[asterisk-bugs] [JIRA] (ASTERISK-30028) Can't make outgoing calls : sip/2.0 401 unauthorized
Hery RARIVO (JIRA)
noreply at issues.asterisk.org
Tue Apr 26 08:43:40 CDT 2022
Hery RARIVO created ASTERISK-30028:
--------------------------------------
Summary: Can't make outgoing calls : sip/2.0 401 unauthorized
Key: ASTERISK-30028
URL: https://issues.asterisk.org/jira/browse/ASTERISK-30028
Project: Asterisk
Issue Type: Bug
Security Level: None
Components: Resources/res_pjsip
Affects Versions: 16.25.2
Environment: Ubuntu 20.04
Reporter: Hery RARIVO
Severity: Major
Hi,
I can't make outgoing calls.I thought it was my ISP but they say that there is no traffic from me. It's just ringing and after " Everyone is busy/congested at this time (1:0/0/1)
So heres's my pjsip.conf :
[transport-udp]
type=transport
protocol=udp
bind=0.0.0.0
local_net = 192.168.0.0/24
external_media_address = 217.146.224.140
external_signaling_address = 217.146.224.140
[0174901008]
type=endpoint
context=sipmivoaka
disallow=all
allow=g729
allow=ulaw
allow=alaw
aors=0174901008
auth=0174901008
[0174901008]
type=aor
max_contacts=1
[0174901008]
type=auth
auth_type=userpass
password=test
username=0174901008
Log :
PJSIP Logging enabled
<--- Received SIP request (1016 bytes) from UDP:192.168.0.22:22706 --->
INVITE sip:0969363030 at 192.168.0.2 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.22:22706;branch=z9hG4bK-d87543-291fa2309f5d423c-1--d87543-;rport
Max-Forwards: 70
Contact: <sip:0174901008 at 192.168.0.22:22706>
To: "0969363030"<sip:0969363030 at 192.168.0.2>
From: <sip:0174901008 at 192.168.0.2>;tag=a463f547
Call-ID: ODVkYmVkMmE4YjdiYjEwNmQ5OTVhMjMxYTM2NWE3Y2E.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1011s stamp 41150
Content-Length: 467
v=0
o=- 5 2 IN IP4 192.168.0.22
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.0.22
t=0 0
m=audio 3880 RTP/AVP 107 119 100 106 0 105 98 8 101
a=alt:1 2 : Y7jmOD3P eVN2xzMi 192.168.0.22 3880
a=alt:2 1 : ReUU43bs BHpj9E9S 192.168.56.1 3880
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:119 BV32-FEC/16000
a=rtpmap:100 SPEEX/16000
a=rtpmap:106 SPEEX-FEC/16000
a=rtpmap:105 SPEEX-FEC/8000
a=rtpmap:98 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
<--- Transmitting SIP response (559 bytes) to UDP:192.168.0.22:22706 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.22:22706;rport=22706;received=192.168.0.22;branch=z9hG4bK-d87543-291fa2309f5d423c-1--d87543-
Call-ID: ODVkYmVkMmE4YjdiYjEwNmQ5OTVhMjMxYTM2NWE3Y2E.
From: <sip:0174901008 at 192.168.0.2>;tag=a463f547
To: "0969363030" <sip:0969363030 at 192.168.0.2>;tag=z9hG4bK-d87543-291fa2309f5d423c-1--d87543-
CSeq: 1 INVITE
WWW-Authenticate: Digest realm="asterisk",nonce="1650980332/8708d9b872bc30e1f038e02c9451dce6",opaque="663078bc7460d60f",algorithm=md5,qop="auth"
Server: Asterisk PBX 16.25.2
Content-Length: 0
<--- Received SIP request (365 bytes) from UDP:192.168.0.22:22706 --->
ACK sip:0969363030 at 192.168.0.2 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.22:22706;branch=z9hG4bK-d87543-291fa2309f5d423c-1--d87543-;rport
To: "0969363030" <sip:0969363030 at 192.168.0.2>;tag=z9hG4bK-d87543-291fa2309f5d423c-1--d87543-
From: <sip:0174901008 at 192.168.0.2>;tag=a463f547
Call-ID: ODVkYmVkMmE4YjdiYjEwNmQ5OTVhMjMxYTM2NWE3Y2E.
