Настройка асинхронного DialPlan

Onotoliy · Сообщение **Onotoliy** » Вчера, 12:09

Задача. При вызове номера 400365 необходимо асинхронно набрать два номера 100029 и 100030

Написал DialPlan на Lua. Который при звонке на номер 400365 отправляет два асинхронных вызова, но при ответе на вызов происходит сброс звонка.

Из-за чего может происходить сброс звонка?

Ниже приведены настройки DialPlan и логи Asterisk.

DialPlan.

Скрипт "/usr/share/asterisk/agi-bin/gate-get-accounts.py" получает список номеров куда звонить (100029 и 100030) в формате PJSIP/100029&PJSIP/100030

function new_fun_mycompany_gate(context, exten, num)
app.Log("NOTICE", "NEW Gate. Received call to " .. num .. " at " .. os.date("%Y-%m-%d %H:%M:%S"))
app.Wait(1)

if num and num ~= "" then
app.Log("NOTICE", "Originate " .. num)
app.Originate(num, "exten", "mycompany_internal", 1, 15)
end
end

function fun_mycompany_gate(context, exten)
app.Log("NOTICE", "Gate. Received call to " .. exten .. " at " .. os.date("%Y-%m-%d %H:%M:%S"))
app.AGI("/usr/share/asterisk/agi-bin/gate-get-accounts.py")

local targets = channel["DIAL_TARGETS"]:get()

app.Log("NOTICE", "Dial DIAL_TARGETS: " .. (targets or ""))

if targets and targets ~= "" then
for num in string.gmatch(targets, "([^&]+)") do
num = num:match("^%s*(.-)%s*$")
if num ~= "" then
local success, err_message = pcall(function()
new_fun_mycompany_gate(context, exten, num)
end)

if not success then
channel.Hangup()
app.NoOp("An error occurred: " .. tostring(err_message)) -- Log the error
else
app.NoOp("Function executed without errors.")
end
end
end
end
end

function fun_mycompany_intercom(context, exten)
app.Log("NOTICE", "Intercom. Received call to " .. exten .. " at " .. os.date("%Y-%m-%d %H:%M:%S"))
app.Wait(1)
app.Dial("PJSIP/" .. exten, 15)
app.Hangup()
end

function fun_mycompany_internal(context, exten)
app.Log("NOTICE", "Hello fun_mycompany_internal")
end

extensions = {
["mycompany_internal"] = {
["_X"] = fun_mycompany_internal;
["_XX"] = fun_mycompany_internal;
["_XXX"] = fun_mycompany_internal;
["_XXXX"] = fun_mycompany_internal;
["_XXXXX"] = fun_mycompany_internal;
["_XXXXXX"] = fun_mycompany_internal;
};

["mycompany_gate"] = {
["_X"] = fun_mycompany_gate;
};

["mycompany_intercom"] = {
["_X"] = fun_mycompany_intercom;
};

["mycompany"] = {
["_[456789]XXXX"] = fun_mycompany_gate;
["_[456789]XXXXX"] = fun_mycompany_gate;
["_[456789]XXXXXX"] = fun_mycompany_gate;
["_[456789]XXXXXXX"] = fun_mycompany_gate;
["_[123]XXXXX"] = fun_mycompany_intercom;
};
}

hints = {
}

Логи

u-um-test*CLI> pjsip set logger on
PJSIP Logging enabled
<--- Received SIP request (428 bytes) from TCP:176.123.206.47:14659 --->
BYE sip:176.123.192.56:5060;transport=TCP SIP/2.0
Via: SIP/2.0/TCP 176.123.206.47:14659;rport;branch=z9hG4bKPjEiuottzB72oIh9AAUpEs.jVwX9uocwKB;alias
Max-Forwards: 70
From: "100027" <sip:100027@176.123.192.56>;tag=jYKXMoQbkMFqcg1kmlYdVdthAgCzVe4t
To: sip:400365@176.123.192.56;tag=0810abd0-7a08-4d7d-81c8-6d23e2c49061
Call-ID: wyvYSLKDH73NsVKNj4aj3jAoAUtV81Hl
CSeq: 26543 BYE
User-Agent: Beward SIP
Content-Length: 0

<--- Transmitting SIP response (412 bytes) to TCP:176.123.206.47:14659 --->
SIP/2.0 200 OK
Via: SIP/2.0/TCP 176.123.206.47:14659;rport=14659;received=176.123.206.47;branch=z9hG4bKPjEiuottzB72oIh9AAUpEs.jVwX9uocwKB;alias
Call-ID: wyvYSLKDH73NsVKNj4aj3jAoAUtV81Hl
From: "100027" <sip:100027@176.123.192.56>;tag=jYKXMoQbkMFqcg1kmlYdVdthAgCzVe4t
To: <sip:400365@176.123.192.56>;tag=0810abd0-7a08-4d7d-81c8-6d23e2c49061
CSeq: 26543 BYE
Server: Asterisk PBX 22.7.0
Content-Length: 0

