[asterisk-users] Error SIP/2.0 488 Not acceptable here

[Matthew Jordan]

----- Original Message ----- 

> From: "Stefan at WPF" <stefan.at.wpf at googlemail.com>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users at lists.digium.com>
> Sent: Monday, June 18, 2012 3:04:32 PM
> Subject: [asterisk-users] Error SIP/2.0 488 Not acceptable here

> Hello,

> a person trying to call me by my phone number is getting the error
> 488 Not acceptable here. I googled that error, seems like this error
> is normally caused by a failed codec negotation, though I have no
> clue how I could have read this out of the logs. Anyway, my setup is
> as follows:
> Asterisk 1.8.13.0 - NAT - Sipgate SIP Provider
> The user calling me is also using Sipgate and is calling my landline
> phone number from Sipgate (not [my sip id]@ sipgate.de ).

> My sip.conf including the codec restrictions looks like this (I left
> out my local sip account)

> > [general]
> 
> > port=5060
> 
> > bindaddr=0.0.0.0
> 
> > context=other
> 
> > language=de
> 
> > allowguest=no
> 

> > qualify=no
> 
> > disallow=all
> 
> > allow=alaw
> 
> > allow=ulaw
> 
> > allow=g729
> 
> > allow=gsm
> 
> > allow=slinear
> 
> > srvlookup=yes
> 

> > register => <MY_SIP_ID>:<password>@ sipgate.de/ <MY_SIP_ID>
> 

> > [sipgate]
> 
> > type=friend
> 
> > insecure=invite
> 
> > nat=yes
> 
> > username=<MY_SIP_ID>
> 
> > fromuser=<MY_SIP_ID>
> 
> > fromdomain= sipgate.de
> 
> > secret=<password>
> 
> > host= sipgate.de
> 
> > qualify=yes
> 
> > canreinvite=no
> 
> > dtmfmode=rfc2833
> 
> > context = from_external_voip_provider
> 

> The relevant part from my full asterisk log /var/log/asterisk/full
> including the 488 Not acceptable here error message:

> > [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c:
> 
> > <--- SIP read from UDP: 217.10.79.9:5060 --->
> 
> > INVITE sip:<MY_SIP_ID>@ 192.168.5.11:5060 SIP/2.0
> 
> > Record-Route: <sip:217.10.79.9;lr;ftag=8cgn1bajqb>
> 
> > Record-Route: <sip:172.20.40.3;lr=on>
> 
> > Record-Route: <sip:217.10.79.9;lr;ftag=8cgn1bajqb>
> 
> > Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bK8f5c.48627b3.0
> 
> > Via: SIP/2.0/UDP 172.20.40.3;branch=z9hG4bK8f5c.48627b3.0
> 
> > Via: SIP/2.0/UDP
> > 217.10.79.9:5060;received=217.10.68.222;branch=z9hG4bK-un6p0cm50qse
> 
> > Via: SIP/2.0/UDP
> > 192.168.0.8:2048;received=<CALLING_PARTY_IP_ADDRESS>;branch=z9hG4bK-un6p0cm50qse;rport=2048
> 
> > From: " sipgate.de " <sip:<CALLING_PARTY_SIP_ID>@ sipgate.de
> > >;tag=8cgn1bajqb
> 
> > To: <sip:0049<MY_PHONE_NUMBER>@ sipgate.de ;user=phone>
> 
> > Call-ID: 4fdf703d880d-ywqwnfbbj1h7
> 
> > CSeq: 2 INVITE
> 
> > Max-Forwards: 67
> 
> > Contact:
> > <sip:<CALLING_PARTY_SIP_ID>@<CALLING_PARTY_IP_ADDRESS>:2048;line=swnt2d3t>;reg-id=1
> 
> > X-Serialnumber: 000413251D76
> 
> > User-Agent: snom300/ 8.7.3.7
> 
> > Accept: application/sdp
> 
> > Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE,
> > PRACK, MESSAGE, INFO, UPDATE
> 
> > Allow-Events: talk, hold, refer, call-info
> 
> > Supported: timer, 100rel, replaces, from-change
> 
> > Session-Expires: 3600;refresher=uas
> 
> > Min-SE: 90
> 
> > Content-Type: application/sdp
> 
> > Content-Length: 522
> 
> > P-Asserted-Identity: <sip:<CALLING_PARTY_PHONE_NUMBER>@ sipgate.de
> > >
> 