CSeq: 1 ACK
Content-Length: 0
<--- Received SIP request (1310 bytes) from UDP:192.168.0.22:22706 --->
INVITE sip:0969363030 at 192.168.0.2 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.22:22706;branch=z9hG4bK-d87543-8220d018a045397b-1--d87543-;rport
Max-Forwards: 70
Contact: <sip:0174901008 at 192.168.0.22:22706>
To: "0969363030"<sip:0969363030 at 192.168.0.2>
From: <sip:0174901008 at 192.168.0.2>;tag=a463f547
Call-ID: ODVkYmVkMmE4YjdiYjEwNmQ5OTVhMjMxYTM2NWE3Y2E.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1011s stamp 41150
Authorization: Digest username="0174901008",realm="asterisk",nonce="1650980332/8708d9b872bc30e1f038e02c9451dce6",uri="sip:0969363030 at 192.168.0.2",response="573750b29d0db61028cd4b1c6fa44793",cnonce="557780aeb6d493f9e95a4e958c65383d",nc=00000001,qop=auth,algorithm=md5,opaque="663078bc7460d60f"
Content-Length: 467
v=0
o=- 5 2 IN IP4 192.168.0.22
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.0.22
t=0 0
m=audio 3880 RTP/AVP 107 119 100 106 0 105 98 8 101
a=alt:1 2 : Y7jmOD3P eVN2xzMi 192.168.0.22 3880
a=alt:2 1 : ReUU43bs BHpj9E9S 192.168.56.1 3880
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:119 BV32-FEC/16000
a=rtpmap:100 SPEEX/16000
a=rtpmap:106 SPEEX-FEC/16000
a=rtpmap:105 SPEEX-FEC/8000
a=rtpmap:98 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
<--- Transmitting SIP response (360 bytes) to UDP:192.168.0.22:22706 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.22:22706;rport=22706;received=192.168.0.22;branch=z9hG4bK-d87543-8220d018a045397b-1--d87543-
Call-ID: ODVkYmVkMmE4YjdiYjEwNmQ5OTVhMjMxYTM2NWE3Y2E.
From: <sip:0174901008 at 192.168.0.2>;tag=a463f547
To: "0969363030" <sip:0969363030 at 192.168.0.2>
CSeq: 2 INVITE
Server: Asterisk PBX 16.25.2
Content-Length: 0
-- Executing [0969363030 at sipmivoaka:1] Set("PJSIP/0174901008-00000000", "CALLERID(num)=0974901008") in new stack
-- Executing [0969363030 at sipmivoaka:2] Gosub("PJSIP/0174901008-00000000", "my-gosub,s,1") in new stack
-- Executing [s at my-gosub:1] NoOp("PJSIP/0174901008-00000000", "") in new stack
-- Executing [s at my-gosub:2] Set("PJSIP/0174901008-00000000", "fname=/records/mp3/1650980332.0-2022-04-26-13_38-0974901008-s") in new stack
-- Executing [s at my-gosub:3] Set("PJSIP/0174901008-00000000", "CDR(filename)=1650980332.0-2022-04-26-13_38-0974901008-s.mp3") in new stack
-- Executing [s at my-gosub:4] Set("PJSIP/0174901008-00000000", "MONITOR_OPT=nice -n 19 /usr/bin/lame -b 16 -silent "/records/mp3/1650980332.0-2022-04-26-13_38-0974901008-s.wav" "/records/mp3/1650980332.0-2022-04-26-13_38-0974901008-s.mp3" && rm -f "/records/mp3/1650980332.0-2022-04-26-13_38-0974901008-s.wav"") in new stack
-- Executing [s at my-gosub:5] MixMonitor("PJSIP/0174901008-00000000", "/records/mp3/1650980332.0-2022-04-26-13_38-0974901008-s.wav,b,nice -n 19 /usr/bin/lame -b 16 -silent "/records/mp3/1650980332.0-2022-04-26-13_38-0974901008-s.wav" "/records/mp3/1650980332.0-2022-04-26-13_38-0974901008-s.mp3" && rm -f "/records/mp3/1650980332.0-2022-04-26-13_38-0974901008-s.wav"") in new stack
== Begin MixMonitor Recording PJSIP/0174901008-00000000
-- Executing [s at my-gosub:6] Return("PJSIP/0174901008-00000000", "") in new stack
-- Executing [0969363030 at sipmivoaka:3] Dial("PJSIP/0174901008-00000000", "PJSIP/0969363030 at axialys,40,tr") in new stack
-- Called PJSIP/0969363030 at axialys
<--- Transmitting SIP response (546 bytes) to UDP:192.168.0.22:22706 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.22:22706;rport=22706;received=192.168.0.22;branch=z9hG4bK-d87543-8220d018a045397b-1--d87543-
Call-ID: ODVkYmVkMmE4YjdiYjEwNmQ5OTVhMjMxYTM2NWE3Y2E.
