[Asterisk-Users] Woooo, Polycom and * on crack - can't register!
Gabriel Afana
asterisk at gafana.com
Mon Mar 13 18:56:29 MST 2006
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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