Продажа IP телефонов Fanvil по низким ценам

Не проходят международные звонки

Технический форум по цифровым АТС Panasonic, серии KX-TDA, KX-TDE, KX-NCP. Модели: Panasonic KX-TDA30, Panasonic KX-TDA100, Panasonic KX-TDA100D, Panasonic KX-TDA200, Panasonic KX-TDA600, Panasonic KX-TDE100, Panasonic KX-TDE200, Panasonic KX-TDE600, Panasonic KX-NCP500, Panasonic KX-NCP1000.

Модераторы: Wi$e, Mammon

Не проходят международные звонки

Сообщение Simmer » 06 дек 2020, 23:01

Добрый вечер. Очень прошу вашей помощи. Прошу сильно не пинать, т.к. с темой телефонии я только начал разбираться.

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

<--- 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
Simmer
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Re: Не проходят международные звонки

Сообщение SergA » 07 дек 2020, 09:38

Спрашиваете в теме Panasonic, а какие-то логи показываете от астериска...
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Re: Не проходят международные звонки

Сообщение Simmer » 07 дек 2020, 10:02

Так, мне кажется, что Panasonic что-то не передает Asterisk... из-за этого такое и получается. С Asterisk то звонок проходит...
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Продажа IP телефонов Fanvil по низким ценам


Re: Не проходят международные звонки

Сообщение SergA » 07 дек 2020, 10:34

Так тогда для правильного сравнения средствами Panasonic соберите 2 звонка - с 8-кой, но один из них - международка.
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Re: Не проходят международные звонки

Сообщение Simmer » 08 дек 2020, 11:47

Сделал лог удачного звонка на мобильный клиентом Panasonic. Не могу понять... нету разницы между проваленной международкой и этим звонком. Оба идут одинаково, но где на этом звонке SIP/2.0 180 Ringing, на международке SIP/2.0 503 Service Unavailable
<--- SIP read from UDP:10.60.0.2:35060 --->
INVITE sip:8985XXXXXXX@10.60.0.4 SIP/2.0
Via: SIP/2.0/UDP 10.60.0.2:35060;branch=z9hG4bK00003bb3;rport
Max-Forwards: 70
To: sip:8985XXXXXXX@10.60.0.4
From: "��������" <sip:101@10.60.0.4>;tag=3544
Call-ID: 000043d8-38e3985e3bdd100098ff0080f0c171e8@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 12238 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:12239
<------------->
[2020-12-08 09:33:57] VERBOSE[2447] chan_sip.c: --- (14 headers 14 lines) ---
[2020-12-08 09:33:57] VERBOSE[2447] chan_sip.c: Sending to 10.60.0.2:35060 (NAT)
[2020-12-08 09:33:57] VERBOSE[2447][C-0000001e] chan_sip.c: Sending to 10.60.0.2:35060 (NAT)
[2020-12-08 09:33:57] VERBOSE[2447][C-0000001e] chan_sip.c: Using INVITE request as basis request - 000043d8-38e3985e3bdd100098ff0080f0c171e8@10.60.0.2
[2020-12-08 09:33:57] VERBOSE[2447][C-0000001e] chan_sip.c: Found peer 'sttc-ncp1000' for '101' from 10.60.0.2:35060
[2020-12-08 09:33:57] VERBOSE[2447][C-0000001e] chan_sip.c:
<--- 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=z9hG4bK00003bb3;received=10.60.0.2;rport=35060
From: "��������" <sip:101@10.60.0.4>;tag=3544
To: sip:8985XXXXXXX@10.60.0.4;tag=as5abc2232
Call-ID: 000043d8-38e3985e3bdd100098ff0080f0c171e8@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="14161160"
Content-Length: 0


<------------>
[2020-12-08 09:33:57] VERBOSE[2447][C-0000001e] chan_sip.c: Scheduling destruction of SIP dialog '000043d8-38e3985e3bdd100098ff0080f0c171e8@10.60.0.2' in 6400 ms (Method: INVITE)
[2020-12-08 09:33:57] VERBOSE[2447] chan_sip.c:
<--- SIP read from UDP:10.60.0.2:35060 --->
ACK sip:8985XXXXXXX@10.60.0.4 SIP/2.0
Via: SIP/2.0/UDP 10.60.0.2:35060;branch=z9hG4bK00003bb3;rport
Max-Forwards: 70
To: sip:8985XXXXXXX@10.60.0.4;tag=as5abc2232
From: "��������" <sip:101@10.60.0.4>;tag=3544
Call-ID: 000043d8-38e3985e3bdd100098ff0080f0c171e8@10.60.0.2
CSeq: 1 ACK
Content-Length: 0