From: <sip:0174901008 at 192.168.0.2>;tag=a463f547
To: "0969363030" <sip:0969363030 at 192.168.0.2>;tag=85aa64e6-f3c8-4101-b3da-84d092c117b7
CSeq: 2 INVITE
Server: Asterisk PBX 16.25.2
Contact: <sip:192.168.0.2:5060>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Content-Length: 0
<--- Transmitting SIP request (1011 bytes) to UDP:217.146.224.140:5060 --->
INVITE sip:0969363030 at sip-ng.axialys.net SIP/2.0
Via: SIP/2.0/UDP 217.146.224.140:5060;rport;branch=z9hG4bKPj47c39e29-79d1-4922-b6cc-1dbeb027d73e
From: <sip:0974901008 at sip-ng.axialys.net>;tag=064e4245-db58-48ab-89b3-fd35fa3bc589
To: <sip:0969363030 at sip-ng.axialys.net>
Contact: <sip:asterisk at 217.146.224.140:5060>
Call-ID: bfe6b798-1415-42eb-8de0-b6e3ecd3476e
CSeq: 3600 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 16.25.2
Content-Type: application/sdp
Content-Length: 314
v=0
o=- 1086778031 1086778031 IN IP4 217.146.224.140
s=Asterisk
c=IN IP4 217.146.224.140
t=0 0
m=audio 18246 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/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
<--- Transmitting SIP request (1011 bytes) to UDP:217.146.224.140:5060 --->
INVITE sip:0969363030 at sip-ng.axialys.net SIP/2.0
Via: SIP/2.0/UDP 217.146.224.140:5060;rport;branch=z9hG4bKPj47c39e29-79d1-4922-b6cc-1dbeb027d73e
From: <sip:0974901008 at sip-ng.axialys.net>;tag=064e4245-db58-48ab-89b3-fd35fa3bc589
To: <sip:0969363030 at sip-ng.axialys.net>
Contact: <sip:asterisk at 217.146.224.140:5060>
Call-ID: bfe6b798-1415-42eb-8de0-b6e3ecd3476e
CSeq: 3600 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 16.25.2
Content-Type: application/sdp
Content-Length: 314
v=0
o=- 1086778031 1086778031 IN IP4 217.146.224.140
s=Asterisk
c=IN IP4 217.146.224.140
t=0 0
m=audio 18246 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/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
<--- Transmitting SIP request (1011 bytes) to UDP:217.146.224.140:5060 --->
INVITE sip:0969363030 at sip-ng.axialys.net SIP/2.0
Via: SIP/2.0/UDP 217.146.224.140:5060;rport;branch=z9hG4bKPj47c39e29-79d1-4922-b6cc-1dbeb027d73e
From: <sip:0974901008 at sip-ng.axialys.net>;tag=064e4245-db58-48ab-89b3-fd35fa3bc589
To: <sip:0969363030 at sip-ng.axialys.net>
Contact: <sip:asterisk at 217.146.224.140:5060>
Call-ID: bfe6b798-1415-42eb-8de0-b6e3ecd3476e
CSeq: 3600 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 16.25.2
Content-Type: application/sdp
Content-Length: 314
v=0
o=- 1086778031 1086778031 IN IP4 217.146.224.140
s=Asterisk
c=IN IP4 217.146.224.140
t=0 0
m=audio 18246 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/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
<--- Transmitting SIP request (1011 bytes) to UDP:217.146.224.140:5060 --->
