Pjsip responds with a 488 Not Acceptable Here

Asterisk Asterisk SIP
1

Hi!

My asterisk+pjsip responded to an INVITE with a 488 Not Acceptable Here, and I am seeking help:

*CLI> core set verbose 6
Console verbose was 5 and is now 6.
*CLI> core set debug 6
Core debug was 5 and is now 6.
*CLI> pjsip set logger on
PJSIP Logging enabled
*CLI>
*CLI> <--- Received SIP request (1025 bytes) from TLS:180.12.160.67:443 --->
INVITE sip:05037085114@103.125.216.190:5061 SIP/2.0
Via: SIP/2.0/TLS 180.12.160.67:443;branch=z9hG4bKNA+p2O3O, SIP/2.0/UDP 10.10.10.13:6061;branch=z9hG4bKFYNJeeTf
Record-Route: <sip:nxs_4d74a7c4_flow=74b2711372464121@180.12.160.67:443;transport=tcp;lr>
From: <sip:08026735774@voip.ntt.com>;tag=5af0deb0
To: <sip:05037085114@voip.ntt.com>
Call-ID: GWSN2f9c49404ffd10438e06000069e3@10.124.237.107--_6bbb6075
CSeq: 1 INVITE
Max-Forwards: 69
Contact: <sip:10.10.10.13:6061;transport=udp>
Supported: timer
Allow: INVITE,ACK,BYE,CANCEL,UPDATE,PRACK
Session-Expires: 180;refresher=uas
Min-SE: 180
Privacy: none
P-Asserted-Identity: <sip:08026735774@voip.ntt.com>
Content-Type: application/sdp
Content-Length: 300

v=0
o=- 1598825884 1598825884 IN IP4 180.12.160.85
s=-
c=IN IP4 180.12.160.85
t=0 0
m=audio 50624 RTP/SAVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:zXT41XxDZAuMZYTUt56HTcREboPeF0IOI1XVeBRX

  == Setting global variable 'SIPDOMAIN' to '103.125.216.190'
<--- Transmitting SIP response (484 bytes) to TLS:180.12.160.67:443 --->
SIP/2.0 100 Trying
Via: SIP/2.0/TLS 180.12.160.67:443;rport=443;received=180.12.160.67;branch=z9hG4bKNA+p2O3O
Via: SIP/2.0/UDP 10.10.10.13:6061;branch=z9hG4bKFYNJeeTf
Record-Route: <sip:nxs_4d74a7c4_flow=74b2711372464121@180.12.160.67:443;transport=tcp;lr>
Call-ID: GWSN2f9c49404ffd10438e06000069e3@10.124.237.107--_6bbb6075
From: <sip:08026735774@voip.ntt.com>;tag=5af0deb0
To: <sip:05037085114@voip.ntt.com>
CSeq: 1 INVITE
Server: Asterisk PBX 17.6.0
Content-Length:  0


<--- Transmitting SIP response (538 bytes) to TLS:180.12.160.67:443 --->
SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/TLS 180.12.160.67:443;rport=443;received=180.12.160.67;branch=z9hG4bKNA+p2O3O
Via: SIP/2.0/UDP 10.10.10.13:6061;branch=z9hG4bKFYNJeeTf
Record-Route: <sip:nxs_4d74a7c4_flow=74b2711372464121@180.12.160.67:443;transport=tcp;lr>
Call-ID: GWSN2f9c49404ffd10438e06000069e3@10.124.237.107--_6bbb6075
From: <sip:08026735774@voip.ntt.com>;tag=5af0deb0
To: <sip:05037085114@voip.ntt.com>;tag=3794d0d3-be31-4dfa-9766-7082c4e25650
CSeq: 1 INVITE
Server: Asterisk PBX 17.6.0
Content-Length:  0


<--- Received SIP request (358 bytes) from TLS:180.12.160.67:443 --->
ACK sip:05037085114@103.125.216.190:5061 SIP/2.0
Via: SIP/2.0/TLS 180.12.160.67:443;branch=z9hG4bKNA+p2O3O
From: <sip:08026735774@voip.ntt.com>;tag=5af0deb0
To: <sip:05037085114@voip.ntt.com>;tag=3794d0d3-be31-4dfa-9766-7082c4e25650
Call-ID: GWSN2f9c49404ffd10438e06000069e3@10.124.237.107--_6bbb6075
CSeq: 1 ACK
Max-Forwards: 69
Content-Length: 0



*CLI>

Thank you in advance for your help!
Tatsuo

2

You have not provided the configuration of Asterisk so purely guessing I would say you have not configured it for SDES media encryption, or the res_srtp module is not built/loaded. You may also have not enabled G729 as an allowed codec.

3

you have not configured it for SDES media encryption

This was it. Thank you jcolp!

4

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