Matthew, thank you very much for the fast reply and very likely the solution!<br>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.<br>
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?<br><br>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).<br>
<br><br><div class="gmail_quote">2012/6/18 Matthew Jordan <span dir="ltr"><<a href="mailto:mjordan@digium.com" target="_blank">mjordan@digium.com</a>></span><br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<br>
<br>
----- Original Message -----<br>
<br>
> From: "Stefan at WPF" <<a href="mailto:stefan.at.wpf@googlemail.com">stefan.at.wpf@googlemail.com</a>><br>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"<br>
> <<a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a>><br>
> Sent: Monday, June 18, 2012 3:04:32 PM<br>
> Subject: [asterisk-users] Error SIP/2.0 488 Not acceptable here<br>
<div><div class="h5"><br>
> Hello,<br>
<br>
> a person trying to call me by my phone number is getting the error<br>
> 488 Not acceptable here. I googled that error, seems like this error<br>
> is normally caused by a failed codec negotation, though I have no<br>
> clue how I could have read this out of the logs. Anyway, my setup is<br>
> as follows:<br>
> Asterisk 1.8.13.0 - NAT - Sipgate SIP Provider<br>
> The user calling me is also using Sipgate and is calling my landline<br>
> phone number from Sipgate (not [my sip id]@ <a href="http://sipgate.de" target="_blank">sipgate.de</a> ).<br>
<br>
> My sip.conf including the codec restrictions looks like this (I left<br>
> out my local sip account)<br>
<br>
> > [general]<br>
><br>
> > port=5060<br>
><br>
> > bindaddr=0.0.0.0<br>
><br>
> > context=other<br>
><br>
> > language=de<br>
><br>
> > allowguest=no<br>
><br>
<br>
> > qualify=no<br>
><br>
> > disallow=all<br>
><br>
> > allow=alaw<br>
><br>
> > allow=ulaw<br>
><br>
> > allow=g729<br>
><br>
> > allow=gsm<br>
><br>
> > allow=slinear<br>
><br>
> > srvlookup=yes<br>
><br>
<br>
</div></div>> > register => <MY_SIP_ID>:<password>@ <a href="http://sipgate.de/" target="_blank">sipgate.de/</a> <MY_SIP_ID><br>
<div class="im">><br>
<br>
> > [sipgate]<br>
><br>
> > type=friend<br>
><br>
> > insecure=invite<br>
><br>
> > nat=yes<br>
><br>
> > username=<MY_SIP_ID><br>
><br>
> > fromuser=<MY_SIP_ID><br>
><br>
> > fromdomain= <a href="http://sipgate.de" target="_blank">sipgate.de</a><br>
><br>
> > secret=<password><br>
><br>
> > host= <a href="http://sipgate.de" target="_blank">sipgate.de</a><br>
><br>
> > qualify=yes<br>
><br>
> > canreinvite=no<br>
><br>
> > dtmfmode=rfc2833<br>
><br>
> > context = from_external_voip_provider<br>
><br>
<br>
> The relevant part from my full asterisk log /var/log/asterisk/full<br>
> including the 488 Not acceptable here error message:<br>
<br>
> > [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c:<br>
><br>
> > <--- SIP read from UDP: <a href="http://217.10.79.9:5060" target="_blank">217.10.79.9:5060</a> ---><br>
><br>
</div>> > INVITE sip:<MY_SIP_ID>@ <a href="http://192.168.5.11:5060" target="_blank">192.168.5.11:5060</a> SIP/2.0<br>
<div class="im">><br>
> > Record-Route: <sip:217.10.79.9;lr;ftag=8cgn1bajqb><br>
><br>
> > Record-Route: <sip:172.20.40.3;lr=on><br>
><br>
> > Record-Route: <sip:217.10.79.9;lr;ftag=8cgn1bajqb><br>
><br>
> > Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bK8f5c.48627b3.0<br>
><br>
> > Via: SIP/2.0/UDP 172.20.40.3;branch=z9hG4bK8f5c.48627b3.0<br>
><br>
> > Via: SIP/2.0/UDP<br>
> > 217.10.79.9:5060;received=217.10.68.222;branch=z9hG4bK-un6p0cm50qse<br>
><br>
> > Via: SIP/2.0/UDP<br>
> > 192.168.0.8:2048;received=<CALLING_PARTY_IP_ADDRESS>;branch=z9hG4bK-un6p0cm50qse;rport=2048<br>
