Hello,
I installed Asterisk 15.2.1 on Amazon Ubuntu lightsails instance. On client side, i’m using sipML5 live demo on two different browsers (Chrome and Firefox). User registered successfully and call establish on both ends. After call connected, no audio receive at each end. I also check different questions related to audio problem but i got no luck.
sip.conf
[general]
context=public
allowoverlap=no
realm=voip.goz.am
udpbindaddr=0.0.0.0
tcpenable=no
tcpbindaddr=0.0.0.0
transport=udp,ws,wss
srvlookup=yes
localnet=172.26.9.37/255.255.240.0
externaddr = 54.156.244.93
icesupport = yes
I make two sip users in sip.conf:
[199]
secret=1234
context=default
host=dynamic
trustrpid=yes
sendrpid=no
type=friend
qualify=yes
qualifyfreq=600
transport=udp,ws,wss
encryption=yes
avpf=yes
icesupport=yes
rtcp_mux=yes
directmedia=no
disallow=all
allow=opus
allow=ulaw
allow=gsm
dtlsenable=yes
dtlscertfile=/etc/asterisk/keys/asterisk.pem
dtlsprivatekey=/etc/asterisk/keys/asterisk.key
dtlscafile=/etc/asterisk/keys/asterisk.crt
dtlssetup=actpass
nat=force_rport,comedia
[200]
secret=1234
context=default
host=dynamic
trustrpid=yes
sendrpid=no
type=friend
qualify=yes
qualifyfreq=600
transport=udp,ws,wss
encryption=yes
avpf=yes
icesupport=yes
rtcp_mux=yes
directmedia=no
disallow=all
allow=opus
allow=ulaw
allow=gsm
dtlsenable=yes
dtlscertfile=/etc/asterisk/keys/asterisk.pem
dtlsprivatekey=/etc/asterisk/keys/asterisk.key
dtlscafile=/etc/asterisk/keys/asterisk.crt
dtlssetup=actpass
nat=force_rport,comedia
http.conf
[general]
enabled=yes
bindaddr=0.0.0.0
bindport=8088
tlsenable=yes
tlsbindaddr=0.0.0.0:8089
tlscertfile=/etc/asterisk/keys/asterisk.pe
tlsprivatekey=/etc/asterisk/keys/asterisk.key
estensions.conf
exten => 199,1,Dial(SIP/199)
exten => 200,1,Dial(SIP/200)
rtp.conf
[general]
rtpstart=10000
rtpend=20000
icesupport=yes
stunaddr=stun.l.google.com:19302
sip debug logs from asterisk cli
<------------>
-- Executing [199@default:1] Dial("SIP/200-00000004", "SIP/199") in new stack
== DTLS ECDH initialized (automatic), faster PFS enabled
== Using SIP RTP CoS mark 5
Audio is at 16954
Adding codec opus to SDP
Adding codec ulaw to SDP
Adding codec gsm to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 39.46.144.245:49876:
INVITE sips:199@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=wss SIP/2.0
Via: SIP/2.0/WS 54.156.244.93:0;branch=z9hG4bK28c59243;rport
Max-Forwards: 70
From: "200" <sip:200@54.156.244.93:0>;tag=as69ea055c
To: <sips:199@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=wss>
Contact: <sip:200@54.156.244.93:0;transport=ws>
Call-ID: 24eef8092f81b7da53482b3e591377b8@54.156.244.93:0
CSeq: 102 INVITE
User-Agent: Asterisk PBX 15.2.1
Date: Fri, 16 Feb 2018 04:04:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 894
v=0
o=root 1629693611 1629693611 IN IP4 54.156.244.93
s=Asterisk PBX 15.2.1
c=IN IP4 54.156.244.93
t=0 0
m=audio 16954 UDP/TLS/RTP/SAVPF 107 0 3 101
a=rtpmap:107 opus/48000/2
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:20
a=ice-ufrag:3e62beb420061b3e4468600b624ea555
a=ice-pwd:4a77054329ddc969477349985aa45bc6
a=candidate:Hac1a0925 1 UDP 2130706431 172.26.9.37 16954 typ host
a=candidate:S369cf45d 1 UDP 1694498815 54.156.244.93 16954 typ srflx raddr 172.26.9.37 rport 16954
a=candidate:Hac1a0925 2 UDP 2130706430 172.26.9.37 16955 typ host
a=candidate:S369cf45d 2 UDP 1694498814 54.156.244.93 16955 typ srflx raddr 172.26.9.37 rport 16955
a=connection:new
a=setup:actpass