<--- Received SIP request (1099 bytes) from TCP:176.123.206.47:14659 --->
INVITE sip:400365@176.123.192.56 SIP/2.0
Via: SIP/2.0/TCP 176.123.206.47:14659;rport;branch=z9hG4bKPj6jI12Td-W8qc7I8dY3dl1BMSln1rO-mT;alias
Max-Forwards: 70
From: "100027" <sip:100027@176.123.192.56>;tag=YgkfQPB8slF5REqSOIwT9lnZNUESRDyM
To: sip:400365@176.123.192.56
Contact: "100027" <sip:100027@10.156.10.98:5060;ob>
Call-ID: cIPiDhmhu.QgXfxheBwHhxLGXdcqv6h9
CSeq: 30723 INVITE
Allow: INVITE, ACK, BYE, CANCEL, UPDATE, INFO, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: Beward SIP
X-PID: 2323
Content-Type: application/sdp
Content-Length: 479

v=0
o=- 3980343019 3980343019 IN IP4 176.123.206.47
s=Door station call
b=AS:458
t=0 0
a=X-nat:0
m=audio 4248 RTP/AVP 8 101
c=IN IP4 176.123.206.47
b=TIAS:64000
a=rtcp:4249 IN IP4 10.156.10.98
a=sendrecv
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
m=video 4250 RTP/AVP 96
c=IN IP4 176.123.206.47
b=TIAS:356352
a=rtcp:4251 IN IP4 10.156.10.98
a=sendonly
a=rtpmap:96 H264/90000
a=fmtp:96 profile-level-id=4d001f;packetization-mode=1

<--- Transmitting SIP response (576 bytes) to TCP:176.123.206.47:14659 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/TCP 176.123.206.47:14659;rport=14659;received=176.123.206.47;branch=z9hG4bKPj6jI12Td-W8qc7I8dY3dl1BMSln1rO-mT;alias
Call-ID: cIPiDhmhu.QgXfxheBwHhxLGXdcqv6h9
From: "100027" <sip:100027@176.123.192.56>;tag=YgkfQPB8slF5REqSOIwT9lnZNUESRDyM
To: <sip:400365@176.123.192.56>;tag=z9hG4bKPj6jI12Td-W8qc7I8dY3dl1BMSln1rO-mT
CSeq: 30723 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1771318958/c7ba0ebc71d046cd5c72816c7a837cd4",opaque="691546ee5db60a1b",algorithm=MD5,qop="auth"
Server: Asterisk PBX 22.7.0
Content-Length: 0

<--- Received SIP request (397 bytes) from TCP:176.123.206.47:14659 --->
ACK sip:400365@176.123.192.56 SIP/2.0
Via: SIP/2.0/TCP 176.123.206.47:14659;rport;branch=z9hG4bKPj6jI12Td-W8qc7I8dY3dl1BMSln1rO-mT;alias
Max-Forwards: 70
From: "100027" <sip:100027@176.123.192.56>;tag=YgkfQPB8slF5REqSOIwT9lnZNUESRDyM
To: sip:400365@176.123.192.56;tag=z9hG4bKPj6jI12Td-W8qc7I8dY3dl1BMSln1rO-mT
Call-ID: cIPiDhmhu.QgXfxheBwHhxLGXdcqv6h9
CSeq: 30723 ACK
Content-Length: 0

<--- Received SIP request (1397 bytes) from TCP:176.123.206.47:14659 --->
INVITE sip:400365@176.123.192.56 SIP/2.0
Via: SIP/2.0/TCP 176.123.206.47:14659;rport;branch=z9hG4bKPji2BUJrJ82BgoBZX6crqw7d0H35o3xa6u;alias
Max-Forwards: 70
From: "100027" <sip:100027@176.123.192.56>;tag=YgkfQPB8slF5REqSOIwT9lnZNUESRDyM
To: sip:400365@176.123.192.56
Contact: "100027" <sip:100027@10.156.10.98:5060;ob>
Call-ID: cIPiDhmhu.QgXfxheBwHhxLGXdcqv6h9
CSeq: 30724 INVITE
Allow: INVITE, ACK, BYE, CANCEL, UPDATE, INFO, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: Beward SIP
X-PID: 2323
Authorization: Digest username="100027", realm="asterisk", nonce="1771318958/c7ba0ebc71d046cd5c72816c7a837cd4", uri="sip:400365@176.123.192.56", response="bcccd89563d83ccca9a797bc02f8a0ac", algorithm=MD5, cnonce="-j3p1WxlYKkQ5a0xCq0PZ.Xj-SPw.Ak2", opaque="691546ee5db60a1b", qop=auth, nc=00000001
Content-Type: application/sdp
Content-Length: 479

v=0
o=- 3980343019 3980343019 IN IP4 176.123.206.47
s=Door station call
b=AS:458
t=0 0
a=X-nat:0
m=audio 4248 RTP/AVP 8 101
c=IN IP4 176.123.206.47
b=TIAS:64000
a=rtcp:4249 IN IP4 10.156.10.98
a=sendrecv
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
m=video 4250 RTP/AVP 96
c=IN IP4 176.123.206.47
b=TIAS:356352
a=rtcp:4251 IN IP4 10.156.10.98
a=sendonly
a=rtpmap:96 H264/90000
a=fmtp:96 profile-level-id=4d001f;packetization-mode=1