><br>
> > From: " <a href="http://sipgate.de" target="_blank">sipgate.de</a> " <sip:<CALLING_PARTY_SIP_ID>@ <a href="http://sipgate.de" target="_blank">sipgate.de</a><br>
> > >;tag=8cgn1bajqb<br>
><br>
</div>> > To: <sip:0049<MY_PHONE_NUMBER>@ <a href="http://sipgate.de" target="_blank">sipgate.de</a> ;user=phone><br>
<div><div class="h5">><br>
> > Call-ID: 4fdf703d880d-ywqwnfbbj1h7<br>
><br>
> > CSeq: 2 INVITE<br>
><br>
> > Max-Forwards: 67<br>
><br>
> > Contact:<br>
> > <sip:<CALLING_PARTY_SIP_ID>@<CALLING_PARTY_IP_ADDRESS>:2048;line=swnt2d3t>;reg-id=1<br>
><br>
> > X-Serialnumber: 000413251D76<br>
><br>
> > User-Agent: snom300/ 8.7.3.7<br>
><br>
> > Accept: application/sdp<br>
><br>
> > Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE,<br>
> > PRACK, MESSAGE, INFO, UPDATE<br>
><br>
> > Allow-Events: talk, hold, refer, call-info<br>
><br>
> > Supported: timer, 100rel, replaces, from-change<br>
><br>
> > Session-Expires: 3600;refresher=uas<br>
><br>
> > Min-SE: 90<br>
><br>
> > Content-Type: application/sdp<br>
><br>
> > Content-Length: 522<br>
><br>
> > P-Asserted-Identity: <sip:<CALLING_PARTY_PHONE_NUMBER>@ <a href="http://sipgate.de" target="_blank">sipgate.de</a><br>
> > ><br>
><br>
<br>
> > v=0<br>
><br>
> > o=root 269390684 269390684 IN IP4 192.168.0.8<br>
><br>
> > s=call<br>
><br>
> > c=IN IP4 217.10.77.20<br>
><br>
> > t=0 0<br>
><br>
> > m=audio 62652 RTP/AVP 9 0 8 3 99 108 18 101<br>
><br>
> > a=crypto:1 AES_CM_128_HMAC_SHA1_32<br>
> > inline:Ed8iHaP3BXNVeXHj98PRa6sJyImNer3ImjUvDZps<br>
><br>
> > a=rtpmap:9 G722/8000<br>
><br>
> > a=rtpmap:0 PCMU/8000<br>
><br>
> > a=rtpmap:8 PCMA/8000<br>
><br>
> > a=rtpmap:3 GSM/8000<br>
><br>
> > a=rtpmap:99 G726-32/8000<br>
><br>
> > a=rtpmap:108 AAL2-G726-32/8000<br>
><br>
> > a=rtpmap:18 G729/8000<br>
><br>
> > a=fmtp:18 annexb=no<br>
><br>
> > a=rtpmap:101 telephone-event/8000<br>
><br>
> > a=fmtp:101 0-15<br>
><br>
> > a=ptime:20<br>
><br>
> > a=sendrecv<br>
><br>
> > a=direction:active<br>
><br>
> > a=nortpproxy:yes<br>
><br>
> > <-------------><br>
><br>
> > [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: --- (25 headers 21<br>
> > lines)<br>
> > ---<br>
><br>
> > [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Sending to<br>
> > <a href="http://217.10.79.9:5060" target="_blank">217.10.79.9:5060</a> (NAT)<br>
><br>
> > [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Using INVITE request as<br>
> > basis request - 4fdf703d880d-ywqwnfbbj1h7<br>
><br>
> > [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found peer 'sipgate'<br>
> > for<br>
> > '<CALLING_PARTY_SIP_ID>' from <a href="http://217.10.79.9:5060" target="_blank">217.10.79.9:5060</a><br>
><br>
> > [Jun 18 20:15:26] VERBOSE[1164] netsock2.c: == Using SIP RTP CoS<br>
> > mark<br>
> > 5<br>
><br>
> > [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found RTP audio format<br>
> > 9<br>
><br>
> > [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found RTP audio format<br>
> > 0<br>
><br>
> > [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found RTP audio format<br>
> > 8<br>
><br>
> > [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found RTP audio format<br>
> > 3<br>
><br>
> > [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found RTP audio format<br>
> > 99<br>
><br>
> > [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found RTP audio format<br>
> > 108<br>
><br>
> > [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found RTP audio format<br>
> > 18<br>
><br>
> > [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found RTP audio format<br>
> > 101<br>
><br>
> > [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found audio description<br>