a=fingerprint:SHA-256 E7:F1:3D:80:27:CD:09:0E:51:CB:67:AA:D8:73:F6:E3:8E:34:BB:57:F8:3F:10:E4:B2:D7:AA:AD:19:6E:18:BE
a=rtcp-mux
a=sendrecv
---
-- Called SIP/199
<--- SIP read from WS:39.46.144.245:49876 --->
SIP/2.0 100 Trying (sent from the Transaction Layer)
Via: SIP/2.0/WS 54.156.244.93;rport;branch=z9hG4bK28c59243
From: "200"<sip:200@54.156.244.93>;tag=as69ea055c
To: <sips:199@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=wss>
Call-ID: 24eef8092f81b7da53482b3e591377b8@54.156.244.93:0
CSeq: 102 INVITE
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
<--- SIP read from WS:39.46.144.245:49876 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/WS 54.156.244.93;rport;branch=z9hG4bK28c59243
From: "200"<sip:200@54.156.244.93>;tag=as69ea055c
To: <sips:199@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=wss>;tag=6yX0vJDyET65tQ9HuuOI
Contact: <sips:199@df7jal23ls0d.invalid;transport=wss>
Call-ID: 24eef8092f81b7da53482b3e591377b8@54.156.244.93:0
CSeq: 102 INVITE
Content-Length: 0
Allow: ACK, BYE, CANCEL, INVITE, MESSAGE, NOTIFY, OPTIONS, PRACK, REFER, UPDATE
<------------->
--- (9 headers 0 lines) ---
sip_route_dump: route/path hop: <sips:199@df7jal23ls0d.invalid;transport=wss>
-- SIP/199-00000005 is ringing
<--- Transmitting (NAT) to 39.46.144.245:50046 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKculJxqFObtidPG1tHg5XU1nMMm8Pop4F;received=39.46.144.245;rport=50046
From: "200"<sip:200@voip.goz.am>;tag=vtz5PRDczEUzLXzut4ko
To: <sip:199@voip.goz.am>;tag=as7549aaf6
Call-ID: d2d10d93-a6e3-9cee-b0a1-60269f0dd099
CSeq: 11626 INVITE
Server: Asterisk PBX 15.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:199@54.156.244.93:0;transport=ws>
Content-Length: 0
<------------>
-- SIP/199-00000005 requested media update control 26, passing it to SIP/200-00000004
<--- SIP read from UDP:216.170.119.151:5131 --->
<------------->
<--- SIP read from UDP:216.170.119.151:5131 --->
REGISTER sip:54.156.244.93 SIP/2.0
Via: SIP/2.0/UDP 216.170.119.151:5131;branch=z9hG4bK14049ba87f8e28d85a865846;rport
From: "200" <sip:200@54.156.244.93>;tag=1527ba8494d
To: "200" <sip:200@54.156.244.93>
Call-ID: ba820c4-24803bb6851d5ee-a7408c2c@54.156.244.93
CSeq: 887 REGISTER
Contact: <sip:200@216.170.119.151:5131>
User-Agent: VaxSIPUserAgent/3.5
Expires: 0
Max-Forwards: 70
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
Sending to 216.170.119.151:5131 (no NAT)
<--- Transmitting (NAT) to 216.170.119.151:5131 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 216.170.119.151:5131;branch=z9hG4bK14049ba87f8e28d85a865846;received=216.170.119.151;rport=5131
From: "200" <sip:200@54.156.244.93>;tag=1527ba8494d
To: "200" <sip:200@54.156.244.93>;tag=as3b4b539f
Call-ID: ba820c4-24803bb6851d5ee-a7408c2c@54.156.244.93
CSeq: 887 REGISTER
Server: Asterisk PBX 15.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="voip.goz.am", nonce="62f97d64"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'ba820c4-24803bb6851d5ee-a7408c2c@54.156.244.93' in 32000 ms (Method: REGISTER)
<--- SIP read from UDP:216.170.119.151:5131 --->
REGISTER sip:54.156.244.93 SIP/2.0
Via: SIP/2.0/UDP 216.170.119.151:5131;branch=z9hG4bK14050ba87fcc28d85a865846;rport
From: "200" <sip:200@54.156.244.93>;tag=1527ba8494d
To: "200" <sip:200@54.156.244.93>
Call-ID: ba820c4-24803bb6851d5ee-a7408c2c@54.156.244.93
CSeq: 888 REGISTER
Contact: <sip:200@216.170.119.151:5131>