<------------->
[2020-12-08 09:33:57] VERBOSE[2447] chan_sip.c: --- (8 headers 0 lines) ---
[2020-12-08 09:33:57] VERBOSE[2447] chan_sip.c:
<--- SIP read from UDP:10.60.0.2:35060 --->
INVITE sip:8985XXXXXXX@10.60.0.4 SIP/2.0
Via: SIP/2.0/UDP 10.60.0.2:35060;branch=z9hG4bK00002e3b;rport
Max-Forwards: 70
To: sip:8985XXXXXXX@10.60.0.4
From: "��������" <sip:101@10.60.0.4>;tag=3544
Call-ID: 000043d8-38e3985e3bdd100098ff0080f0c171e8@10.60.0.2
CSeq: 2 INVITE
Contact: sip:sttc-ncp1000@10.60.0.2:35060
Supported: timer,100rel
Authorization: Digest realm="asterisk", nonce="14161160", algorithm=MD5, uri="sip:8985XXXXXXX@10.60.0.4", username="sttc-ncp1000", response="7aef50edec7068cdc72566815c6c364e"
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 12238 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:12239
<------------->
[2020-12-08 09:33:57] VERBOSE[2447] chan_sip.c: --- (15 headers 14 lines) ---
[2020-12-08 09:33:57] VERBOSE[2447][C-0000001e] chan_sip.c: Sending to 10.60.0.2:35060 (no NAT)
[2020-12-08 09:33:57] VERBOSE[2447][C-0000001e] chan_sip.c: Using INVITE request as basis request - 000043d8-38e3985e3bdd100098ff0080f0c171e8@10.60.0.2
[2020-12-08 09:33:57] VERBOSE[2447][C-0000001e] chan_sip.c: Found peer 'sttc-ncp1000' for '101' from 10.60.0.2:35060
[2020-12-08 09:33:57] VERBOSE[2447][C-0000001e] netsock2.c: Using SIP RTP TOS bits 184
[2020-12-08 09:33:57] VERBOSE[2447][C-0000001e] netsock2.c: Using SIP RTP CoS mark 5
[2020-12-08 09:33:57] VERBOSE[2447][C-0000001e] chan_sip.c: Got SDP version 1 and unique parts [- 1 IN IP4 10.60.0.3]
[2020-12-08 09:33:57] VERBOSE[2447][C-0000001e] chan_sip.c: Found RTP audio format 8
[2020-12-08 09:33:57] VERBOSE[2447][C-0000001e] chan_sip.c: Found RTP audio format 0
[2020-12-08 09:33:57] VERBOSE[2447][C-0000001e] chan_sip.c: Found RTP audio format 18
[2020-12-08 09:33:57] VERBOSE[2447][C-0000001e] chan_sip.c: Found RTP audio format 101
[2020-12-08 09:33:57] VERBOSE[2447][C-0000001e] chan_sip.c: Found audio description format PCMA for ID 8
[2020-12-08 09:33:57] VERBOSE[2447][C-0000001e] chan_sip.c: Found audio description format PCMU for ID 0
[2020-12-08 09:33:57] VERBOSE[2447][C-0000001e] chan_sip.c: Found audio description format G729 for ID 18
[2020-12-08 09:33:57] VERBOSE[2447][C-0000001e] chan_sip.c: Found audio description format telephone-event for ID 101
[2020-12-08 09:33:57] VERBOSE[2447][C-0000001e] chan_sip.c: Capabilities: us - (alaw|ulaw|g729), peer - audio=(ulaw|alaw|g729)/video=(nothing)/text=(nothing), combined - (alaw|ulaw|g729)
[2020-12-08 09:33:57] VERBOSE[2447][C-0000001e] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[2020-12-08 09:33:57] VERBOSE[2447][C-0000001e] chan_sip.c: Peer audio RTP is at port 10.60.0.3:12238
[2020-12-08 09:33:57] VERBOSE[2447][C-0000001e] chan_sip.c: Looking for 8985XXXXXXX in from-internal (domain 10.60.0.4)
[2020-12-08 09:33:57] VERBOSE[2447][C-0000001e] sip/route.c: sip_route_dump: route/path hop: <sip:sttc-ncp1000@10.60.0.2:35060>
[2020-12-08 09:33:57] VERBOSE[2447][C-0000001e] chan_sip.c:
<--- 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=z9hG4bK00002e3b;received=10.60.0.2;rport=35060
From: "��������" <sip:101@10.60.0.4>;tag=3544
To: sip:8985XXXXXXX@10.60.0.4
Call-ID: 000043d8-38e3985e3bdd100098ff0080f0c171e8@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:8985XXXXXXX@10.60.0.4:5060>
Content-Length: 0


