[Asterisk-Users] SNOM 360 extension lights
Matias G.
listas_ast at reliable.com.ar
Fri Jun 3 09:45:07 MST 2005
http://www.voip-info.org/tiki-index.php?page=Asterisk%20phone%20snom
remember always to search the wiki it has tons of info...
I'm using a 360 and the lights work fine. Pay attention to the hint stuff in the dialplan.
bye,
M.
----- Original Message -----
From: Ross Kevlin
To: asterisk-users at lists.digium.com
Sent: Friday, June 03, 2005 12:35 PM
Subject: [Asterisk-Users] SNOM 360 extension lights
I contacted SNOM and they told me to change a couple of options but still no lights, here is what they told me
Line page SIP tab:
o Long SIP-Contact (RFC3840) to "off"
o Support broken Registrar to "on"
Advanced page:
o Filter Packets from Registrar to "off"
And please ask the Asterisk community for help, I'm sure they solved that
issue 100%, and we are not knowing so much about Asterisk.
Your snom support Team
has anyone gotten a 360 to work with the lights? what options and modifications to .conf files did you have to make?
here are the subscribe and notifies.
it seems it terminates the subscription as soon as its created. I don't think its a proxy authentication problem
because it eventually sends the proxy authentication information
Using latest SUBSCRIBE request as basis request
Sending to 192.168.2.230 : 2051 (non-NAT)
Found peer '83'
Transmitting (no NAT) to 192.168.2.230:2051:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.2.230:2051;branch=z9hG4bK-eu4b88eh049n
From: <sip:83 at 192.168.2.252>;tag=z6kvtd67bu
To: <sip:117 at 192.168.2.252;user=phone>;tag=as6c1cb2a5
Call-ID: 3c2670ad35b6-68nuemr6pg58 at snom360
CSeq: 1 SUBSCRIBE
User-Agent: MVC 001
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: <sip:117 at 192.168.2.252>
Proxy-Authenticate: Digest realm="asterisk", nonce="16747f76"
Content-Length: 0
---
Scheduling destruction of call '3c2670ad35b6-68nuemr6pg58 at snom360' in 15000 ms
sip1*CLI>
<-- SIP read from 192.168.2.230:2051:
SUBSCRIBE sip:117 at 192.168.2.252;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.2.230:2051;branch=z9hG4bK-eu4b88eh049n;rport
From: <sip:83 at 192.168.2.252>;tag=z6kvtd67bu
To: <sip:117 at 192.168.2.252;user=phone>
Call-ID: 3c2670ad35b6-68nuemr6pg58 at snom360
CSeq: 1 SUBSCRIBE
Max-Forwards: 70
Contact: <sip:83 at 192.168.2.230:2051;line=kcx1qlml>
Event: dialog
Accept: application/dialog-info+xml
Expires: 3600
Content-Length: 0
--- (12 headers 0 lines)---
Ignoring this SUBSCRIBE request
Found peer '83'
Transmitting (no NAT) to 192.168.2.230:2051:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.2.230:2051;branch=z9hG4bK-eu4b88eh049n
From: <sip:83 at 192.168.2.252>;tag=z6kvtd67bu
To: <sip:117 at 192.168.2.252;user=phone>;tag=as6c1cb2a5
Call-ID: 3c2670ad35b6-68nuemr6pg58 at snom360
CSeq: 1 SUBSCRIBE
User-Agent: MVC 001
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: <sip:117 at 192.168.2.252>
Proxy-Authenticate: Digest realm="asterisk", nonce="16747f76"
Content-Length: 0
---
Scheduling destruction of call '3c2670ad35b6-68nuemr6pg58 at snom360' in 15000 ms
sip1*CLI>
<-- SIP read from 192.168.2.230:2051:
SUBSCRIBE sip:117 at 192.168.2.252;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.2.230:2051;branch=z9hG4bK-otquirkrmu7x;rport
From: <sip:83 at 192.168.2.252>;tag=z6kvtd67bu
To: <sip:117 at 192.168.2.252;user=phone>
Call-ID: 3c2670ad35b6-68nuemr6pg58 at snom360
CSeq: 2 SUBSCRIBE
Max-Forwards: 70
Contact: <sip:83 at 192.168.2.230:2051;line=kcx1qlml>
Event: dialog
Accept: application/dialog-info+xml
Proxy-Authorization: Digest username="83",realm="asterisk",nonce="16747f76",uri=
"sip:117 at 192.168.2.252;user=phone",response="15d72104244317e2c0afa3499220e4ab",a
lgorithm=md5
Expires: 3600
Content-Length: 0
--- (13 headers 0 lines)---
Using latest SUBSCRIBE request as basis request
Sending to 192.168.2.230 : 2051 (non-NAT)
Found peer '83'
Looking for 117 in localusers-C2021-1
Transmitting (no NAT) to 192.168.2.230:2051:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.230:2051;branch=z9hG4bK-otquirkrmu7x
From: <sip:83 at 192.168.2.252>;tag=z6kvtd67bu
To: <sip:117 at 192.168.2.252;user=phone>;tag=as77c7b911
Call-ID: 3c2670ad35b6-68nuemr6pg58 at snom360
CSeq: 2 SUBSCRIBE
User-Agent: MVC 001
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Expires: 3600
Contact: <sip:117 at 192.168.2.252>;expires=3600
Content-Length: 0
---
Scheduling destruction of call '3c2670ad35b6-68nuemr6pg58 at snom360' in 3610000 ms
Reliably Transmitting (no NAT) to 192.168.2.230:2051:
NOTIFY sip:83 at 192.168.2.252 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.252:5060;branch=z9hG4bK56396cd7;rport
From: <sip:117 at 192.168.2.252;user=phone>;tag=as77c7b911
To: <sip:83 at 192.168.2.252>;tag=z6kvtd67bu
Contact: <sip:117 at 192.168.2.252>
Call-ID: 3c2670ad35b6-68nuemr6pg58 at snom360
CSeq: 102 NOTIFY
User-Agent: MVC 001
Event: dialog
Content-Type: application/dialog-info+xml
Content-Length: 203
<?xml version="1.0"?>
<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="0" state="full"
entity="sip:83 at 192.168.2.252">
<dialog id="117">
<state>terminated</state>
</dialog>
</dialog-info>
---
sip1*CLI>
<-- SIP read from 192.168.2.230:2051:
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.2.252:5060;branch=z9hG4bK56396cd7;rport=5060
From: <sip:117 at 192.168.2.252;user=phone>;tag=as77c7b911
To: <sip:83 at 192.168.2.252>;tag=z6kvtd67bu
Call-ID: 3c2670ad35b6-68nuemr6pg58 at snom360
CSeq: 102 NOTIFY
Content-Length: 0
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