----- Original Message -----
> From: "Stefan at WPF" <[email protected]> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <[email protected]> > 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 -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
