search for: sipfarm

...times to no avail. It seems older vers of ast don't do this. near end is error: Sip read: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 206.80.66.138:5060;branch=z9hG4bK243afb2c From: "Matt S" <sip:+13142664000@206.80.66.138>;tag=as5f22d23c To: <sip:3142498555@sipfarm.netlogic.net>;tag=b27e1a1d33761e85846fc98f5f3a7e 58.2b61 Call-ID: 6b3bf846529168f37cd8465357608dee@206.80.66.138 CSeq: 102 INVITE Proxy-Authenticate: Digest realm="sipfarm.netlogic.net", nonce="blah blah", qop="auth" Server: Sip EXpress router (0.8.14 (i386/linux))...

Does anyone know if Asterisk can accept cleartext auth (SIP), as in it recv's a call destined to: 1234:blah@har.har.com The problem I'm having is simply for faxing, normal calls come in as g729 and of course we need ULAW for faxes. sip.conf snippet [sipfarm] insecure=very host=blahblah.netlogic.net type=peer context=sip-out username=+18165551212 secret=blah canreinvite=no disallow=all allow=g729 When a call comes in it's always matching that sip peer, as you can see we're prefering g729 because of voice. My thought was to create another pee...

I asked this question once before with no answer. Hopefully someone can help me as I cannot see a way to do this. I am wanting to differentiate inbound calls voice from FAX. The purpose of course voice gets g729 and FAX gets 711 (ulaw). The problem I'm having is everytime it matches the SIP peer (like it should) but it's always goes to the prefered codec. Anyone have suggestions on how to