The other side of this question is that Audiocodes gateways can handle those "status 407" messages. Try the following: Configure a user, say M1000, on the sipx. On the mediant side configure user (M1000) for "whole gateway registration", but don't enable registration. Then mediant will use that user for handling "407 proxy authentication required". And you'll be able to use M1000's permissions in the sipx to control which dialplan rules are "permitted" for the calls from this gateway. Regards, Nikolay.
> -----Original Message----- > From: [email protected] > [mailto:[email protected]] On Behalf Of > Heros Deidda > Sent: Friday, May 28, 2010 7:54 PM > To: [email protected] > Subject: [sipx-users] audiocode mediant 1000 receive as > answer from sipx a "407 authorization required" > > Hello guys, > I'm facing a weird issue. > On a new installation a SCS 3.0 (sipx 4.04) is answering to > the audiocode INVITE with a "407 proxy authentication > required". SCS and audiocode are on LAN. > The matching rule on dialplan should call a autoattendant. > > > > I Never seen this before....why should SCS ask audiocode for > authentication? > > Any suggestion is appreciated. > > Thanks, > Heros > > > > > > Details: > > > INVITE sip:[email protected];user=phone SIP/2.0 > Via: SIP/2.0/UDP 172.16.172.3;branch=z9hG4bKac688684571 > Max-Forwards: 70 > From: <sip:[email protected]>;tag=1c688678648 > To: <sip:[email protected];user=phone> > Call-ID: [email protected] > CSeq: 1 INVITE > Contact: <sip:[email protected]> > Supported: > em,100rel,timer,replaces,path,early-session,resource-priority > Allow: > REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO > ,SUBSCRIBE,UPDATE > User-Agent: Audiocodes-Sip-Gateway-Mediant 1000/v.5.40A.032 > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 261 > > v=0 > o=AudiocodesGW 688671077 688670754 IN IP4 172.16.172.3 > s=Phone-Call > c=IN IP4 172.16.172.3 > t=0 0 > m=audio 6300 RTP/AVP 8 96 > a=rtpmap:8 PCMA/8000 > a=rtpmap:96 telephone-event/8000 > a=fmtp:96 0-15 > a=ptime:20 > a=sendrecv > a=rtcp:6301 IN IP4 172.16.172.3 > > SIP/2.0 100 Trying > From: <sip:[email protected]>;tag=1c688678648 > To: <sip:[email protected];user=phone> > Call-Id: [email protected] > Cseq: 1 INVITE > Via: SIP/2.0/UDP 172.16.172.3;branch=z9hG4bKac688684571 > Content-Length: 0 > > > SIP/2.0 407 Proxy Authentication Required > From: <sip:[email protected]>;tag=1c688678648 > To: <sip:[email protected];user=phone>;tag=88706dfb > Call-Id: [email protected] > Cseq: 1 INVITE > Via: SIP/2.0/UDP 172.16.172.3;branch=z9hG4bKac688684571 > Record-Route: <sip:172.16.172.2:5060;lr> > Proxy-Authenticate: Digest realm="20022.it", > nonce="36728e1c36347d483978517a86491c254bfd16be" > Server: sipXecs/4.0.4 sipXecs/sipXproxy (Linux) > Date: Wed, 26 May 2010 12:40:30 GMT > Content-Length: 0 > > > Ing. Heros Deidda > > Pre-sales VoIP Engineer > > > > _______________________________________________ > sipx-users mailing list [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-users > Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users > sipXecs IP PBX -- http://www.sipfoundry.org/ _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users sipXecs IP PBX -- http://www.sipfoundry.org/
