In my current organization, asterisk 13.7.2 is being used but when a call is being answered from a web based soft phone we are getting SIP/2.0 488 Not Acceptable Here. Below are some of the invite messages when call is received and until call is answered.
Invite messages
INVITE sip:8j4m5m0l@vb0092b16792.invalid;transport=ws SIP/2.0
Via: SIP/2.0/WS 10.76.3.4:5060;branch=z9hG4bK5cf95f99;rport
Max-Forwards: 70
From: <sip:1061@10.76.3.4>;tag=as7b9eba32
To: <sip:8j4m5m0l@vb0092b16792.invalid;transport=ws>
Contact: <sip:1061@10.76.3.4:5060;transport=WS>
Call-ID: 428ee01c02aba04a6c22f24001de218a@10.76.3.4:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 13.7.2
Date: Wed, 08 Mar 2023 20:04:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 436
v=0
o=root 1916885320 1916885320 IN IP4 10.76.3.4
s=Asterisk PBX 13.7.2
c=IN IP4 10.76.3.4
t=0 0
m=audio 11448 RTP/SAVPF 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=connection:new
a=setup:actpass
a=fingerprint:SHA-256 81:85:52:70:54:A5:BF:7D:34:97:9B:76:E9:7B:7D:39:5D:BF:4D:2F:DE:8B:79:56:D3:71:27:2C:78:73:9B:23
a=sendrecv
SIP/2.0 100 Trying
Via: SIP/2.0/WS 10.76.3.4:5060;branch=z9hG4bK5cf95f99;rport
To: <sip:8j4m5m0l@vb0092b16792.invalid;transport=ws>
From: <sip:1061@10.76.3.4>;tag=as7b9eba32
Call-ID: 428ee01c02aba04a6c22f24001de218a@10.76.3.4:5060
CSeq: 102 INVITE
Supported: timer,ice,replaces,outbound
Content-Length: 0
SIP/2.0 180 Ringing
Via: SIP/2.0/WS 10.76.3.4:5060;branch=z9hG4bK5cf95f99;rport
To: <sip:8j4m5m0l@vb0092b16792.invalid;transport=ws>;tag=0ul22ku1bt
From: <sip:1061@10.76.3.4>;tag=as7b9eba32
Call-ID: 428ee01c02aba04a6c22f24001de218a@10.76.3.4:5060
CSeq: 102 INVITE
Contact: <sip:8j4m5m0l@vb0092b16792.invalid;transport=ws>
Supported: timer,ice,replaces,outbound
Content-Length: 0
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/WS 10.76.3.4:5060;branch=z9hG4bK5cf95f99;rport
To: <sip:8j4m5m0l@vb0092b16792.invalid;transport=ws>;tag=0ul22ku1bt
From: <sip:1061@10.76.3.4>;tag=as7b9eba32
Call-ID: 428ee01c02aba04a6c22f24001de218a@10.76.3.4:5060
CSeq: 102 INVITE
Supported: timer,ice,replaces,outbound
Content-Length: 0
Configuration files
;sip.conf
[general]
realm=127.0.0.1 ; Replace this with your IP address
udpbindaddr=127.0.0.1 ; Replace this with your IP address
transport=udp
[1060] ; This will be WebRTC client
type=friend
username=1060 ; The Auth user for SIP.js
host=dynamic ; Allows any host to register
secret=password ; The SIP Password for SIP.js
encryption=yes ; Tell Asterisk to use encryption for this peer
avpf=yes ; Tell Asterisk to use AVPF for this peer
icesupport=yes ; Tell Asterisk to use ICE for this peer
context=default ; Tell Asterisk which context to use when this peer is dialing
directmedia=no ; Asterisk will relay media for this peer
transport=udp,ws,wss ; Asterisk will allow this peer to register on UDP or WebSockets
force_avp=yes ; Force Asterisk to use avp. Introduced in Asterisk 11.11
dtlsenable=yes ; Tell Asterisk to enable DTLS for this peer
dtlsverify=fingerprint ; Tell Asterisk to verify DTLS fingerprint
dtlscertfile=/etc/asterisk/keys/asterisk.pem ; Tell Asterisk where your DTLS cert file is
dtlssetup=actpass ; Tell Asterisk to use actpass SDP parameter when setting up DTLS
rtcp_mux=yes ; Tell Asterisk to do RTCP mux
[1061] ; This will be the legacy SIP client
type=friend
username=1061
host=dynamic
secret=password
context=default
;extensions.conf
[default]
exten => 1060,1,Dial(SIP/1060) ; Dialing 1060 will call the SIP client registered to 1060
exten => 1061,1,Dial(SIP/1061) ; Dialing 1061 will call the SIP client registered to 1061