<--- Transmitting SIP response (378 bytes) to TCP:176.123.206.47:14659 --->
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 176.123.206.47:14659;rport=14659;received=176.123.206.47;branch=z9hG4bKPji2BUJrJ82BgoBZX6crqw7d0H35o3xa6u;alias
Call-ID: cIPiDhmhu.QgXfxheBwHhxLGXdcqv6h9
From: "100027" <sip:100027@176.123.192.56>;tag=YgkfQPB8slF5REqSOIwT9lnZNUESRDyM
To: <sip:400365@176.123.192.56>
CSeq: 30724 INVITE
Server: Asterisk PBX 22.7.0
Content-Length: 0

[2026-02-17 09:02:39] NOTICE[1423871][C-000000e8]: Ext. 400365:1 @ umkey: Gate. Received call to 400365 at 2026-02-17 09:02:39
[2026-02-17 09:02:39] NOTICE[1423871][C-000000e8]: Ext. 400365:1 @ umkey: Dial DIAL_TARGETS: PJSIP/100029&PJSIP/100030&PJSIP/65@100025
[2026-02-17 09:02:39] NOTICE[1423871][C-000000e8]: Ext. 400365:1 @ umkey: NEW Gate. Received call to PJSIP/100029 at 2026-02-17 09:02:39
[2026-02-17 09:02:50] NOTICE[1423871][C-000000e8]: Ext. 400365:1 @ umkey: Originate PJSIP/100029
[2026-02-17 09:02:50] ERROR[1423823]: res_pjsip.c:993 ast_sip_create_dialog_uac: Endpoint '100029': Could not create dialog to invalid URI '100029'. Is endpoint registered and reachable?
[2026-02-17 09:02:50] ERROR[1423823]: chan_pjsip.c:2726 request: Failed to create outgoing session to endpoint '100029'
[2026-02-17 09:02:50] NOTICE[1423871][C-000000e8]: Ext. 400365:1 @ umkey: Function executed without errors.
[2026-02-17 09:02:50] NOTICE[1423871][C-000000e8]: Ext. 400365:1 @ umkey: NEW Gate. Received call to PJSIP/100030 at 2026-02-17 09:02:50
<--- Received SIP request (771 bytes) from TCP:185.56.117.165:48069 --->
REGISTER sip:test.umkey.ru:5060 SIP/2.0
Via: SIP/2.0/TCP 10.94.0.93:47774;alias;branch=z9hG4bK.GvYDO7aFU;rport
From: <sip:100030@test.umkey.ru>;tag=NCFpbh10w
To: sip:100030@test.umkey.ru
CSeq: 20 REGISTER
Call-ID: PoyzLbrToD
Max-Forwards: 70
Supported: replaces, outbound, gruu, path
Accept: application/sdp
Accept: text/plain
Accept: application/vnd.gsma.rcs-ft-http+xml
Contact: <sip:100030@10.94.0.93:47774;pn-prid=e6zMRGW2SkKEwyjLqEoouW:APA91bHtK0fn5Iy5LsgbzcVZH7TXdiZqxn6Ga1wsuHKJdHHWuXtcjDw0oVFpeUAJ4ntNOQgfH2sJrzz78Y9iUexrBzv4ppDgKp2blGvaz2AVKoMGJHKA-RU;pn-provider=fcm;pn-param=10837285283;pn-silent=1;pn-timeout=0;transport=tcp>;+sip.instance="<urn:uuid:8889c654-e337-4cff-9720-75e0e08ac914>"
Expires: 3600
User-Agent: Unknown
Content-Length: 0

<--- Transmitting SIP response (467 bytes) to TCP:185.56.117.165:48069 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/TCP 10.94.0.93:47774;rport=48069;received=185.56.117.165;branch=z9hG4bK.GvYDO7aFU;alias
Call-ID: PoyzLbrToD
From: <sip:100030@test.umkey.ru>;tag=NCFpbh10w
To: <sip:100030@test.umkey.ru>;tag=z9hG4bK.GvYDO7aFU
CSeq: 20 REGISTER
WWW-Authenticate: Digest realm="asterisk",nonce="1771318972/cf4b3e70edcce2504efd2cac520d3181",opaque="569dbd2a25c35da2",algorithm=MD5,qop="auth"
Server: Asterisk PBX 22.7.0
Content-Length: 0

