Difference between revisions of "SIP/2.0 488 Not acceptable here"
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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. | |||
<br><br> | |||
---- | ---- | ||
'''Check also:''' | '''Check also:''' | ||
* [[SIP Error Codes]] | * [[SIP Error Codes]] | ||
Latest revision as of 11:39, 20 September 2012
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.
Check also:
