[Asterisk-Users] Woooo, Polycom and * on crack - can'tregister!
Gabriel Afana
asterisk at gafana.com
Mon Mar 13 21:13:51 MST 2006
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" <asterisk at gafana.com>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at 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:303 at 216.152.244.70>;tag=2CB88E3F-E17FB2BC
> To: <sip:303 at 216.152.244.70>;tag=as743ca65c
> Call-ID: 287890e3-97211381-efdf49ce at 192.168.1.100
> CSeq: 1 REGISTER
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
> Max-Forwards: 70
> Contact: <sip:303 at 216.152.244.70>
> 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" <asterisk at gafana.com>
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
> <asterisk-users at 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" <geisj at pagestation.com>
> > To: <asterisk-users at 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
> > > _______________________________________________
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> >
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