For anybody that read my post, I got it working again. *AFTER* the phone decided it could not connect to my primary server and then failed over to my secondary server (the polycoms can do this), I then had to unplug the router and then plug it back in. I have absolutely no idea why this is.
My phone was still able to contacts the primary server, it just would not authorize properly. Maybe when I unplugged my phone and plugged it back into the router, the router assigned it a new local port number and asterisk was caching the NAT port and did not authorize when they did not match. Anybody know why this happened? - Gabe ----- Original Message ----- From: "Gabriel Afana" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Sent: Monday, March 13, 2006 5:56 PM Subject: Re: [Asterisk-Users] Woooo, Polycom and * on crack - can'tregister! > I just noticed this message with "sip debug" on: > > > Transmitting (no NAT) to 24.50.66.128:5060: > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP > 24.50.66.128:5060;branch=z9hG4bK1da0c8655B1F9600;received=24.50.66.128 > From: "Gabriel Afana" <sip:[EMAIL PROTECTED]>;tag=2CB88E3F-E17FB2BC > To: <sip:[EMAIL PROTECTED]>;tag=as743ca65c > Call-ID: [EMAIL PROTECTED] > CSeq: 1 REGISTER > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY > Max-Forwards: 70 > Contact: <sip:[EMAIL PROTECTED]> > WWW-Authenticate: Digest realm="asterisk", nonce="77bfdfcb" > Content-Length: 0 > > > Pretty obvious, there is a problem with the registration info. However, my > sip info didn't change. Nothing changed. All my sip.conf info matches my > 501 exactly (as it did before). And its not giving me the usual message on > the CLI saying there is an unauthorized registration; I am not seeing > anything on the CLI. > > Any ideas? this just started happening. > > - Gabe > > > > > ----- Original Message ----- > From: "Gabriel Afana" <[EMAIL PROTECTED]> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users@lists.digium.com> > Sent: Monday, March 13, 2006 5:34 PM > Subject: [Asterisk-Users] Woooo, Polycom and * on crack - can't register! > > > > Hey guys, > > Got a strange one. I've been using my Polycom Phone 501 on Asterisk > for > > months with no problems like this. Today I added Polycom 301 on my desk > > next to my 501 to play with some presences features and some other things. > > > > I got the 301 setup then noticed the 501 wasn't registered anymore. > > Everything is configured correct (like ususal). I disconnected the 301, > > rebooted everything (Asterisk, the 501, my router...etc). I have a backup > > server that the 501 registered to no problem, but it refuses to register > to > > my primary server. > > > > When I check "sip show peers", it shows (301, 302 and 303 are all > > extensions on my Polycom 501): > > > > 301 (Unspecified) D 0 UNKNOWN > > 302 (Unspecified) D 0 UNKNOWN > > 303 (Unspecified) D 0 UNKNOWN > > > > However, when I check "sip show channels", things get interesting: > > > > support*CLI> sip show channels > > Peer User/ANR Call ID Seq (Tx/Rx) Form Hold Last > > Message > > 24.50.66.128 301 64dd711d-6f 00101/00001 unkn No Rx: > > SUBSCRIBE > > 24.50.66.128 (None) 8581f21c-4a 00101/00001 unkn No Rx: > > REGISTER > > 24.50.66.128 (None) 1339e59f-60 00101/00001 unkn No Rx: > > REGISTER > > 24.50.66.128 (None) f95f1c68-7d 00101/00001 unkn No Rx: > > REGISTER > > 4 active SIP channels > > > > > > support*CLI> sip show channels > > Peer User/ANR Call ID Seq (Tx/Rx) Form Hold Last > > Message > > 24.50.66.128 (None) 0e46cbb32cd 00102/00000 unkn No > Init: > > OPTIONS > > 24.50.66.128 (None) 4f4ccf9b-90 00101/00001 unkn No Rx: > > REGISTER > > 24.50.66.128 (None) 1e0b7033-12 00101/00001 unkn No Rx: > > REGISTER > > 24.50.66.128 (None) 380745da-47 00101/00001 unkn No Rx: > > REGISTER > > 4 active SIP channels > > > > > > support*CLI> sip show channels > > Peer User/ANR Call ID Seq (Tx/Rx) Form Hold Last > > Message > > 24.50.66.128 (None) 54d4194f-33 00101/00001 unkn No Rx: > > REGISTER > > 24.50.66.128 (None) 8828fe55-54 00101/00001 unkn No Rx: > > REGISTER > > 24.50.66.128 (None) 8d3cde74-8d 00101/00001 unkn No Rx: > > REGISTER > > 3 active SIP channels > > > > > > support*CLI> sip show channels > > Peer User/ANR Call ID Seq (Tx/Rx) Form Hold Last > > Message > > 24.50.66.128 (None) 66d9ce2f15c 00102/00000 unkn No > Init: > > OPTIONS > > 24.50.66.128 (None) 54d4194f-33 00101/00001 unkn No Rx: > > REGISTER > > 24.50.66.128 (None) 8828fe55-54 00101/00001 unkn No Rx: > > REGISTER > > 24.50.66.128 (None) 8d3cde74-8d 00101/00001 unkn No Rx: > > REGISTER > > 4 active SIP channels > > > > > > > > > > What the hell????? Any ideas?? I am not getting any errors or > anything > > on the CLI (verbose 100). > > > > - Gabe > > > > > > > > ----- Original Message ----- > > From: "Jerry Geis" <[EMAIL PROTECTED]> > > To: <asterisk-users@lists.digium.com> > > Sent: Monday, March 13, 2006 4:54 PM > > Subject: [Asterisk-Users] saydigits > > > > > > > I was searching on voip-info.org for saydigits. > > > I see no indication it is not valid in 1.2.4 asterisk. > > > however, when trying to use it I get and error "no application > saydigits". > > > > > > what is the correct way to echo back digits in asterisk 1.2.4? > > > > > > I tried "say digits 123" and "saydigits 123" both gave "no application " > > > error > > > > > > Thanks > > > jerry > > > _______________________________________________ > > > --Bandwidth and Colocation provided by Easynews.com -- > > > > > > Asterisk-Users mailing list > > > To UNSUBSCRIBE or update options visit: > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > _______________________________________________ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > Asterisk-Users mailing list > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
