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

Mon Jun 18 16:05:04 CDT 2012

Matthew, thank you very much for the fast reply and very likely the
solution!
Using your hint I could locally reproduce the "488 Not Acceptable" on my
Snom 300 with FW 8.7.3.7 by setting RTP encryption (SRTP) to on and
RTP/SAVP to off (RTP/SAVP mandatory or SRTP off -> all fine). The person
calling me has the same phone and FW, so that should really be the problem,
I will tell him to change it.
However I am wondering why it's possible to configure a Snom phone in such
a wrong way at all? Is that necessary for some legacy systems?

Also I am wondering if it's possible to tell asterisk to just ignore the
crypto line when the profile is just RTP/AVP / not to take things that
serious? In this specific case I can tell the calling person to change the
setting, but unfortunately I can't tell this every person calling me (not
only because they simply can't call me).

2012/6/18 Matthew Jordan <mjordan at digium.com>

>
>
> ----- 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
>
> --
> _____________________________________________________________________
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