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Ross Kevlin

2005-Jun-01 14:00 UTC

[Asterisk-Users] SNOM 360 extension lights

I recently got a SNOM 360 and have been trying to get the extension lights to
work. I can see the subscriptions with sip show subscriptions but I don't
see any notifies when a call is made. I must be missing something because
I've tried looking to see if anyone else has had this problem but the only
solutions I've seen have been to put hints in and I have those. Any
suggestions?
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Olle E. Johansson

2005-Jun-01 22:54 UTC

head link

[Asterisk-Users] SNOM 360 extension lights

Ross Kevlin wrote:> I recently got a SNOM 360 and have been trying to get the extension
> lights to work. I can see the subscriptions with sip show subscriptions
> but I don't see any notifies when a call is made. I must be missing
> something because I've tried looking to see if anyone else has had this
> problem but the only solutions I've seen have been to put hints in and
I
> have those. Any suggestions?
> There is a page on the wiki that subscribes this. If you have a proper
subscription and you have the hints, it "should" work as we usually
say
in this business when we don't know what the problem is. If you are
using CVS head, you can limit the context reachable for subscriptions
with the "subscribecontext" setting. I haven't tried with the 360
yet,
so I might have to power up that phone.

For those of you that haven't explored the subscription support in the
Asterisk SIP channel:

The key to get device state notification in SIP subscriptions and the AMI,
the Asterisk manager interface, is the use of a "hint" priority. The
phone subscribe to an extension, but in order for the PBX to know which
phone that is connected to an extension, you need to tell Asterisk what
relationship you have between an extension and a device.

exten => 3000,hint,SIP/olle
exten => 3000,dial(SIP/olle,30)

This extension in your dialplan tells Asterisk that if anyone subscribes
to extension 3000, they want to know the status of SIP/olle. Without the
hint, the extension will always be available and there's no notification
at all.

In CVS head, you can do this with IAX2 as well. I have a patch that has
been in the bugtracker for a few months that adds a bit more. If you
apply a call limit with the incominglimit, you will see that the notify
function will tell you not only if the device is available or not, but
also if they're on a call. This works beautifully with Xten's Eyebeam.

I am close to securing funding for development of another addition to
chan_sip during the summer. This will be the "shared line apperances"
solution developed by Broadsoft and now implemented in phones from many
manufacturers like Sipura, Aastra and I also suspect Grandstream and
Polycom. When we have this working, you will be able to get Asterisk
with one of these SIP phones to behave much more like a key system.

A lot of text that really did not answer the question, sorry.

/Olle

----
Astricon - the Asterisk User's conference - Madrid June 15-17
http://www.astricon.net/europe/ - Register today!

Ross Kevlin

2005-Jun-03 08:35 UTC

head link

[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@192.168.2.252>;tag=z6kvtd67bu
To: <sip:117@192.168.2.252;user=phone>;tag=as6c1cb2a5
Call-ID: 3c2670ad35b6-68nuemr6pg58@snom360
CSeq: 1 SUBSCRIBE
User-Agent: MVC 001
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: <sip:117@192.168.2.252>
Proxy-Authenticate: Digest realm="asterisk",
nonce="16747f76"
Content-Length: 0


---
Scheduling destruction of call '3c2670ad35b6-68nuemr6pg58@snom360' in
15000 ms
sip1*CLI>
<-- SIP read from 192.168.2.230:2051:
SUBSCRIBE sip:117@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@192.168.2.252>;tag=z6kvtd67bu
To: <sip:117@192.168.2.252;user=phone>
Call-ID: 3c2670ad35b6-68nuemr6pg58@snom360
CSeq: 1 SUBSCRIBE
Max-Forwards: 70
Contact: <sip:83@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@192.168.2.252>;tag=z6kvtd67bu
To: <sip:117@192.168.2.252;user=phone>;tag=as6c1cb2a5
Call-ID: 3c2670ad35b6-68nuemr6pg58@snom360
CSeq: 1 SUBSCRIBE
User-Agent: MVC 001
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: <sip:117@192.168.2.252>
Proxy-Authenticate: Digest realm="asterisk",
nonce="16747f76"
Content-Length: 0


---
Scheduling destruction of call '3c2670ad35b6-68nuemr6pg58@snom360' in
15000 ms
sip1*CLI>
<-- SIP read from 192.168.2.230:2051:
SUBSCRIBE sip:117@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@192.168.2.252>;tag=z6kvtd67bu
To: <sip:117@192.168.2.252;user=phone>
Call-ID: 3c2670ad35b6-68nuemr6pg58@snom360
CSeq: 2 SUBSCRIBE
Max-Forwards: 70
Contact: <sip:83@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@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@192.168.2.252>;tag=z6kvtd67bu
To: <sip:117@192.168.2.252;user=phone>;tag=as77c7b911
Call-ID: 3c2670ad35b6-68nuemr6pg58@snom360
CSeq: 2 SUBSCRIBE
User-Agent: MVC 001
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Expires: 3600
Contact: <sip:117@192.168.2.252>;expires=3600
Content-Length: 0


---
Scheduling destruction of call '3c2670ad35b6-68nuemr6pg58@snom360' in
3610000 ms
Reliably Transmitting (no NAT) to 192.168.2.230:2051:
NOTIFY sip:83@192.168.2.252 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.252:5060;branch=z9hG4bK56396cd7;rport
From: <sip:117@192.168.2.252;user=phone>;tag=as77c7b911
To: <sip:83@192.168.2.252>;tag=z6kvtd67bu
Contact: <sip:117@192.168.2.252>
Call-ID: 3c2670ad35b6-68nuemr6pg58@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@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@192.168.2.252;user=phone>;tag=as77c7b911
To: <sip:83@192.168.2.252>;tag=z6kvtd67bu
Call-ID: 3c2670ad35b6-68nuemr6pg58@snom360
CSeq: 102 NOTIFY
Content-Length: 0
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Ross Kevlin

2005-Jun-03 13:40 UTC

head link

[Asterisk-Users] SNOM 360 extension lights

I have a snom 360 that im trying to get the extension lights working i can
see the subscription being sent and a reply but the reply is a terminate.
Using sip show peer i can see the subscriptions but no uri. i have the hints
in place but i dont see any notifys when a line is in use.

>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
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>

Olle E. Johansson

2005-Jun-03 13:49 UTC

head link

[Asterisk-Users] SNOM 360 extension lights

Ross Kevlin wrote:>  I have a snom 360 that im trying to get the extension lights working i can
> see the subscription being sent and a reply but the reply is a terminate.Terminate being?

/O

Michael Puchol

2005-Jun-03 13:55 UTC

head link

[Asterisk-Users] Interesting article on new SIP phones

Hi all,

Just a bit of news I picked up today

http://www.theregister.co.uk/2005/06/02/computex_skype_handsets/

even though the s-word is mentioned, the handsets are also geared 
towards SIP.

Cheers,

Mike

Ross Kevlin

2005-Jun-03 14:01 UTC

head link

[Asterisk-Users] SNOM 360 extension lights

the subscription is sent a reply and the reply has content that indicates
its state is terminated

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>

from what I understand the terminated state is the end of dialog and the
subscription should end, but I still see the subscription in asterisk.
>Ross Kevlin wrote:
>  I have a snom 360 that i'm trying to get the extension lights working
I
can> see the subscription being sent and a reply but the reply is a terminate.
>Terminate being?

Ross Kevlin

2005-Jun-06 05:45 UTC

head link

[Asterisk-Users] SNOM 360 extension lights

i think this is the full packet. if it isn't, how do see the full packet

 > 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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