SIP/2.0 488 Not acceptable here

Very often problem is with incompatible codecs.

Example:

Capabilities: us - 0x0 (nothing), peer - audio=0x906 (gsm|ulaw|g726|g729)/video=0x0 (nothing), combined - 0x0 (nothing)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)

<--- Reliably Transmitting (NAT) to 11.22.33.44:5060 --->
SIP/2.0 488 Not acceptable here

This line is important:

Capabilities: us - 0x0 (nothing), peer - audio=0x906 (gsm|ulaw|g726|g729)/video=0x0 (nothing), combined - 0x0 (nothing)

One end does not allow any codecs for this connection.

Enable some of these codecs: gsm|ulaw|g726|g729 and call should pass.

It can happen that Asterisk will return "488 Not acceptable here" if caller device asks for T38 support, but it is no enabled in Device settings.

Example of such invite:

<--- SIP read from UDP:1.1.1.1:5060 --->
INVITE sip:1234567890@2.2.2.2;user=phone SIP/2.0
Max-Forwards: 69
Session-Expires: 3600;refresher=uac
Min-SE: 600
Supported: timer, 100rel
To: <sip:1234567890@2.2.2.2:5060;user=phone>
From: <sip:0987654321@1.1.1.1>;tag=3557129436-488864
P-Asserted-Identity: <sip:0987654321@1.1.1.1;user=phone>
Call-ID: 30549129-3557129436-488859@abc.domain.com
CSeq: 1 INVITE
Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE, MESSAGE, PUBLISH
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK5fe3cef54d2beba3e277c973dfde60d7
Contact: <sip:0987654321@1.1.1.1:5060>
Call-Info: <sip:1.1.1.1>;method="NOTIFY;Event=telephone-event;Duration=1000"
Content-Type: application/sdp
Content-Length: 360

v=0
o=SBC2 839077453 839077453 IN IP4 1.1.1.1
s=sip call
c=IN IP4 1.1.1.1
t=0 0
m=audio 52754 RTP/AVP 18 8 101
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
m=image 52756 udptl t38
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:1024
a=T38FaxMaxDatagram:238

In this case option "T.38 Support" should be enabled on Device settings to allow call go through.

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Check also:

• SIP Error Codes