Authorization: Digest username="200", realm="voip.goz.am", nonce="62f97d64", uri="sip:54.156.244.93", response="d19d7c48a24132989f96d1ad2ef2ea64", algorithm=MD5
User-Agent: VaxSIPUserAgent/3.5
Expires: 0
Max-Forwards: 70
Content-Length: 0
<------------->
--- (12 headers 0 lines) ---
Sending to 216.170.119.151:5131 (no NAT)
-- Unregistered SIP '200'
<--- Transmitting (NAT) to 216.170.119.151:5131 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 216.170.119.151:5131;branch=z9hG4bK14050ba87fcc28d85a865846;received=216.170.119.151;rport=5131
From: "200" <sip:200@54.156.244.93>;tag=1527ba8494d
To: "200" <sip:200@54.156.244.93>;tag=as3b4b539f
Call-ID: ba820c4-24803bb6851d5ee-a7408c2c@54.156.244.93
CSeq: 888 REGISTER
Server: Asterisk PBX 15.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 0
Date: Fri, 16 Feb 2018 04:05:01 GMT
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'ba820c4-24803bb6851d5ee-a7408c2c@54.156.244.93' in 32000 ms (Method: REGISTER)
<--- SIP read from UDP:216.170.119.151:5105 --->
REGISTER sip:54.156.244.93 SIP/2.0
Via: SIP/2.0/UDP 216.170.119.151:5105;branch=z9hG4bK14051ba889cf28e15a865849;rport
From: "200" <sip:200@54.156.244.93>;tag=1533ba8b2b0
To: "200" <sip:200@54.156.244.93>
Call-ID: ba889cf-3be7bf0c7d30ac-ac85bd86@54.156.244.93
CSeq: 889 REGISTER
Contact: <sip:200@216.170.119.151:5105>
User-Agent: VaxSIPUserAgent/3.5
Expires: 1800
Max-Forwards: 70
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
Sending to 216.170.119.151:5105 (no NAT)
Sending to 216.170.119.151:5105 (no NAT)
<--- Transmitting (NAT) to 216.170.119.151:5105 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 216.170.119.151:5105;branch=z9hG4bK14051ba889cf28e15a865849;received=216.170.119.151;rport=5105
From: "200" <sip:200@54.156.244.93>;tag=1533ba8b2b0
To: "200" <sip:200@54.156.244.93>;tag=as1d784ff2
Call-ID: ba889cf-3be7bf0c7d30ac-ac85bd86@54.156.244.93
CSeq: 889 REGISTER
Server: Asterisk PBX 15.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="voip.goz.am", nonce="6e152986"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'ba889cf-3be7bf0c7d30ac-ac85bd86@54.156.244.93' in 32000 ms (Method: REGISTER)
<--- SIP read from UDP:216.170.119.151:5105 --->
REGISTER sip:54.156.244.93 SIP/2.0
Via: SIP/2.0/UDP 216.170.119.151:5105;branch=z9hG4bK14052ba88a0d28e15a865849;rport
From: "200" <sip:200@54.156.244.93>;tag=1533ba8b2b0
To: "200" <sip:200@54.156.244.93>
Call-ID: ba889cf-3be7bf0c7d30ac-ac85bd86@54.156.244.93
CSeq: 890 REGISTER
Contact: <sip:200@216.170.119.151:5105>
Authorization: Digest username="200", realm="voip.goz.am", nonce="6e152986", uri="sip:54.156.244.93", response="25f86226c18e554d7352675c4e81eff3", algorithm=MD5
User-Agent: VaxSIPUserAgent/3.5
Expires: 1800
Max-Forwards: 70
Content-Length: 0
<------------->
--- (12 headers 0 lines) ---
Sending to 216.170.119.151:5105 (no NAT)
-- Registered SIP '200' at 216.170.119.151:5105
Reliably Transmitting (NAT) to 216.170.119.151:5105:
OPTIONS sip:200@216.170.119.151:5105 SIP/2.0
Via: SIP/2.0/UDP 54.156.244.93:5060;branch=z9hG4bK7eb14c36;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@54.156.244.93>;tag=as5a76256c
To: <sip:200@216.170.119.151:5105>
Contact: <sip:asterisk@54.156.244.93:5060>
Call-ID: 4dcef79670a3de437ce0254b01d8b72f@54.156.244.93:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 15.2.1
Date: Fri, 16 Feb 2018 04:05:04 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
---
<--- Transmitting (NAT) to 216.170.119.151:5105 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 216.170.119.151:5105;branch=z9hG4bK14052ba88a0d28e15a865849;received=216.170.119.151;rport=5105
From: "200" <sip:200@54.156.244.93>;tag=1533ba8b2b0
To: "200" <sip:200@54.156.244.93>;tag=as1d784ff2
Call-ID: ba889cf-3be7bf0c7d30ac-ac85bd86@54.156.244.93
CSeq: 890 REGISTER
Server: Asterisk PBX 15.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 1800
Contact: <sip:200@216.170.119.151:5105>;expires=1800
Date: Fri, 16 Feb 2018 04:05:04 GMT
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'ba889cf-3be7bf0c7d30ac-ac85bd86@54.156.244.93' in 32000 ms (Method: REGISTER)
<--- SIP read from UDP:216.170.119.151:5105 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 54.156.244.93:5060;branch=z9hG4bK7eb14c36;rport
From: "asterisk" <sip:asterisk@54.156.244.93>;tag=as5a76256c
To: <sip:200@216.170.119.151:5105>
Call-ID: 4dcef79670a3de437ce0254b01d8b72f@54.156.244.93:5060
CSeq: 102 OPTIONS
Contact: <sip:200@216.170.119.151:5105>
User-Agent: VaxSIPUserAgent/3.5
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '4dcef79670a3de437ce0254b01d8b72f@54.156.244.93:5060' Method: OPTIONS
Really destroying SIP dialog 'ba7ad2a-21b5cf8017a9af5f-a61bebe6@54.156.244.93' Method: REGISTER
Really destroying SIP dialog 'ba82c6c-fe3518525990947-d7397140@216.170.119.151' Method: ACK
Really destroying SIP dialog '6a1a421f-7511-16e0-acf5-8ddead2ac1f4' Method: REGISTER
== WebSocket connection from '39.46.144.245:49984' closed
Really destroying SIP dialog 'ba839da-fb4667681eb321d9-710b602@216.170.119.151' Method: ACK
<--- SIP read from UDP:216.170.119.151:5105 --->
INVITE sip:24648322132922@54.156.244.93 SIP/2.0
Via: SIP/2.0/UDP 216.170.119.151:5105;branch=z9hG4bK14053ba8a3b028f55a86584f;rport
From: "200" <sip:200@54.156.244.93>;tag=1534ba8cca5
To: "24648322132922" <sip:24648322132922@54.156.244.93>
Call-ID: ba8a3b0-12adc2f1c972700-15c3d6c0@216.170.119.151
CSeq: 1 INVITE
Contact: <sip:200@216.170.119.151:5105>
User-Agent: VaxSIPUserAgent/3.5
Max-Forwards: 70
Allow: ACK, INFO, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, SUBSCRIBE, MESSAGE, PRACK
Content-Type: application/sdp
Content-Length: 241
v=0
o=200 15187538710 15187538710 IN IP4 216.170.119.151
s=VaxSoft
c=IN IP4 216.170.119.151
t=0 0
m=audio 7450 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=sendrecv
<------------->
--- (12 headers 11 lines) ---
Sending to 216.170.119.151:5105 (no NAT)
Sending to 216.170.119.151:5105 (no NAT)
Using INVITE request as basis request - ba8a3b0-12adc2f1c972700-15c3d6c0@216.170.119.151
Found peer '200' for '200' from 216.170.119.151:5105
<--- Reliably Transmitting (NAT) to 216.170.119.151:5105 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 216.170.119.151:5105;branch=z9hG4bK14053ba8a3b028f55a86584f;received=216.170.119.151;rport=5105
From: "200" <sip:200@54.156.244.93>;tag=1534ba8cca5
To: "24648322132922" <sip:24648322132922@54.156.244.93>;tag=as7b24000b
Call-ID: ba8a3b0-12adc2f1c972700-15c3d6c0@216.170.119.151
CSeq: 1 INVITE
Server: Asterisk PBX 15.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="voip.goz.am", nonce="3498b278"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'ba8a3b0-12adc2f1c972700-15c3d6c0@216.170.119.151' in 6400 ms (Method: INVITE)
<--- SIP read from UDP:216.170.119.151:5105 --->
ACK sip:24648322132922@54.156.244.93 SIP/2.0
Via: SIP/2.0/UDP 216.170.119.151:5105;branch=z9hG4bK14053ba8a3b028f55a86584f;rport
From: "200" <sip:200@54.156.244.93>;tag=1534ba8cca5