INVITE sip:0969363030 at sip-ng.axialys.net SIP/2.0
Via: SIP/2.0/UDP 217.146.224.140:5060;rport;branch=z9hG4bKPj47c39e29-79d1-4922-b6cc-1dbeb027d73e
From: <sip:0974901008 at sip-ng.axialys.net>;tag=064e4245-db58-48ab-89b3-fd35fa3bc589
To: <sip:0969363030 at sip-ng.axialys.net>
Contact: <sip:asterisk at 217.146.224.140:5060>
Call-ID: bfe6b798-1415-42eb-8de0-b6e3ecd3476e
CSeq: 3600 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 16.25.2
Content-Type: application/sdp
Content-Length: 314
v=0
o=- 1086778031 1086778031 IN IP4 217.146.224.140
s=Asterisk
c=IN IP4 217.146.224.140
t=0 0
m=audio 18246 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/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
<--- Transmitting SIP request (1011 bytes) to UDP:217.146.224.140:5060 --->
INVITE sip:0969363030 at sip-ng.axialys.net SIP/2.0
Via: SIP/2.0/UDP 217.146.224.140:5060;rport;branch=z9hG4bKPj47c39e29-79d1-4922-b6cc-1dbeb027d73e
From: <sip:0974901008 at sip-ng.axialys.net>;tag=064e4245-db58-48ab-89b3-fd35fa3bc589
To: <sip:0969363030 at sip-ng.axialys.net>
Contact: <sip:asterisk at 217.146.224.140:5060>
Call-ID: bfe6b798-1415-42eb-8de0-b6e3ecd3476e
CSeq: 3600 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 16.25.2
Content-Type: application/sdp
Content-Length: 314
v=0
o=- 1086778031 1086778031 IN IP4 217.146.224.140
s=Asterisk
c=IN IP4 217.146.224.140
t=0 0
m=audio 18246 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/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
<--- Transmitting SIP request (1011 bytes) to UDP:217.146.224.140:5060 --->
INVITE sip:0969363030 at sip-ng.axialys.net SIP/2.0
Via: SIP/2.0/UDP 217.146.224.140:5060;rport;branch=z9hG4bKPj47c39e29-79d1-4922-b6cc-1dbeb027d73e
From: <sip:0974901008 at sip-ng.axialys.net>;tag=064e4245-db58-48ab-89b3-fd35fa3bc589
To: <sip:0969363030 at sip-ng.axialys.net>
Contact: <sip:asterisk at 217.146.224.140:5060>
Call-ID: bfe6b798-1415-42eb-8de0-b6e3ecd3476e
CSeq: 3600 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 16.25.2
Content-Type: application/sdp
Content-Length: 314
v=0
o=- 1086778031 1086778031 IN IP4 217.146.224.140
s=Asterisk
c=IN IP4 217.146.224.140
t=0 0
m=audio 18246 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/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
<--- Received SIP request (548 bytes) from UDP:192.168.0.22:22706 --->
SUBSCRIBE sip:0174901008 at 192.168.0.2 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.22:22706;branch=z9hG4bK-d87543-462e1405613ff014-1--d87543-;rport
Max-Forwards: 70
Contact: <sip:0174901008 at 192.168.0.22:22706>
To: <sip:0174901008 at 192.168.0.2>
From: <sip:0174901008 at 192.168.0.2>;tag=3b387b37
Call-ID: ZTU4ZDY3ZjQxOTEwMzM4NWQ1N2QzYWM4ZjI5ZTFiYzA.