> > v=0
> 
> > o=root 269390684 269390684 IN IP4 192.168.0.8
> 
> > s=call
> 
> > c=IN IP4 217.10.77.20
> 
> > t=0 0
> 
> > m=audio 62652 RTP/AVP 9 0 8 3 99 108 18 101
> 
> > a=crypto:1 AES_CM_128_HMAC_SHA1_32
> > inline:Ed8iHaP3BXNVeXHj98PRa6sJyImNer3ImjUvDZps
> 
> > a=rtpmap:9 G722/8000
> 
> > a=rtpmap:0 PCMU/8000
> 
> > a=rtpmap:8 PCMA/8000
> 
> > a=rtpmap:3 GSM/8000
> 
> > a=rtpmap:99 G726-32/8000
> 
> > a=rtpmap:108 AAL2-G726-32/8000
> 
> > a=rtpmap:18 G729/8000
> 
> > a=fmtp:18 annexb=no
> 
> > a=rtpmap:101 telephone-event/8000
> 
> > a=fmtp:101 0-15
> 
> > a=ptime:20
> 
> > a=sendrecv
> 
> > a=direction:active
> 
> > a=nortpproxy:yes
> 
> > <------------->
> 
> > [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: --- (25 headers 21
> > lines)
> > ---
> 
> > [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Sending to
> > 217.10.79.9:5060 (NAT)
> 
> > [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Using INVITE request as
> > basis request - 4fdf703d880d-ywqwnfbbj1h7
> 
> > [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found peer 'sipgate'
> > for
> > '<CALLING_PARTY_SIP_ID>' from 217.10.79.9:5060
> 
> > [Jun 18 20:15:26] VERBOSE[1164] netsock2.c: == Using SIP RTP CoS
> > mark
> > 5
> 
> > [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found RTP audio format
> > 9
> 
> > [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found RTP audio format
> > 0
> 
> > [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found RTP audio format
> > 8
> 
> > [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found RTP audio format
> > 3
> 
> > [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found RTP audio format
> > 99
> 
> > [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found RTP audio format
> > 108
> 
> > [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found RTP audio format
> > 18
> 
> > [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found RTP audio format
> > 101
> 
> > [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found audio description
> > format G722 for ID 9
> 
> > [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found audio description
> > format PCMU for ID 0
> 
> > [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found audio description
> > format PCMA for ID 8
> 
> > [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found audio description
> > format GSM for ID 3
> 
> > [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found audio description
> > format G726-32 for ID 99
> 
> > [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found audio description
> > format AAL2-G726-32 for ID 108
> 
> > [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found audio description
> > format G729 for ID 18
> 
> > [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found audio description
> > format telephone-event for ID 101
> 
> > [Jun 18 20:15:26] WARNING[1164] chan_sip.c: We are requesting SRTP,
> > but they responded without it!
> 
> > [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c:
> 
> > <--- Reliably Transmitting (NAT) to 217.10.79.9:5060 --->
> 
> > SIP/2.0 488 Not acceptable here
> 
> > Via: SIP/2.0/UDP
> > 217.10.79.9:5060;branch=z9hG4bK8f5c.48627b3.0;received=217.10.79.9;rport=5060
> 
> > Via: SIP/2.0/UDP 172.20.40.3;branch=z9hG4bK8f5c.48627b3.0
> 
> > Via: SIP/2.0/UDP
> > 217.10.79.9:5060;received=217.10.68.222;branch=z9hG4bK-un6p0cm50qse
> 
> > Via: SIP/2.0/UDP
> > 192.168.0.8:2048;received=<CALLING_PARTY_IP_ADDRESS>;branch=z9hG4bK-un6p0cm50qse;rport=2048
> 
> > From: " sipgate.de " <sip:<CALLING_PARTY_SIP_ID>@ sipgate.de
> > >;tag=8cgn1bajqb
> 
> > To: <sip:0049<MY_PHONE_NUMBER>@ sipgate.de
> > ;user=phone>;tag=as6364b798
> 
> > Call-ID: 4fdf703d880d-ywqwnfbbj1h7
> 
> > CSeq: 2 INVITE
> 
> > Server: Asterisk PBX 1.8.13.0~dfsg-1
> 
> > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
> > INFO, PUBLISH
> 
> > Supported: replaces, timer
> 
> > Content-Length: 0
> 

> I am having problems to see to what "488 Not acceptable here" relates
> to? What is not acceptable? Is it maybe about

> > [Jun 18 20:15:26] WARNING[1164] chan_sip.c: We are requesting SRTP,
> > but they responded without it!
> 

Yes, that would be the problem.

The SIP UA is doing something a little wrong here by offering a security
description (crypto) without specifying that the audio/video protocol that
should be used as SRTP (RTP/SAVP).  Because the UA appears to be attempting
to negotiate a SRTP connection, Asterisk is checking if the peer has encryption
enabled.  Since the peer corresponding with the UA does not have encryption
enabled for it, Asterisk is responding with a 488 response.

SRTP security descriptions (such as 'crypto') must only be used with the
SRTP transport specified, e.g., RTP/SAVP or RTP/SAVPF.

--
Matthew Jordan
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org