<--- Received SIP request (1059 bytes) from TCP:185.56.117.165:48069 --->
REGISTER sip:test.umkey.ru:5060 SIP/2.0
Via: SIP/2.0/TCP 192.168.10.103:46404;alias;branch=z9hG4bK.6BRBb-STa;rport
From: <sip:100030@test.umkey.ru>;tag=NCFpbh10w
To: sip:100030@test.umkey.ru
CSeq: 21 REGISTER
Call-ID: PoyzLbrToD
Max-Forwards: 70
Supported: replaces, outbound, gruu, path
Accept: application/sdp
Accept: text/plain
Accept: application/vnd.gsma.rcs-ft-http+xml
Contact: <sip:100030@185.56.117.165:48069;pn-prid=e6zMRGW2SkKEwyjLqEoouW:APA91bHtK0fn5Iy5LsgbzcVZH7TXdiZqxn6Ga1wsuHKJdHHWuXtcjDw0oVFpeUAJ4ntNOQgfH2sJrzz78Y9iUexrBzv4ppDgKp2blGvaz2AVKoMGJHKA-RU;pn-provider=fcm;pn-param=10837285283;pn-silent=1;pn-timeout=0;transport=tcp>;+sip.instance="<urn:uuid:8889c654-e337-4cff-9720-75e0e08ac914>"
Expires: 3600
User-Agent: Unknown
Content-Length: 0
Authorization: Digest realm="asterisk", nonce="1771318972/cf4b3e70edcce2504efd2cac520d3181", algorithm=MD5, opaque="569dbd2a25c35da2", username="100030", uri="sip:test.umkey.ru:5060", response="498c88ebf077b6bc742d7bfe8eea2303", cnonce="yiPoPlxloFdvikNW", nc=00000001, qop=auth

<--- Transmitting SIP response (651 bytes) to TCP:185.56.117.165:48069 --->
SIP/2.0 200 OK
Via: SIP/2.0/TCP 192.168.10.103:46404;rport=48069;received=185.56.117.165;branch=z9hG4bK.6BRBb-STa;alias
Call-ID: PoyzLbrToD
From: <sip:100030@test.umkey.ru>;tag=NCFpbh10w
To: <sip:100030@test.umkey.ru>;tag=z9hG4bK.6BRBb-STa
CSeq: 21 REGISTER
Date: Tue, 17 Feb 2026 09:02:53 GMT
Contact: <sip:100030@185.56.117.165:48069;transport=TCP;pn-prid=e6zMRGW2SkKEwyjLqEoouW:APA91bHtK0fn5Iy5LsgbzcVZH7TXdiZqxn6Ga1wsuHKJdHHWuXtcjDw0oVFpeUAJ4ntNOQgfH2sJrzz78Y9iUexrBzv4ppDgKp2blGvaz2AVKoMGJHKA-RU;pn-provider=fcm;pn-param=10837285283;pn-silent=1;pn-timeout=0>;expires=3599
Expires: 3600
Server: Asterisk PBX 22.7.0
Content-Length: 0

[2026-02-17 09:02:53] NOTICE[1423871][C-000000e8]: Ext. 400365:1 @ umkey: Originate PJSIP/100030
<--- Transmitting SIP request (1422 bytes) to TCP:185.56.117.165:48069 --->
INVITE sip:100030@185.56.117.165:48069;transport=TCP;pn-prid=e6zMRGW2SkKEwyjLqEoouW:APA91bHtK0fn5Iy5LsgbzcVZH7TXdiZqxn6Ga1wsuHKJdHHWuXtcjDw0oVFpeUAJ4ntNOQgfH2sJrzz78Y9iUexrBzv4ppDgKp2blGvaz2AVKoMGJHKA-RU;pn-provider=fcm;pn-param=10837285283;pn-silent=1;pn-timeout=0 SIP/2.0
Via: SIP/2.0/TCP 176.123.192.56:5060;rport;branch=z9hG4bKPj18e8885b-c81c-4a40-a9fe-5c80eb51f476;alias
From: "Anonymous" <sip:anonymous@test.umkey.ru>;tag=bcacf8b0-0e32-481e-875d-9c109e92f2e2
To: <sip:100030@185.56.117.165;pn-prid=e6zMRGW2SkKEwyjLqEoouW:APA91bHtK0fn5Iy5LsgbzcVZH7TXdiZqxn6Ga1wsuHKJdHHWuXtcjDw0oVFpeUAJ4ntNOQgfH2sJrzz78Y9iUexrBzv4ppDgKp2blGvaz2AVKoMGJHKA-RU;pn-provider=fcm;pn-param=10837285283;pn-silent=1;pn-timeout=0>
Contact: <sip:asterisk@176.123.192.56:5060;transport=TCP>
Call-ID: 24fd8f99-682c-458d-8773-8e096007218e
CSeq: 20064 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, INFO, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 22.7.0
Content-Type: application/sdp
Content-Length: 265

v=0
o=- 1931593297 1931593297 IN IP4 176.123.192.56
s=Asterisk
c=IN IP4 176.123.192.56
t=0 0
m=audio 10476 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:140
a=sendrecv