To: "24648322132922" <sip:24648322132922@54.156.244.93>;tag=as7b24000b
Call-ID: ba8a3b0-12adc2f1c972700-15c3d6c0@216.170.119.151
CSeq: 1 ACK
Contact: <sip:200@216.170.119.151:5105>
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from UDP:216.170.119.151:5105 --->
INVITE sip:24648322132922@54.156.244.93 SIP/2.0
Via: SIP/2.0/UDP 216.170.119.151:5105;branch=z9hG4bK14054ba8a54628f85a865850;rport
From: "200" <sip:200@54.156.244.93>;tag=1534ba8cca5
To: "24648322132922" <sip:24648322132922@54.156.244.93>
Call-ID: ba8a3b0-12adc2f1c972700-15c3d6c0@216.170.119.151
CSeq: 2 INVITE
Contact: <sip:200@216.170.119.151:5105>
Authorization: Digest username="200", realm="voip.goz.am", nonce="3498b278", uri="sip:24648322132922@54.156.244.93", response="13197cda257e5a1b8407e6354958186d", algorithm=MD5
User-Agent: VaxSIPUserAgent/3.5
Max-Forwards: 70
Allow: ACK, INFO, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, SUBSCRIBE, MESSAGE, PRACK
Content-Type: application/sdp
Content-Length: 241
v=0
o=200 15187538710 15187538710 IN IP4 216.170.119.151
s=VaxSoft
c=IN IP4 216.170.119.151
t=0 0
m=audio 7450 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=sendrecv
<------------->
--- (13 headers 11 lines) ---
Sending to 216.170.119.151:5105 (NAT)
Using INVITE request as basis request - ba8a3b0-12adc2f1c972700-15c3d6c0@216.170.119.151
Found peer '200' for '200' from 216.170.119.151:5105
== DTLS ECDH initialized (automatic), faster PFS enabled
== Using SIP RTP CoS mark 5
[Feb 16 04:05:11] NOTICE[12073][C-000000d0]: chan_sip.c:10430 process_sdp: Received AVP profile in audio answer but AVPF is enabled, disabling: audio 7450 RTP/AVP 0 8 101
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
[Feb 16 04:05:11] WARNING[12073][C-000000d0]: chan_sip.c:10837 process_sdp: Failed to receive SDP offer/answer with required SRTP crypto attributes for audio
<--- Reliably Transmitting (NAT) to 216.170.119.151:5105 --->
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP 216.170.119.151:5105;branch=z9hG4bK14054ba8a54628f85a865850;received=216.170.119.151;rport=5105
From: "200" <sip:200@54.156.244.93>;tag=1534ba8cca5
To: "24648322132922" <sip:24648322132922@54.156.244.93>;tag=as7b24000b
Call-ID: ba8a3b0-12adc2f1c972700-15c3d6c0@216.170.119.151
CSeq: 2 INVITE
Server: Asterisk PBX 15.2.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'ba8a3b0-12adc2f1c972700-15c3d6c0@216.170.119.151' in 6400 ms (Method: INVITE)
<--- SIP read from UDP:216.170.119.151:5105 --->
ACK sip:24648322132922@54.156.244.93 SIP/2.0
Via: SIP/2.0/UDP 216.170.119.151:5105;branch=z9hG4bK14054ba8a54628f85a865850;rport
From: "200" <sip:200@54.156.244.93>;tag=1534ba8cca5
To: "24648322132922" <sip:24648322132922@54.156.244.93>;tag=as7b24000b
Call-ID: ba8a3b0-12adc2f1c972700-15c3d6c0@216.170.119.151
CSeq: 2 ACK
Contact: <sip:200@216.170.119.151:5105>
Authorization: Digest username="200", realm="voip.goz.am", nonce="3498b278", uri="sip:24648322132922@54.156.244.93", response="08fb39ef88f0d291ed16aefde2b50dd7", algorithm=MD5
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog 'db377864-2b15-3c8e-021d-3f416d03d4ab' Method: REGISTER
Really destroying SIP dialog 'ba84062-8881eb3bc93d7a4-eacaaafc@216.170.119.151' Method: ACK
Really destroying SIP dialog 'ba84738-2d4753a42529d780-af671808@216.170.119.151' Method: ACK
I am beginner to asterisk and i am learning from asterisk forums and from different sites. But I’m stuck here and I don’t understand where to go from here.