<------------>
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [8985XXXXXXX@from-internal:1] Macro("SIP/sttc-ncp1000-00000039", "user-callerid,LIMIT,EXTERNAL,") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-user-callerid:1] Set("SIP/sttc-ncp1000-00000039", "TOUCH_MONITOR=1607409237.57") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-user-callerid:2] Set("SIP/sttc-ncp1000-00000039", "AMPUSER=101") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-user-callerid:3] Set("SIP/sttc-ncp1000-00000039", "HOTDESCKCHAN=sttc-ncp1000-00000039") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-user-callerid:4] Set("SIP/sttc-ncp1000-00000039", "HOTDESKEXTEN=sttc") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-user-callerid:5] Set("SIP/sttc-ncp1000-00000039", "HOTDESKCALL=0") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-user-callerid:6] ExecIf("SIP/sttc-ncp1000-00000039", "0?Set(HOTDESKCALL=1)") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-user-callerid:7] ExecIf("SIP/sttc-ncp1000-00000039", "0?Set(CALLERID(name)=)") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-user-callerid:8] GotoIf("SIP/sttc-ncp1000-00000039", "0?report") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-user-callerid:9] ExecIf("SIP/sttc-ncp1000-00000039", "1?Set(REALCALLERIDNUM=101)") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-user-callerid:10] Set("SIP/sttc-ncp1000-00000039", "AMPUSER=") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-user-callerid:11] GotoIf("SIP/sttc-ncp1000-00000039", "0?limit") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-user-callerid:12] Set("SIP/sttc-ncp1000-00000039", "AMPUSERCIDNAME=") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-user-callerid:13] ExecIf("SIP/sttc-ncp1000-00000039", "0?Set(__CIDMASQUERADING=TRUE)") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-user-callerid:14] GotoIf("SIP/sttc-ncp1000-00000039", "1?report") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx_builtins.c: Goto (macro-user-callerid,s,23)
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-user-callerid:23] NoOp("SIP/sttc-ncp1000-00000039", "Macro Depth is 1") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-user-callerid:24] GotoIf("SIP/sttc-ncp1000-00000039", "1?report2:macroerror") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx_builtins.c: Goto (macro-user-callerid,s,25)
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-user-callerid:25] GotoIf("SIP/sttc-ncp1000-00000039", "1?continue") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx_builtins.c: Goto (macro-user-callerid,s,44)
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-user-callerid:44] Set("SIP/sttc-ncp1000-00000039", "CALLERID(number)=101") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-user-callerid:45] Set("SIP/sttc-ncp1000-00000039", "CALLERID(name)=��������") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-user-callerid:46] GotoIf("SIP/sttc-ncp1000-00000039", "0?cnum") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-user-callerid:47] Set("SIP/sttc-ncp1000-00000039", "CDR(cnam)=��������") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-user-callerid:48] Set("SIP/sttc-ncp1000-00000039", "CDR(cnum)=101") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-user-callerid:49] Set("SIP/sttc-ncp1000-00000039", "CHANNEL(language)=ru") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [8985XXXXXXX@from-internal:2] Gosub("SIP/sttc-ncp1000-00000039", "sub-record-check,s,1(out,8985XXXXXXX,dontcare)") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@sub-record-check:1] GotoIf("SIP/sttc-ncp1000-00000039", "0?initialized") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@sub-record-check:2] Set("SIP/sttc-ncp1000-00000039", "__REC_STATUS=INITIALIZED") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@sub-record-check:3] Set("SIP/sttc-ncp1000-00000039", "NOW=1607409237") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@sub-record-check:4] Set("SIP/sttc-ncp1000-00000039", "__DAY=08") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@sub-record-check:5] Set("SIP/sttc-ncp1000-00000039", "__MONTH=12") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@sub-record-check:6] Set("SIP/sttc-ncp1000-00000039", "__YEAR=2020") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@sub-record-check:7] Set("SIP/sttc-ncp1000-00000039", "__TIMESTR=20201208-093357") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@sub-record-check:8] Set("SIP/sttc-ncp1000-00000039", "__FROMEXTEN=101") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@sub-record-check:9] Set("SIP/sttc-ncp1000-00000039", "__MON_FMT=wav") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@sub-record-check:10] NoOp("SIP/sttc-ncp1000-00000039", "Recordings initialized") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@sub-record-check:11] ExecIf("SIP/sttc-ncp1000-00000039", "0?Set(ARG3=dontcare)") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@sub-record-check:12] Set("SIP/sttc-ncp1000-00000039", "REC_POLICY_MODE_SAVE=") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@sub-record-check:13] ExecIf("SIP/sttc-ncp1000-00000039", "0?Set(REC_STATUS=NO)") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@sub-record-check:14] GotoIf("SIP/sttc-ncp1000-00000039", "3?checkaction") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx_builtins.c: Goto (sub-record-check,s,17)
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@sub-record-check:17] GotoIf("SIP/sttc-ncp1000-00000039", "1?sub-record-check,out,1") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx_builtins.c: Goto (sub-record-check,out,1)
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [out@sub-record-check:1] NoOp("SIP/sttc-ncp1000-00000039", "Outbound Recording Check from 101 to 8985XXXXXXX") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [out@sub-record-check:2] Set("SIP/sttc-ncp1000-00000039", "RECMODE=") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [out@sub-record-check:3] ExecIf("SIP/sttc-ncp1000-00000039", "1?Goto(routewins)") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx_builtins.c: Goto (sub-record-check,out,7)
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [out@sub-record-check:7] Gosub("SIP/sttc-ncp1000-00000039", "recordcheck,1(dontcare,out,8985XXXXXXX)") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [recordcheck@sub-record-check:1] NoOp("SIP/sttc-ncp1000-00000039", "Starting recording check against dontcare") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [recordcheck@sub-record-check:2] Goto("SIP/sttc-ncp1000-00000039", "dontcare") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx_builtins.c: Goto (sub-record-check,recordcheck,3)
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [recordcheck@sub-record-check:3] Return("SIP/sttc-ncp1000-00000039", "") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [out@sub-record-check:8] Return("SIP/sttc-ncp1000-00000039", "") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [8985XXXXXXX@from-internal:3] ExecIf("SIP/sttc-ncp1000-00000039", "0 ?Set(CDR(accountcode)=)") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [8985XXXXXXX@from-internal:4] Set("SIP/sttc-ncp1000-00000039", "_ROUTEID=7") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [8985XXXXXXX@from-internal:5] Set("SIP/sttc-ncp1000-00000039", "_ROUTENAME=Мобильные") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [8985XXXXXXX@from-internal:6] Set("SIP/sttc-ncp1000-00000039", "MOHCLASS=default") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [8985XXXXXXX@from-internal:7] Set("SIP/sttc-ncp1000-00000039", "_CALLERIDNAMEINTERNAL=��������") in new stack
[2020-12-08 09:33:57] ERROR[5293][C-0000001e] json.c: Error building JSON from '{s: s, s: s}': Invalid UTF-8 string.
[2020-12-08 09:33:57] ERROR[5293][C-0000001e] : Got 13 backtrace records
# 0: /usr/sbin/asterisk(ast_json_vpack+0xb5) [0x507522]
# 1: /usr/sbin/asterisk(ast_json_pack+0x9f) [0x50745d]
# 2: /usr/sbin/asterisk(ast_channel_publish_varset+0x3b) [0x59a671]
# 3: /usr/sbin/asterisk(pbx_builtin_setvar_helper+0x34f) [0x5486f4]
# 4: /usr/sbin/asterisk(pbx_builtin_setvar+0x176) [0x5488de]
# 5: /usr/sbin/asterisk(pbx_exec+0x11c) [0x53e3a9]
# 6: /usr/sbin/asterisk() [0x529dd3]
# 7: /usr/sbin/asterisk(ast_spawn_extension+0x64) [0x52dbdf]
# 8: /usr/sbin/asterisk() [0x52e899]
# 9: /usr/sbin/asterisk() [0x530070]
#10: /usr/sbin/asterisk() [0x5bff17]
#11: /lib64/libpthread.so.0(+0x7ea5) [0x7f3595859ea5]
#12: /lib64/libc.so.6(clone+0x6d) [0x7f35948f88dd]