> > format G722 for ID 9<br>
><br>
> > [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found audio description<br>
> > format PCMU for ID 0<br>
><br>
> > [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found audio description<br>
> > format PCMA for ID 8<br>
><br>
> > [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found audio description<br>
> > format GSM for ID 3<br>
><br>
> > [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found audio description<br>
> > format G726-32 for ID 99<br>
><br>
> > [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found audio description<br>
> > format AAL2-G726-32 for ID 108<br>
><br>
> > [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found audio description<br>
> > format G729 for ID 18<br>
><br>
> > [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c: Found audio description<br>
> > format telephone-event for ID 101<br>
><br>
> > [Jun 18 20:15:26] WARNING[1164] chan_sip.c: We are requesting SRTP,<br>
> > but they responded without it!<br>
><br>
> > [Jun 18 20:15:26] VERBOSE[1164] chan_sip.c:<br>
><br>
> > <--- Reliably Transmitting (NAT) to <a href="http://217.10.79.9:5060" target="_blank">217.10.79.9:5060</a> ---><br>
><br>
> > SIP/2.0 488 Not acceptable here<br>
><br>
> > Via: SIP/2.0/UDP<br>
> > 217.10.79.9:5060;branch=z9hG4bK8f5c.48627b3.0;received=217.10.79.9;rport=5060<br>
><br>
> > Via: SIP/2.0/UDP 172.20.40.3;branch=z9hG4bK8f5c.48627b3.0<br>
><br>
> > Via: SIP/2.0/UDP<br>
> > 217.10.79.9:5060;received=217.10.68.222;branch=z9hG4bK-un6p0cm50qse<br>
><br>
> > Via: SIP/2.0/UDP<br>
> > 192.168.0.8:2048;received=<CALLING_PARTY_IP_ADDRESS>;branch=z9hG4bK-un6p0cm50qse;rport=2048<br>
><br>
> > From: " <a href="http://sipgate.de" target="_blank">sipgate.de</a> " <sip:<CALLING_PARTY_SIP_ID>@ <a href="http://sipgate.de" target="_blank">sipgate.de</a><br>
> > >;tag=8cgn1bajqb<br>
><br>
> > To: <sip:0049<MY_PHONE_NUMBER>@ <a href="http://sipgate.de" target="_blank">sipgate.de</a><br>
</div></div><div class="im">> > ;user=phone>;tag=as6364b798<br>
><br>
> > Call-ID: 4fdf703d880d-ywqwnfbbj1h7<br>
><br>
> > CSeq: 2 INVITE<br>
><br>
> > Server: Asterisk PBX 1.8.13.0~dfsg-1<br>
><br>
> > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,<br>
> > INFO, PUBLISH<br>
><br>
> > Supported: replaces, timer<br>
><br>
> > Content-Length: 0<br>
><br>
<br>
> I am having problems to see to what "488 Not acceptable here" relates<br>
> to? What is not acceptable? Is it maybe about<br>
<br>
> > [Jun 18 20:15:26] WARNING[1164] chan_sip.c: We are requesting SRTP,<br>
> > but they responded without it!<br>
><br>
<br>
</div>Yes, that would be the problem.<br>
<br>
The SIP UA is doing something a little wrong here by offering a security<br>
description (crypto) without specifying that the audio/video protocol that<br>
should be used as SRTP (RTP/SAVP). Because the UA appears to be attempting<br>
to negotiate a SRTP connection, Asterisk is checking if the peer has encryption<br>
enabled. Since the peer corresponding with the UA does not have encryption<br>
enabled for it, Asterisk is responding with a 488 response.<br>
<br>
SRTP security descriptions (such as 'crypto') must only be used with the<br>
SRTP transport specified, e.g., RTP/SAVP or RTP/SAVPF.<br>
<br>
--<br>
Matthew Jordan<br>
Digium, Inc. | Software Developer<br>
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA<br>
Check us out at: <a href="http://digium.com" target="_blank">http://digium.com</a> & <a href="http://asterisk.org" target="_blank">http://asterisk.org</a><br>
<br>
--<br>
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</blockquote></div><br>