On 3/29/2011 12:52 PM, Jeremy Kister wrote: > I recently configured a SIP peer which i must specify my fromuser as > my ten digit "DID". I send calls to this peer, but whenever Asterisk > sends an options message, the fromuser is "asterisk". > > Is this a bug? Or is there some other config I must make ? > > > > register = 2155551941:[email protected]/2155551941~600 > > [peer](!) > type=peer > context=inbound > qualify=yes > qualifyfreq=300 > insecure=port,invite > nat=yes > outgoinglimit=4 > incominglimit=4 > > [mypeer](peer) > host=10.0.138.226 > defaultuser=2155551941 > fromuser=2155551941 > md5secret=023f30a320a5781e8ffd1af9888012af > incominglimit=10 > > > IP (tos 0x0, ttl 64, id 9242, offset 0, flags [none], proto UDP (17), > length 555) 10.0.1.3.5060 > 10.0.138.226.5060: SIP, length: 527 > OPTIONS sip:10.0.138.226 SIP/2.0 > Via: SIP/2.0/UDP 10.0.83.61:5060;branch=z9hG4bK6abb74e3;rport > Max-Forwards: 70 > From: "asterisk" <sip:[email protected]>;tag=as7444eb08 > To: <sip:10.0.138.226> > Contact: <sip:[email protected]:5060> > Call-ID: [email protected]:5060 > CSeq: 102 OPTIONS > User-Agent: Asterisk PBX 1.8.2.3 > Date: Tue, 29 Mar 2011 17:43:05 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, > NOTIFY, INFO, PUBLISH > Supported: replaces > Content-Length: 0 > > > IP (tos 0xb8, ttl 250, id 0, offset 0, flags [none], proto UDP (17), > length 411) 10.0.138.226.5060 > 10.0.1.3.5060: SIP, length: 383 > SIP/2.0 403 From: URI not recognized > Via: SIP/2.0/UDP > 10.245.83.61:5060;received=10.0.83.61;branch=z9hG4bK6abb74e3;rport=5060 > From: "asterisk" <sip:[email protected]>;tag=as7444eb08 > To: <sip:10.0.138.226>;tag=metaswitch+1+0+e288612a > Call-ID: [email protected]:5060 > CSeq: 102 OPTIONS > Server: DC-SIP/2.0 > Organization: > Content-Length: 0 > >
IIRC, you need to define the fromuser in the peer in order for the qualify checks (options packets) to contain the user you want -- Sherwood McGowan <[email protected]> Carrier, ITSP, Call Center, and PBX Solutions Consultant -- _____________________________________________________________________ -- 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