[2020-12-08 09:33:57] ERROR[5293][C-0000001e] stasis_channels.c: Error creating message
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [8985XXXXXXX@from-internal:8] Set("SIP/sttc-ncp1000-00000039", "_CALLERIDNUMINTERNAL=101") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [8985XXXXXXX@from-internal:9] Set("SIP/sttc-ncp1000-00000039", "_EMAILNOTIFICATION=FALSE") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [8985XXXXXXX@from-internal:10] Set("SIP/sttc-ncp1000-00000039", "_NODEST=") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [8985XXXXXXX@from-internal:11] Macro("SIP/sttc-ncp1000-00000039", "dialout-trunk,1,98985XXXXXXX,,off") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-dialout-trunk:1] Set("SIP/sttc-ncp1000-00000039", "DIAL_TRUNK=1") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-dialout-trunk:2] ExecIf("SIP/sttc-ncp1000-00000039", "0?Set(DIAL_OPTIONS=Hhtr)") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-dialout-trunk:3] GosubIf("SIP/sttc-ncp1000-00000039", "0?sub-pincheck,s,1()") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-dialout-trunk:4] ExecIf("SIP/sttc-ncp1000-00000039", "0?Set(CALLERID(num)=)") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-dialout-trunk:5] GotoIf("SIP/sttc-ncp1000-00000039", "0?disabletrunk,1") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-dialout-trunk:6] Set("SIP/sttc-ncp1000-00000039", "DIAL_NUMBER=98985XXXXXXX") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-dialout-trunk:7] Set("SIP/sttc-ncp1000-00000039", "DIAL_TRUNK_OPTIONS=HhTtr") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-dialout-trunk:8] Set("SIP/sttc-ncp1000-00000039", "OUTBOUND_GROUP=OUT_1") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-dialout-trunk:9] Set("SIP/sttc-ncp1000-00000039", "DIAL_TRUNK_OPTIONS=T") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-dialout-trunk:10] GotoIf("SIP/sttc-ncp1000-00000039", "0?nomax") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-dialout-trunk:11] GotoIf("SIP/sttc-ncp1000-00000039", "0?chanfull") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-dialout-trunk:12] GotoIf("SIP/sttc-ncp1000-00000039", "0?skipoutcid") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-dialout-trunk:13] Macro("SIP/sttc-ncp1000-00000039", "outbound-callerid,1") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-outbound-callerid:1] NoOp("SIP/sttc-ncp1000-00000039", "101") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-outbound-callerid:2] NoOp("SIP/sttc-ncp1000-00000039", "") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-outbound-callerid:3] NoOp("SIP/sttc-ncp1000-00000039", "off") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-outbound-callerid:4] ExecIf("SIP/sttc-ncp1000-00000039", "0?Set(CALLERPRES(name-pres)=)") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-outbound-callerid:5] ExecIf("SIP/sttc-ncp1000-00000039", "0?Set(CALLERPRES(num-pres)=)") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-outbound-callerid:6] Set("SIP/sttc-ncp1000-00000039", "HOTDESCKCHAN=sttc-ncp1000-00000039") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-outbound-callerid:7] Set("SIP/sttc-ncp1000-00000039", "HOTDESKEXTEN=sttc") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-outbound-callerid:8] Set("SIP/sttc-ncp1000-00000039", "HOTDESKCALL=0") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-outbound-callerid:9] ExecIf("SIP/sttc-ncp1000-00000039", "0?Set(HOTDESKCALL=1)") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-outbound-callerid:10] ExecIf("SIP/sttc-ncp1000-00000039", "0?Set(CALLERID(name)=)") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-outbound-callerid:11] Set("SIP/sttc-ncp1000-00000039", "ALLOWTHISROUTE=NO") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-outbound-callerid:12] ExecIf("SIP/sttc-ncp1000-00000039", "0?Set(ALLOWTHISROUTE=YES)") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-outbound-callerid:13] ExecIf("SIP/sttc-ncp1000-00000039", "0?Hangup()") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-outbound-callerid:14] ExecIf("SIP/sttc-ncp1000-00000039", "0?Set(REALCALLERIDNUM=101)") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-outbound-callerid:15] ExecIf("SIP/sttc-ncp1000-00000039", "0?Set(AMPUSER=101)") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-outbound-callerid:16] GotoIf("SIP/sttc-ncp1000-00000039", "1?normcid") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx_builtins.c: Goto (macro-outbound-callerid,s,20)
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-outbound-callerid:20] Set("SIP/sttc-ncp1000-00000039", "USEROUTCID=") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-outbound-callerid:21] Set("SIP/sttc-ncp1000-00000039", "EMERGENCYCID=") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-outbound-callerid:22] ExecIf("SIP/sttc-ncp1000-00000039", "0?Set(EMERGENCYCID=)") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-outbound-callerid:23] Set("SIP/sttc-ncp1000-00000039", "TRUNKOUTCID=495XXXXXXX") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-outbound-callerid:24] GotoIf("SIP/sttc-ncp1000-00000039", "1?trunkcid") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx_builtins.c: Goto (macro-outbound-callerid,s,30)
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-outbound-callerid:30] ExecIf("SIP/sttc-ncp1000-00000039", "1?Set(CALLERID(all)=495XXXXXXX)") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-outbound-callerid:31] ExecIf("SIP/sttc-ncp1000-00000039", "0?Set(CALLERID(all)=)") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-outbound-callerid:32] ExecIf("SIP/sttc-ncp1000-00000039", "0?Set(CALLERID(all)=)") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-outbound-callerid:33] ExecIf("SIP/sttc-ncp1000-00000039", "0?Set(CALLERID(all)=101)") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-outbound-callerid:34] Set("SIP/sttc-ncp1000-00000039", "TIOHIDE=no") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-outbound-callerid:35] ExecIf("SIP/sttc-ncp1000-00000039", "0?Set(CALLERPRES(name-pres)=prohib_passed_screen)") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-outbound-callerid:36] ExecIf("SIP/sttc-ncp1000-00000039", "0?Set(CALLERPRES(num-pres)=prohib_passed_screen)") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-outbound-callerid:37] ExecIf("SIP/sttc-ncp1000-00000039", "0?Set(CALLERPRES(name-pres)=prohib_passed_screen)") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-outbound-callerid:38] ExecIf("SIP/sttc-ncp1000-00000039", "0?Set(CALLERPRES(num-pres)=prohib_passed_screen)") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-outbound-callerid:39] Set("SIP/sttc-ncp1000-00000039", "CDR(outbound_cnum)=495XXXXXXX") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-outbound-callerid:40] Set("SIP/sttc-ncp1000-00000039", "CDR(outbound_cnam)=") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-dialout-trunk:14] GosubIf("SIP/sttc-ncp1000-00000039", "0?sub-flp-1,s,1()") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-dialout-trunk:15] Set("SIP/sttc-ncp1000-00000039", "OUTNUM=98985XXXXXXX") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-dialout-trunk:16] Set("SIP/sttc-ncp1000-00000039", "custom=SIP/selectro(7455995)") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-dialout-trunk:17] ExecIf("SIP/sttc-ncp1000-00000039", "0?Set(DIAL_TRUNK_OPTIONS=M(setmusic^default)T)") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-dialout-trunk:18] ExecIf("SIP/sttc-ncp1000-00000039", "0?Set(DIAL_TRUNK_OPTIONS=TM(confirm))") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-dialout-trunk:19] Macro("SIP/sttc-ncp1000-00000039", "dialout-trunk-predial-hook,") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-dialout-trunk-predial-hook:1] Set("SIP/sttc-ncp1000-00000039", "CALLERID(name)=") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-dialout-trunk-predial-hook:2] Set("SIP/sttc-ncp1000-00000039", "CALLERID(all)= <495XXXXXXX>") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-dialout-trunk-predial-hook:3] MacroExit("SIP/sttc-ncp1000-00000039", "") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-dialout-trunk:20] GotoIf("SIP/sttc-ncp1000-00000039", "0?skipcrm") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-dialout-trunk:21] Set("SIP/sttc-ncp1000-00000039", "__CRM_DIRECTION=OUTBOUND") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-dialout-trunk:22] Set("SIP/sttc-ncp1000-00000039", "__CRM_DESTINATION=98985XXXXXXX") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-dialout-trunk:23] Set("SIP/sttc-ncp1000-00000039", "__CRM_SOURCE=") in new stack
[2020-12-08 09:33:57] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-dialout-trunk:24] AGI("SIP/sttc-ncp1000-00000039", "agi://127.0.0.1/sangomacrm.agi") in new stack
[2020-12-08 09:33:57] VERBOSE[2447] chan_sip.c: Really destroying SIP dialog '1e5008d1798757c752fa0efa21ffdb3d@<PROVIDER SRV IP ADDRESS>:5060' Method: OPTIONS
[2020-12-08 09:33:57] VERBOSE[2447] chan_sip.c: Really destroying SIP dialog '38eceb7948cabfb9475212f427a86266@<PROVIDER SRV IP ADDRESS>:5060' Method: NOTIFY
[2020-12-08 09:33:58] VERBOSE[5293][C-0000001e] res_agi.c: <SIP/sttc-ncp1000-00000039>AGI Script agi://127.0.0.1/sangomacrm.agi completed, returning 0
[2020-12-08 09:33:58] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-dialout-trunk:25] Set("SIP/sttc-ncp1000-00000039", "CHANNEL(hangup_handler_push)=crm-hangup,s,1") in new stack
[2020-12-08 09:33:58] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-dialout-trunk:26] NoOp("SIP/sttc-ncp1000-00000039", "CRM Finished") in new stack
[2020-12-08 09:33:58] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-dialout-trunk:27] GotoIf("SIP/sttc-ncp1000-00000039", "0?bypass,1") in new stack
[2020-12-08 09:33:58] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-dialout-trunk:28] ExecIf("SIP/sttc-ncp1000-00000039", "0?Set(CONNECTEDLINE(num,i)=98985XXXXXXX)") in new stack
[2020-12-08 09:33:58] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-dialout-trunk:29] ExecIf("SIP/sttc-ncp1000-00000039", "0?Set(CONNECTEDLINE(name,i)=CID:495XXXXXXX)") in new stack
[2020-12-08 09:33:58] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-dialout-trunk:30] ExecIf("SIP/sttc-ncp1000-00000039", "0?Set(CONNECTEDLINE(name,i)=CID:(Hidden)495XXXXXXX)") in new stack
[2020-12-08 09:33:58] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-dialout-trunk:31] GotoIf("SIP/sttc-ncp1000-00000039", "0?customtrunk") in new stack
[2020-12-08 09:33:58] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-dialout-trunk:32] ExecIf("SIP/sttc-ncp1000-00000039", "0?Set(DIAL_TRUNK_OPTIONS=)") in new stack
[2020-12-08 09:33:58] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-dialout-trunk:33] Set("SIP/sttc-ncp1000-00000039", "HASH(__SIPHEADERS,Alert-Info)=unset") in new stack
[2020-12-08 09:33:58] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@macro-dialout-trunk:34] Dial("SIP/sttc-ncp1000-00000039", "SIP/selectro(7455995)/98985XXXXXXX,300,Tb(func-apply-sipheaders^s^1,(1))") in new stack
[2020-12-08 09:33:58] VERBOSE[5293][C-0000001e] netsock2.c: Using SIP RTP TOS bits 184
[2020-12-08 09:33:58] VERBOSE[5293][C-0000001e] netsock2.c: Using SIP RTP CoS mark 5
[2020-12-08 09:33:58] ERROR[5293][C-0000001e] json.c: Error building JSON from '{s: s, s: s}': Invalid UTF-8 string.
[2020-12-08 09:33:58] ERROR[5293][C-0000001e] : Got 19 backtrace records
# 0: /usr/sbin/asterisk(ast_json_vpack+0xb5) [0x507522]
# 1: /usr/sbin/asterisk(ast_json_pack+0x9f) [0x50745d]
# 2: /usr/sbin/asterisk(ast_channel_publish_varset+0x3b) [0x59a671]
# 3: /usr/sbin/asterisk(ast_channel_inherit_variables+0x22c) [0x4a9a46]
# 4: /usr/lib64/asterisk/modules/app_dial.so(+0xc79d) [0x7f354254479d]
# 5: /usr/lib64/asterisk/modules/app_dial.so(+0x10045) [0x7f3542548045]
# 6: /usr/sbin/asterisk(pbx_exec+0x11c) [0x53e3a9]
# 7: /usr/sbin/asterisk() [0x529dd3]
# 8: /usr/sbin/asterisk(ast_spawn_extension+0x64) [0x52dbdf]
# 9: /usr/lib64/asterisk/modules/app_macro.so(+0x32ec) [0x7f353e5ef2ec]
#10: /usr/lib64/asterisk/modules/app_macro.so(+0x45dc) [0x7f353e5f05dc]
#11: /usr/sbin/asterisk(pbx_exec+0x11c) [0x53e3a9]
#12: /usr/sbin/asterisk() [0x529dd3]
#13: /usr/sbin/asterisk(ast_spawn_extension+0x64) [0x52dbdf]
#14: /usr/sbin/asterisk() [0x52e899]
#15: /usr/sbin/asterisk() [0x530070]
#16: /usr/sbin/asterisk() [0x5bff17]
#17: /lib64/libpthread.so.0(+0x7ea5) [0x7f3595859ea5]
#18: /lib64/libc.so.6(clone+0x6d) [0x7f35948f88dd]

