[Asterisk-Users] SNOM 360 extension lights

Fri Jun 3 10:49:36 MST 2005

Sorry Ross I must have missed your first postings, but what are you
trying to achive?

David

On 03/06/05, Ross Kevlin <RAKevlin at metrostat.net> wrote:
>  
> 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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