Thanks so much for pointing this out. I was curious why the commands in the documentation differed to the commands I was using.
That problem is fixed, but now I have a new issue. I can call with no issues, however, as soon as I answer one of the calls I see the error: ast_srtp_unprotect: SRTP unprotect: authentication failure. Below is a snippet of the debug as the call is answered. v=0 o=root 306031538 306031538 IN IP4 172.16.200.60 s=Asterisk PBX 1.8.2.4 c=IN IP4 172.16.200.60 t=0 0 m=audio 15274 RTP/SAVP 0 3 96 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=ptime:20 a=sendrecv a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:iINHae+LvAVdSJwhOJjE3BtyZLVuYFG6ctUjDZst <------------> [Mar 3 15:02:25] WARNING[13599]: res_srtp.c:338 ast_srtp_unprotect: SRTP unprotect: authentication failure <--- SIP read from TLS:172.16.201.10:50600 ---> BYE sip:[email protected]:5061;transport=TLS SIP/2.0 Via: SIP/2.0/TLS 172.16.201.10:50600;rport;branch=z9hG4bKPjbLo4aOOGOax.f5DovLkV-rasCIhsca7A Max-Forwards: 70 From: "Asterisk" <sip:[email protected]>;tag=Kbf7ZANMEn4pRtHrYTZJkOfqYg226z-I To: <sip:[email protected]>;tag=as21b6a1ac Call-ID: LWPc00KmvuwzLJfizX-2.7fBtE8ILwhX CSeq: 6714 BYE Content-Length: 0 <-------------> --- (8 headers 0 lines) --- <--- Reliably Transmitting (NAT) to 172.16.201.10:50600 ---> SIP/2.0 487 Request Terminated Via: SIP/2.0/TLS 172.16.201.10:50600;branch=z9hG4bKPjbJVHFgqcrclq3kJh9hDZfg-I6joRN3QL;received=172.16.201.10;rport=50600 From: "Asterisk" <sip:[email protected]>;tag=Kbf7ZANMEn4pRtHrYTZJkOfqYg226z-I To: <sip:[email protected]>;tag=as21b6a1ac Call-ID: LWPc00KmvuwzLJfizX-2.7fBtE8ILwhX CSeq: 6713 INVITE Server: Asterisk PBX 1.8.2.4 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0 > > Message: 8 > Date: Tue, 1 Mar 2011 10:04:14 -0600 > From: Terry Wilson <[email protected]> > Subject: Re: [asterisk-users] TLS/SRTP calls go to circuit busy. > To: Asterisk Users Mailing List - Non-Commercial Discussion > <[email protected]> > Message-ID: <[email protected]> > Content-Type: text/plain; charset="us-ascii" > > On Feb 28, 2011, at 7:19 PM, mitch Johnson wrote: > >> I'm in the process of testing a TLS/SRTP install. My experience is >> improving with each new challenge, but this one is a great test of my 2 >> month experience with Asterisk. > >> [myphones] >> >> ;exten => 6001,1,Dial(SIP/6001) >> ;exten => 6001,2,Hangup() >> exten => 6001,1,Set(_SIPSRTP_CRYPTO=enable) >> exten => 6001,2,Dial(SIP/${EXTEN}) >> > > There is no such thing as the _SIPSRTP_CRYPTO variable. That was from a very > old version of the SRTP patch. Ignore pretty much anything on issue 5413 and > instead look at > https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial and > https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Specifics. You > would use encryption=yes/no in sip.conf and > Set(CHANNEL(secure_bridge_signaling)=1) to force SRTP calls. I'm assuming > that you are using Asterisk 1.8 instead of one of the patches on issue > 5413--if not, then do that. ;-) > > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > <http://lists.digium.com/pipermail/asterisk-users/attachments/20110301/f3436edc/attachment-0001.htm> > > ------------------------------ > -- _____________________________________________________________________ -- 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