[2020-12-08 09:33:58] ERROR[5293][C-0000001e] stasis_channels.c: Error creating message
[2020-12-08 09:33:58] VERBOSE[5293][C-0000001e] app_stack.c: SIP/selectro(7455995)-0000003a Internal Gosub(func-apply-sipheaders,s,1(1)) start
[2020-12-08 09:33:58] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@func-apply-sipheaders:1] ExecIf("SIP/selectro(7455995)-0000003a", "0?Set(CHANNEL(hangup_handler_push)=crm-hangup,s,1)") in new stack
[2020-12-08 09:33:58] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@func-apply-sipheaders:2] NoOp("SIP/selectro(7455995)-0000003a", "Applying SIP Headers to channel SIP/selectro(7455995)-0000003a") in new stack
[2020-12-08 09:33:58] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@func-apply-sipheaders:3] Set("SIP/selectro(7455995)-0000003a", "TECH=SIP") in new stack
[2020-12-08 09:33:58] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@func-apply-sipheaders:4] Set("SIP/selectro(7455995)-0000003a", "SIPHEADERKEYS=Alert-Info") in new stack
[2020-12-08 09:33:58] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@func-apply-sipheaders:5] While("SIP/selectro(7455995)-0000003a", "1") in new stack
[2020-12-08 09:33:58] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@func-apply-sipheaders:6] Set("SIP/selectro(7455995)-0000003a", "sipheader=unset") in new stack
[2020-12-08 09:33:58] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@func-apply-sipheaders:7] ExecIf("SIP/selectro(7455995)-0000003a", "1?SIPRemoveHeader(Alert-Info:)") in new stack
[2020-12-08 09:33:58] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@func-apply-sipheaders:8] ExecIf("SIP/selectro(7455995)-0000003a", "0?Set(sipheader=<http://127.0.0.1>;info=unset)") in new stack
[2020-12-08 09:33:58] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@func-apply-sipheaders:9] ExecIf("SIP/selectro(7455995)-0000003a", "0?Set(sipheader=<http://127.0.0.1>unset)") in new stack
[2020-12-08 09:33:58] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@func-apply-sipheaders:10] ExecIf("SIP/selectro(7455995)-0000003a", "0?SIPAddHeader(Alert-Info:unset)") in new stack
[2020-12-08 09:33:58] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@func-apply-sipheaders:11] EndWhile("SIP/selectro(7455995)-0000003a", "") in new stack
[2020-12-08 09:33:58] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@func-apply-sipheaders:5] While("SIP/selectro(7455995)-0000003a", "0") in new stack
[2020-12-08 09:33:58] VERBOSE[5293][C-0000001e] pbx.c: Executing [s@func-apply-sipheaders:12] Return("SIP/selectro(7455995)-0000003a", "") in new stack
[2020-12-08 09:33:58] VERBOSE[5293][C-0000001e] app_stack.c: Spawn extension (from-trunk, 8985XXXXXXX, 1) exited non-zero on 'SIP/selectro(7455995)-0000003a'
[2020-12-08 09:33:58] VERBOSE[5293][C-0000001e] app_stack.c: SIP/selectro(7455995)-0000003a Internal Gosub(func-apply-sipheaders,s,1(1)) complete GOSUB_RETVAL=
[2020-12-08 09:33:58] VERBOSE[5293][C-0000001e] chan_sip.c: Audio is at 15484
[2020-12-08 09:33:58] VERBOSE[5293][C-0000001e] chan_sip.c: Adding codec alaw to SDP
[2020-12-08 09:33:58] VERBOSE[5293][C-0000001e] chan_sip.c: Adding codec ulaw to SDP
[2020-12-08 09:33:58] VERBOSE[5293][C-0000001e] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[2020-12-08 09:33:58] VERBOSE[5293][C-0000001e] chan_sip.c: Reliably Transmitting (NAT) to <PROVIDER SRV IP ADDRESS>:5060:
INVITE sip:98985XXXXXXX@<PROVIDER SRV IP ADDRESS> SIP/2.0
Via: SIP/2.0/UDP <EXT SRV IP ADDRESS>:5060;branch=z9hG4bK535f9334;rport
Max-Forwards: 70
From: <sip:495XXXXXXX@<EXT SRV IP ADDRESS>>;tag=as17d1cdc1
To: <sip:98985XXXXXXX@<PROVIDER SRV IP ADDRESS>>
Contact: <sip:495XXXXXXX@<EXT SRV IP ADDRESS>:5060>
Call-ID: 702aa4e27eb43f5a2118a844545bf896@<EXT SRV IP ADDRESS>:5060
CSeq: 102 INVITE
User-Agent: FPBX-15.0.16.81(16.13.0)
Date: Tue, 08 Dec 2020 06:33:58 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 207793878 207793878 IN IP4 <EXT SRV IP ADDRESS>
s=Asterisk PBX 16.13.0
c=IN IP4 <EXT SRV IP ADDRESS>
t=0 0
m=audio 15484 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