<--- Received SIP response (547 bytes) from TCP:185.56.117.165:48069 --->
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 176.123.192.56:5060;rport;branch=z9hG4bKPj18e8885b-c81c-4a40-a9fe-5c80eb51f476;alias
From: "Anonymous" <sip:anonymous@test.umkey.ru>;tag=bcacf8b0-0e32-481e-875d-9c109e92f2e2
To: <sip:100030@185.56.117.165;pn-prid=e6zMRGW2SkKEwyjLqEoouW:APA91bHtK0fn5Iy5LsgbzcVZH7TXdiZqxn6Ga1wsuHKJdHHWuXtcjDw0oVFpeUAJ4ntNOQgfH2sJrzz78Y9iUexrBzv4ppDgKp2blGvaz2AVKoMGJHKA-RU;pn-provider=fcm;pn-param=10837285283;pn-silent=1;pn-timeout=0>
Call-ID: 24fd8f99-682c-458d-8773-8e096007218e
CSeq: 20064 INVITE
Content-Length: 0

<--- Received SIP response (624 bytes) from TCP:185.56.117.165:48069 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/TCP 176.123.192.56:5060;rport;branch=z9hG4bKPj18e8885b-c81c-4a40-a9fe-5c80eb51f476;alias
From: "Anonymous" <sip:anonymous@test.umkey.ru>;tag=bcacf8b0-0e32-481e-875d-9c109e92f2e2
To: <sip:100030@185.56.117.165;pn-prid=e6zMRGW2SkKEwyjLqEoouW:APA91bHtK0fn5Iy5LsgbzcVZH7TXdiZqxn6Ga1wsuHKJdHHWuXtcjDw0oVFpeUAJ4ntNOQgfH2sJrzz78Y9iUexrBzv4ppDgKp2blGvaz2AVKoMGJHKA-RU;pn-provider=fcm;pn-param=10837285283;pn-silent=1;pn-timeout=0>;tag=p2WiJe-
Call-ID: 24fd8f99-682c-458d-8773-8e096007218e
CSeq: 20064 INVITE
User-Agent: Unknown
Supported: replaces, outbound, gruu, path
Content-Length: 0

<--- Received SIP response (1201 bytes) from TCP:185.56.117.165:48069 --->
SIP/2.0 200 Ok
Via: SIP/2.0/TCP 176.123.192.56:5060;rport;branch=z9hG4bKPj18e8885b-c81c-4a40-a9fe-5c80eb51f476;alias
From: "Anonymous" <sip:anonymous@test.umkey.ru>;tag=bcacf8b0-0e32-481e-875d-9c109e92f2e2
To: <sip:100030@185.56.117.165;pn-prid=e6zMRGW2SkKEwyjLqEoouW:APA91bHtK0fn5Iy5LsgbzcVZH7TXdiZqxn6Ga1wsuHKJdHHWuXtcjDw0oVFpeUAJ4ntNOQgfH2sJrzz78Y9iUexrBzv4ppDgKp2blGvaz2AVKoMGJHKA-RU;pn-provider=fcm;pn-param=10837285283;pn-silent=1;pn-timeout=0>;tag=p2WiJe-
Call-ID: 24fd8f99-682c-458d-8773-8e096007218e
CSeq: 20064 INVITE
User-Agent: Unknown
Supported: replaces, outbound, gruu, path
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, PRACK, UPDATE
Contact: <sip:100030@185.56.117.165:48069;pn-prid=e6zMRGW2SkKEwyjLqEoouW:APA91bHtK0fn5Iy5LsgbzcVZH7TXdiZqxn6Ga1wsuHKJdHHWuXtcjDw0oVFpeUAJ4ntNOQgfH2sJrzz78Y9iUexrBzv4ppDgKp2blGvaz2AVKoMGJHKA-RU;pn-provider=fcm;pn-param=10837285283;pn-silent=1;pn-timeout=0;transport=tcp>;expires=3599
Content-Type: application/sdp
Content-Length: 167

v=0
o=100030 4049 1930 IN IP4 192.168.10.103
s=Talk
c=IN IP4 192.168.10.103
t=0 0
m=audio 43696 RTP/AVP 8 0 101
a=rtpmap:101 telephone-event/8000
a=rtcp:39245

