling destruction of SIP dialog '1f4619432cdaad63687c777d4cff2f7f@<EXT SRV IP ADDRESS>:5060' in 6400 ms (Method: INVITE)
<--- SIP read from UDP:<PROVIDER SRV IP ADDRESS>:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP <EXT SRV IP ADDRESS>:5060;branch=z9hG4bK1245c14b;received=<EXT SRV IP ADDRESS>;rport=5060
From: <sip:495XXXXXXX@<EXT SRV IP ADDRESS>>;tag=as0ac50218
To: <sip:981038ХХХХХХХХХХ@<PROVIDER SRV IP ADDRESS>>;tag=as1b67c728
Call-ID: 1f4619432cdaad63687c777d4cff2f7f@<EXT SRV IP ADDRESS>:5060
CSeq: 103 CANCEL
Server: FPBX-2.11.0(11.17.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
<--- SIP read from UDP:10.60.0.2:35060 --->
ACK sip:81038ХХХХХХХХХХ@10.60.0.4 SIP/2.0
Via: SIP/2.0/UDP 10.60.0.2:35060;branch=z9hG4bK0000697d;rport
Max-Forwards: 70
To: sip:81038ХХХХХХХХХХ@10.60.0.4;tag=as5a78e927
From: "Sokolov S.V." <sip:189@10.60.0.4>;tag=9863
Call-ID: 000041e3-38e3985e3387100098f70080f0c171e8@10.60.0.2
CSeq: 2 ACK
Authorization: Digest realm="asterisk", nonce="76aa5b39", algorithm=MD5, uri="sip:81038ХХХХХХХХХХ@10.60.0.4", username="sttc-ncp1000", response="5919fa45d25dbc1fc8145e193ece7d60"
Content-Length: 0