CSeq: 1 SUBSCRIBE
Expires: 300
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
User-Agent: X-Lite release 1011s stamp 41150
Event: message-summary
Content-Length: 0
<--- Transmitting SIP response (549 bytes) to UDP:192.168.0.22:22706 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.22:22706;rport=22706;received=192.168.0.22;branch=z9hG4bK-d87543-462e1405613ff014-1--d87543-
Call-ID: ZTU4ZDY3ZjQxOTEwMzM4NWQ1N2QzYWM4ZjI5ZTFiYzA.
From: <sip:0174901008 at 192.168.0.2>;tag=3b387b37
To: <sip:0174901008 at 192.168.0.2>;tag=z9hG4bK-d87543-462e1405613ff014-1--d87543-
CSeq: 1 SUBSCRIBE
WWW-Authenticate: Digest realm="asterisk",nonce="1650980354/4cecd3f7fbca64f7f5411d7f6ec4d7d7",opaque="4be95f0921e61bb6",algorithm=md5,qop="auth"
Server: Asterisk PBX 16.25.2
Content-Length: 0
<--- Received SIP request (842 bytes) from UDP:192.168.0.22:22706 --->
SUBSCRIBE sip:0174901008 at 192.168.0.2 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.22:22706;branch=z9hG4bK-d87543-332e8d3ff6149171-1--d87543-;rport
Max-Forwards: 70
Contact: <sip:0174901008 at 192.168.0.22:22706>
To: <sip:0174901008 at 192.168.0.2>
From: <sip:0174901008 at 192.168.0.2>;tag=3b387b37
Call-ID: ZTU4ZDY3ZjQxOTEwMzM4NWQ1N2QzYWM4ZjI5ZTFiYzA.
CSeq: 2 SUBSCRIBE
Expires: 300
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
User-Agent: X-Lite release 1011s stamp 41150
Authorization: Digest username="0174901008",realm="asterisk",nonce="1650980354/4cecd3f7fbca64f7f5411d7f6ec4d7d7",uri="sip:0174901008 at 192.168.0.2",response="06da050976be81aa74a3d06c55dd6f5a",cnonce="f89f148ffcb3974cbe9e0e3fab47acdf",nc=00000001,qop=auth,algorithm=md5,opaque="4be95f0921e61bb6"
Event: message-summary
Content-Length: 0
<--- Transmitting SIP response (400 bytes) to UDP:192.168.0.22:22706 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.0.22:22706;rport=22706;received=192.168.0.22;branch=z9hG4bK-d87543-332e8d3ff6149171-1--d87543-
Call-ID: ZTU4ZDY3ZjQxOTEwMzM4NWQ1N2QzYWM4ZjI5ZTFiYzA.
From: <sip:0174901008 at 192.168.0.2>;tag=3b387b37
To: <sip:0174901008 at 192.168.0.2>;tag=z9hG4bK-d87543-332e8d3ff6149171-1--d87543-
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX 16.25.2
Content-Length: 0
<--- Transmitting SIP request (1011 bytes) to UDP:217.146.224.140:5060 --->
INVITE sip:0969363030 at sip-ng.axialys.net SIP/2.0
Via: SIP/2.0/UDP 217.146.224.140:5060;rport;branch=z9hG4bKPj47c39e29-79d1-4922-b6cc-1dbeb027d73e
From: <sip:0974901008 at sip-ng.axialys.net>;tag=064e4245-db58-48ab-89b3-fd35fa3bc589
To: <sip:0969363030 at sip-ng.axialys.net>
Contact: <sip:asterisk at 217.146.224.140:5060>
Call-ID: bfe6b798-1415-42eb-8de0-b6e3ecd3476e
CSeq: 3600 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 16.25.2
Content-Type: application/sdp
Content-Length: 314
v=0
o=- 1086778031 1086778031 IN IP4 217.146.224.140
s=Asterisk
c=IN IP4 217.146.224.140
t=0 0
m=audio 18246 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/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
== Everyone is busy/congested at this time (1:0/0/1)
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