<--- Transmitting SIP request (860 bytes) to TCP:185.56.117.165:48069 --->
ACK sip:100030@185.56.117.165:48069;transport=tcp;pn-prid=e6zMRGW2SkKEwyjLqEoouW:APA91bHtK0fn5Iy5LsgbzcVZH7TXdiZqxn6Ga1wsuHKJdHHWuXtcjDw0oVFpeUAJ4ntNOQgfH2sJrzz78Y9iUexrBzv4ppDgKp2blGvaz2AVKoMGJHKA-RU;pn-provider=fcm;pn-param=10837285283;pn-silent=1;pn-timeout=0 SIP/2.0
Via: SIP/2.0/TCP 176.123.192.56:5060;rport;branch=z9hG4bKPj1ad64157-a298-4c79-894d-2729b29131ce;alias
From: "Anonymous" <sip:anonymous@test.umkey.ru>;tag=bcacf8b0-0e32-481e-875d-9c109e92f2e2
To: <sip:100030@185.56.117.165;pn-prid=e6zMRGW2SkKEwyjLqEoouW:APA91bHtK0fn5Iy5LsgbzcVZH7TXdiZqxn6Ga1wsuHKJdHHWuXtcjDw0oVFpeUAJ4ntNOQgfH2sJrzz78Y9iUexrBzv4ppDgKp2blGvaz2AVKoMGJHKA-RU;pn-provider=fcm;pn-param=10837285283;pn-silent=1;pn-timeout=0>;tag=p2WiJe-
Call-ID: 24fd8f99-682c-458d-8773-8e096007218e
CSeq: 20064 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 22.7.0
Content-Length: 0

[2026-02-17 09:02:56] NOTICE[1423871][C-000000e8]: Ext. 400365:1 @ umkey: Function executed without errors.
[2026-02-17 09:02:56] NOTICE[1423871][C-000000e8]: Ext. 400365:1 @ umkey: NEW Gate. Received call to PJSIP/65@100025 at 2026-02-17 09:02:56
<--- Transmitting SIP request (883 bytes) to TCP:185.56.117.165:48069 --->
BYE sip:100030@185.56.117.165:48069;transport=tcp;pn-prid=e6zMRGW2SkKEwyjLqEoouW:APA91bHtK0fn5Iy5LsgbzcVZH7TXdiZqxn6Ga1wsuHKJdHHWuXtcjDw0oVFpeUAJ4ntNOQgfH2sJrzz78Y9iUexrBzv4ppDgKp2blGvaz2AVKoMGJHKA-RU;pn-provider=fcm;pn-param=10837285283;pn-silent=1;pn-timeout=0 SIP/2.0
Via: SIP/2.0/TCP 176.123.192.56:5060;rport;branch=z9hG4bKPj47421fa3-6fe2-4ea8-a61a-6bd4dbbfd0bf;alias
From: "Anonymous" <sip:anonymous@test.umkey.ru>;tag=bcacf8b0-0e32-481e-875d-9c109e92f2e2
To: <sip:100030@185.56.117.165;pn-prid=e6zMRGW2SkKEwyjLqEoouW:APA91bHtK0fn5Iy5LsgbzcVZH7TXdiZqxn6Ga1wsuHKJdHHWuXtcjDw0oVFpeUAJ4ntNOQgfH2sJrzz78Y9iUexrBzv4ppDgKp2blGvaz2AVKoMGJHKA-RU;pn-provider=fcm;pn-param=10837285283;pn-silent=1;pn-timeout=0>;tag=p2WiJe-
Call-ID: 24fd8f99-682c-458d-8773-8e096007218e
CSeq: 20065 BYE
Reason: Q.850;cause=0
Max-Forwards: 70
User-Agent: Asterisk PBX 22.7.0
Content-Length: 0

<--- Received SIP response (616 bytes) from TCP:185.56.117.165:48069 --->
SIP/2.0 200 Ok
Via: SIP/2.0/TCP 176.123.192.56:5060;rport;branch=z9hG4bKPj47421fa3-6fe2-4ea8-a61a-6bd4dbbfd0bf;alias
From: "Anonymous" <sip:anonymous@test.umkey.ru>;tag=bcacf8b0-0e32-481e-875d-9c109e92f2e2
To: <sip:100030@185.56.117.165;pn-prid=e6zMRGW2SkKEwyjLqEoouW:APA91bHtK0fn5Iy5LsgbzcVZH7TXdiZqxn6Ga1wsuHKJdHHWuXtcjDw0oVFpeUAJ4ntNOQgfH2sJrzz78Y9iUexrBzv4ppDgKp2blGvaz2AVKoMGJHKA-RU;pn-provider=fcm;pn-param=10837285283;pn-silent=1;pn-timeout=0>;tag=p2WiJe-
Call-ID: 24fd8f99-682c-458d-8773-8e096007218e
CSeq: 20065 BYE
User-Agent: Unknown
Supported: replaces, outbound, gruu, path
Content-Length: 0