---
[2020-12-08 09:33:58] VERBOSE[5293][C-0000001e] app_dial.c: Called SIP/selectro(7455995)/98985XXXXXXX
[2020-12-08 09:33:58] VERBOSE[2447] chan_sip.c:
<--- 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=z9hG4bK535f9334;received=<EXT SRV IP ADDRESS>;rport=5060
From: <sip:495XXXXXXX@<EXT SRV IP ADDRESS>>;tag=as17d1cdc1
To: <sip:98985XXXXXXX@<PROVIDER SRV IP ADDRESS>>;tag=as5d49aa18
Call-ID: 702aa4e27eb43f5a2118a844545bf896@<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="53732894"
Content-Length: 0

<------------->
[2020-12-08 09:33:58] VERBOSE[2447] chan_sip.c: --- (11 headers 0 lines) ---
[2020-12-08 09:33:58] VERBOSE[2447][C-0000001e] chan_sip.c: Transmitting (NAT) to <PROVIDER SRV IP ADDRESS>:5060:
ACK sip:98985XXXXXXX@<PROVIDER SRV IP ADDRESS> SIP/2.0
Via: SIP/2.0/UDP <EXT SRV IP ADDRESS>:5060;branch=z9hG4bK535f9334;rport
Max-Forwards: 70
From: <sip:495XXXXXXX@<EXT SRV IP ADDRESS>>;tag=as17d1cdc1
To: <sip:98985XXXXXXX@<PROVIDER SRV IP ADDRESS>>;tag=as5d49aa18
Contact: <sip:495XXXXXXX@<EXT SRV IP ADDRESS>:5060>
Call-ID: 702aa4e27eb43f5a2118a844545bf896@<EXT SRV IP ADDRESS>:5060
CSeq: 102 ACK
User-Agent: FPBX-15.0.16.81(16.13.0)
Content-Length: 0


---
[2020-12-08 09:33:58] VERBOSE[2447][C-0000001e] chan_sip.c: Audio is at 15484
[2020-12-08 09:33:58] VERBOSE[2447][C-0000001e] chan_sip.c: Adding codec alaw to SDP
[2020-12-08 09:33:58] VERBOSE[2447][C-0000001e] chan_sip.c: Adding codec ulaw to SDP
[2020-12-08 09:33:58] VERBOSE[2447][C-0000001e] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[2020-12-08 09:33:58] VERBOSE[2447][C-0000001e] chan_sip.c: Reliably Transmitting (NAT) to <PROVIDER SRV IP ADDRESS>:5060:
INVITE sip:98985XXXXXXX@<PROVIDER SRV IP ADDRESS> SIP/2.0
Via: SIP/2.0/UDP <EXT SRV IP ADDRESS>:5060;branch=z9hG4bK48e9091e;rport
Max-Forwards: 70
From: <sip:495XXXXXXX@<EXT SRV IP ADDRESS>>;tag=as17d1cdc1
To: <sip:98985XXXXXXX@<PROVIDER SRV IP ADDRESS>>
Contact: <sip:495XXXXXXX@<EXT SRV IP ADDRESS>:5060>
Call-ID: 702aa4e27eb43f5a2118a844545bf896@<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:98985XXXXXXX@<PROVIDER SRV IP ADDRESS>", nonce="53732894", response="d4274e237c7c36f7ac5302d7ae3b423e"
Date: Tue, 08 Dec 2020 06:33:58 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 207793878 207793879 IN IP4 <EXT SRV IP ADDRESS>
s=Asterisk PBX 16.13.0
c=IN IP4 <EXT SRV IP ADDRESS>
t=0 0
m=audio 15484 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

---
[2020-12-08 09:33:58] VERBOSE[2447] chan_sip.c:
<--- 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=z9hG4bK48e9091e;received=<EXT SRV IP ADDRESS>;rport=5060
From: <sip:495XXXXXXX@<EXT SRV IP ADDRESS>>;tag=as17d1cdc1
To: <sip:98985XXXXXXX@<PROVIDER SRV IP ADDRESS>>
Call-ID: 702aa4e27eb43f5a2118a844545bf896@<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:98985XXXXXXX@<PROVIDER SRV IP ADDRESS>:5060>
Content-Length: 0

<------------->
[2020-12-08 09:33:58] VERBOSE[2447] chan_sip.c: --- (12 headers 0 lines) ---
[2020-12-08 09:33:58] VERBOSE[2447] chan_sip.c: Really destroying SIP dialog '310d52f546d17f096f16654f19c0df10@<PROVIDER SRV IP ADDRESS>:5060' Method: OPTIONS
[2020-12-08 09:33:58] VERBOSE[2447] chan_sip.c: Really destroying SIP dialog '6ebcce420885e37927e8cc0e256d2758@<PROVIDER SRV IP ADDRESS>:5060' Method: NOTIFY
[2020-12-08 09:33:59] VERBOSE[2447] chan_sip.c: Really destroying SIP dialog '7c1884b8491067046405c9645dbc9dce@<PROVIDER SRV IP ADDRESS>:5060' Method: OPTIONS
[2020-12-08 09:33:59] VERBOSE[2447] chan_sip.c: Really destroying SIP dialog '7f90e2207aeac41f6223f20a55b65db0@<PROVIDER SRV IP ADDRESS>:5060' Method: NOTIFY
[2020-12-08 09:33:59] VERBOSE[2447] chan_sip.c:
<--- 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=z9hG4bK48e9091e;received=<EXT SRV IP ADDRESS>;rport=5060
From: <sip:495XXXXXXX@<EXT SRV IP ADDRESS>>;tag=as17d1cdc1
To: <sip:98985XXXXXXX@<PROVIDER SRV IP ADDRESS>>;tag=as7704764d
Call-ID: 702aa4e27eb43f5a2118a844545bf896@<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:98985XXXXXXX@<PROVIDER SRV IP ADDRESS>:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 260

