Getting SIP/2.0 488 Not Acceptable Here When call is answered

Asterisk Asterisk SIP
1

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
3

Also, same config file (sip.conf and extensions.conf) when I am doing in the latest version of asterisk 16.3.0, we are able to receive the call and 2 way audio is working fine…

But we have to make it work with Asterisk 13.7.2 as we have some custom changes in C code.

Kindly advise what shall I check in order to make 2 way audio work with 13.7.2 asterisk version

4

Or you could forward port your changes to a supported version… it’s probably unlikely you’re actually stuck on 13.7.2, but you haven’t said what those changes are or what the scope of them is.

5

can you please explain more about port forward option?

6

can you please explain more about port forward option?

I assumed you had developers if you have custom changes made. If you don’t have any, you would need one.

7

That is not the latest. even the highest subversion of 16 is no longer receiving normal bug fixes, and that is even older.

It’s quite possible that your basic problem is trying to run a 2023 browser against a mid 2010s implementation of WebRTC,.

8

This is one of the reasons why I was never interested in using WebRTC, even using a SIP implementation of Asterisk 1.4, in a softphone developed in 2023 everything works great because there is a standard that is maintained and respected, but with WeBRTC today something works and the next day with an update of the Web browser everything stops working.

9

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