[2026-02-17 09:02:58] NOTICE[1423871][C-000000e8]: Ext. 400365:1 @ umkey: Originate PJSIP/65@100025
<--- Transmitting SIP request (994 bytes) to TCP:176.123.206.47:15933 --->
INVITE sip:65@176.123.206.47:15933;transport=TCP;ob SIP/2.0
Via: SIP/2.0/TCP 176.123.192.56:5060;rport;branch=z9hG4bKPj2ddf6b44-5048-4c88-ac91-8de7e78604f9;alias
From: "Anonymous" <sip:anonymous@test.umkey.ru>;tag=ea906625-a7ca-4fb5-a9cf-c8655d31cbd6
To: <sip:65@176.123.206.47;ob>
Contact: <sip:asterisk@176.123.192.56:5060;transport=TCP>
Call-ID: 779639d9-0405-4188-9d72-9b411b74a428
CSeq: 21794 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, INFO, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 22.7.0
Content-Type: application/sdp
Content-Length: 265

v=0
o=- 1732581966 1732581966 IN IP4 176.123.192.56
s=Asterisk
c=IN IP4 176.123.192.56
t=0 0
m=audio 10610 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:140
a=sendrecv

<--- Received SIP response (363 bytes) from TCP:176.123.206.47:15933 --->
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 176.123.192.56:5060;rport=5060;received=176.123.192.56;branch=z9hG4bKPj2ddf6b44-5048-4c88-ac91-8de7e78604f9;alias
Call-ID: 779639d9-0405-4188-9d72-9b411b74a428
From: "Anonymous" <sip:anonymous@test.umkey.ru>;tag=ea906625-a7ca-4fb5-a9cf-c8655d31cbd6
To: <sip:65@176.123.206.47;ob>
CSeq: 21794 INVITE
Content-Length: 0

<--- Received SIP response (398 bytes) from TCP:176.123.206.47:15933 --->
SIP/2.0 503 Busy
Via: SIP/2.0/TCP 176.123.192.56:5060;rport=5060;received=176.123.192.56;branch=z9hG4bKPj2ddf6b44-5048-4c88-ac91-8de7e78604f9;alias
Call-ID: 779639d9-0405-4188-9d72-9b411b74a428
From: "Anonymous" <sip:anonymous@test.umkey.ru>;tag=ea906625-a7ca-4fb5-a9cf-c8655d31cbd6
To: <sip:65@176.123.206.47;ob>;tag=rax218s.K1CwQ6ITd4oLzcQ8VFx-LVog
CSeq: 21794 INVITE
Content-Length: 0

<--- Transmitting SIP request (457 bytes) to TCP:176.123.206.47:15933 --->
ACK sip:65@176.123.206.47:15933;transport=TCP;ob SIP/2.0
Via: SIP/2.0/TCP 176.123.192.56:5060;rport;branch=z9hG4bKPj2ddf6b44-5048-4c88-ac91-8de7e78604f9;alias
From: "Anonymous" <sip:anonymous@test.umkey.ru>;tag=ea906625-a7ca-4fb5-a9cf-c8655d31cbd6
To: <sip:65@176.123.206.47;ob>;tag=rax218s.K1CwQ6ITd4oLzcQ8VFx-LVog
Call-ID: 779639d9-0405-4188-9d72-9b411b74a428
CSeq: 21794 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 22.7.0
Content-Length: 0

[2026-02-17 09:02:58] NOTICE[1423871][C-000000e8]: Ext. 400365:1 @ umkey: Function executed without errors.
<--- Transmitting SIP response (980 bytes) to TCP:176.123.206.47:14659 --->
SIP/2.0 200 OK
Via: SIP/2.0/TCP 176.123.206.47:14659;rport=14659;received=176.123.206.47;branch=z9hG4bKPji2BUJrJ82BgoBZX6crqw7d0H35o3xa6u;alias
Call-ID: cIPiDhmhu.QgXfxheBwHhxLGXdcqv6h9
From: "100027" <sip:100027@176.123.192.56>;tag=YgkfQPB8slF5REqSOIwT9lnZNUESRDyM
To: <sip:400365@176.123.192.56>;tag=f671066b-21fe-4a15-a5c9-3d75e24de775
CSeq: 30724 INVITE
Server: Asterisk PBX 22.7.0
Contact: <sip:176.123.192.56:5060;transport=TCP>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, INFO, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800;refresher=uac
Require: timer
Content-Type: application/sdp
Content-Length: 263

v=0
o=- 3980343019 3980343021 IN IP4 176.123.192.56
s=Asterisk
c=IN IP4 176.123.192.56
t=0 0
m=audio 10088 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:140
a=sendrecv
m=video 0 RTP/AVP 96

<--- Received SIP request (404 bytes) from TCP:176.123.206.47:14659 --->
ACK sip:176.123.192.56:5060;transport=TCP SIP/2.0
Via: SIP/2.0/TCP 176.123.206.47:14659;rport;branch=z9hG4bKPjRTDLOMqUhWnpqTa-NeEqh32Q4Uhu3.Tu;alias
Max-Forwards: 70
From: "100027" <sip:100027@176.123.192.56>;tag=YgkfQPB8slF5REqSOIwT9lnZNUESRDyM
To: sip:400365@176.123.192.56;tag=f671066b-21fe-4a15-a5c9-3d75e24de775
Call-ID: cIPiDhmhu.QgXfxheBwHhxLGXdcqv6h9
CSeq: 30724 ACK
Content-Length: 0

Glukinho · Сообщение **Glukinho** » Вчера, 12:22

Что такое "два асинхронных звонка"? Просто набрать два пира, кто первый ответил?