v=0
o=root 163755762 163755762 IN IP4 <PROVIDER SRV IP ADDRESS>
s=Asterisk PBX 11.17.1
c=IN IP4 <PROVIDER SRV IP ADDRESS>
t=0 0
m=audio 18398 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
<------------->
[2020-12-08 09:33:59] VERBOSE[2447] chan_sip.c: --- (14 headers 12 lines) ---
[2020-12-08 09:33:59] VERBOSE[2447][C-0000001e] sip/route.c: sip_route_dump: route/path hop: <sip:98985XXXXXXX@<PROVIDER SRV IP ADDRESS>:5060>
[2020-12-08 09:33:59] VERBOSE[2447][C-0000001e] chan_sip.c: Got SDP version 163755762 and unique parts [root 163755762 IN IP4 <PROVIDER SRV IP ADDRESS>]
[2020-12-08 09:33:59] VERBOSE[2447][C-0000001e] chan_sip.c: Found RTP audio format 8
[2020-12-08 09:33:59] VERBOSE[2447][C-0000001e] chan_sip.c: Found RTP audio format 0
[2020-12-08 09:33:59] VERBOSE[2447][C-0000001e] chan_sip.c: Found RTP audio format 101
[2020-12-08 09:33:59] VERBOSE[2447][C-0000001e] chan_sip.c: Found audio description format PCMA for ID 8
[2020-12-08 09:33:59] VERBOSE[2447][C-0000001e] chan_sip.c: Found audio description format PCMU for ID 0
[2020-12-08 09:33:59] VERBOSE[2447][C-0000001e] chan_sip.c: Found audio description format telephone-event for ID 101
[2020-12-08 09:33:59] VERBOSE[2447][C-0000001e] chan_sip.c: Capabilities: us - (alaw|ulaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (alaw|ulaw)
[2020-12-08 09:33:59] VERBOSE[2447][C-0000001e] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[2020-12-08 09:33:59] VERBOSE[2447][C-0000001e] chan_sip.c: Peer audio RTP is at port <PROVIDER SRV IP ADDRESS>:18398
[2020-12-08 09:33:59] VERBOSE[5293][C-0000001e] app_dial.c: SIP/selectro(7455995)-0000003a is making progress passing it to SIP/sttc-ncp1000-00000039
[2020-12-08 09:33:59] VERBOSE[5293][C-0000001e] chan_sip.c: Audio is at 15714
[2020-12-08 09:33:59] VERBOSE[5293][C-0000001e] chan_sip.c: Adding codec alaw to SDP
[2020-12-08 09:33:59] VERBOSE[5293][C-0000001e] chan_sip.c: Adding codec ulaw to SDP
[2020-12-08 09:33:59] VERBOSE[5293][C-0000001e] chan_sip.c: Adding codec g729 to SDP
[2020-12-08 09:33:59] VERBOSE[5293][C-0000001e] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[2020-12-08 09:33:59] VERBOSE[5293][C-0000001e] chan_sip.c:
<--- 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=z9hG4bK00002e3b;received=10.60.0.2;rport=35060
From: "��������" <sip:101@10.60.0.4>;tag=3544
To: sip:8985XXXXXXX@10.60.0.4;tag=as7f7a04fd
Call-ID: 000043d8-38e3985e3bdd100098ff0080f0c171e8@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:8985XXXXXXX@10.60.0.4:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 317

v=0
o=root 1432565882 1432565882 IN IP4 10.60.0.4
s=Asterisk PBX 16.13.0
c=IN IP4 10.60.0.4
t=0 0
m=audio 15714 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

<------------>
[2020-12-08 09:34:00] VERBOSE[2447] chan_sip.c:
<--- SIP read from UDP:<PROVIDER SRV IP ADDRESS>:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP <EXT SRV IP ADDRESS>:5060;branch=z9hG4bK48e9091e;received=<EXT SRV IP ADDRESS>;rport=5060
From: <sip:495XXXXXXX@<EXT SRV IP ADDRESS>>;tag=as17d1cdc1
To: <sip:98985XXXXXXX@<PROVIDER SRV IP ADDRESS>>;tag=as7704764d
Call-ID: 702aa4e27eb43f5a2118a844545bf896@<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:98985XXXXXXX@<PROVIDER SRV IP ADDRESS>:5060>
Content-Length: 0
Simmer
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Re: Не проходят международные звонки

Сообщение SergA » 08 дек 2020, 15:06

Нет разницы - Panasonic оправдан???
1. Вас не смущает в логах Scheduling destruction?
2. Пожалуйста, не занимайтесь самобичеванием, смотрите и показывайте трассы как .pcap
3. Оператор связи, к которому подключены по SIP как комментирует ситуацию?
SergA
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Re: Не проходят международные звонки

Сообщение Simmer » 08 дек 2020, 18:05

Думаю да... Реабилитирован...)))
Вас не смущает в логах Scheduling destruction?

Пока Вы не сказали не смущало. Теперь займусь. Уже почитал... понял куда копать.
Пожалуйста, не занимайтесь самобичеванием, смотрите и показывайте трассы как .pcap

Понял, буду привыкать)))
Оператор связи, к которому подключены по SIP как комментирует ситуацию?

Сегодня под нажимом и после того, как завалил логами с конкретным указанием мест, где происходит ошибка сдался и после анализа на своей стороне признался, что была не верно настроена маршрутизация. Но я в этой организации удаленно и проверить смогу в лучшем случае в выходные если с больничного выпустят...
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Re: Не проходят международные звонки

Сообщение SergA » 08 дек 2020, 21:16

Опер, который признал свою ошибку, а потом нашел её источник - уже хорошо...если исправит - точно красавчик! )))
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