Без Lua и доп. скриптов, через обычный диалплан Dial(PJSIP/100029&PJSIP/100030) - звонится нормально? Если нет, то что при этом написано в консоли при core set verbose 5, core set debug 5?

Onotoliy · Сообщение **Onotoliy** » Вчера, 12:40

Попробую объяснить.
У нас есть мобильное приложение, в которое приходит звонок. Это приложение просыпается (регистрируется в Asterisk) по пушу которое отсылается через AGI скрипт.
Если использовать Dial(PJSIP/100029&PJSIP/100030), то звонок приходит тому пользователю кто первым зарегистрировался.
То есть получаются гонки на регистрацию пользователя, а нужно чтобы были гонки на то кто первым трубку возьмет...

Glukinho · Сообщение **Glukinho** » Вчера, 12:57

А какая разница? Если есть проблема, максимально ее локализуйте, будет хотя бы понятно, где чинить.

PJSIP, кажется, умеет звонить одновременно по всем зарегистрированным контактам данного пира, не смотрели в эту сторону?

Onotoliy · Сообщение **Onotoliy** » Вчера, 13:23

Вот как раз суть в "зарегистрированным контактам". Но пользователи изначально НЕ зарегистрированы в Asterisk и не знают будут ли им вообще звонить.

Ожидаемое поведение по временной шкале:
1. Время 13:00:00. Делаем вызов на номер 400365. Пользователи НЕ зарегистрированы в Asterisk.
2. Время 13:00:00. Отправляем пуш пользователю 100029
3. Время 13:00:00. Отправляем пуш пользователю 100030
4. Время 13:00:01. Пользователь 100029 получает пуш, после чего регистрируется в Asterisk и получает звонок. НО ТРУБКУ не берет
5. Время 13:00:02. Пользователь 100030 получает пуш, после чего регистрируется в Asterisk и получает звонок. Отвечает на звонок и общается.

Поведение по временной шкале если использовать Dial(PJSIP/100029&PJSIP/100030):
1. Время 13:00:00. Делаем вызов на номер 400365. Пользователи НЕ зарегистрированы в Asterisk.
2. Время 13:00:00. Отправляем пуш пользователю 100029.
3. Время 13:00:00. Отправляем пуш пользователю 100030.
4. Время 13:00:01. Пользователь 100029 получает пуш, после чего регистрируется в Asterisk и получает звонок. НО ТРУБКУ не берет.
5. Время 13:00:02. Пользователь 100030 получает пуш, после чего регистрируется в Asterisk и звонка нет.

В данном сценарии проблема в том что 100029 не ждет пока 100030 зарегистрируется...
А ждать пока все зарегистрируются бессмысленно, т.к. у 100030 связи может и не быть.

Я написал DialPlan на Lua который выше. Он соответствует поведению которое бизнес ожидает. Но есть проблема при ответе звонка одним из абонентов, звонок сразу сбрасывается.

Glukinho · Сообщение **Glukinho** » Вчера, 13:44

Интересная конструкция со звонками тем, кого еще нет на АТС. Не знаю, должно ли это вообще работать...

В любом случае, сделайте core set verbose 5, core set debug 5 и читайте консоль в момент разрыва звонка. Что-нибудь должно быть написано.

SIP диалоги выведите в sngrep в другое окно, это гораздо удобнее, чем pjsip debug прямо в консоли.

Glukinho · Сообщение **Glukinho** » Вчера, 14:17

Если вы делаете Dial (или Originate в другом месте, не пойму, что используется) с таймаутом - может, этот таймаут истекает примерно тогда же, когда кто-то берет трубку, и астериск рвет звонок? Поставьте таймаут побольше или вообще уберите.

ded · Сообщение **ded** » Вчера, 16:09

Onotoliy, для ваших логов (тут в этом топике) планируем закупить отдельный диск, и подключить его к серверу forum.asterisk.ru

Настройка асинхронного DialPlan

Настройка асинхронного DialPlan

Re: Настройка асинхронного DialPlan

Re: Настройка асинхронного DialPlan

Re: Настройка асинхронного DialPlan

Re: Настройка асинхронного DialPlan

Re: Настройка асинхронного DialPlan

Re: Настройка асинхронного DialPlan

Re: Настройка асинхронного DialPlan