[Asterisk-Users] Re: SNOM and 1.0.9

[Joseph Rothstein]

I still cannot get this to work on 1.0.9.

I am trying to test with two extensions:

Here is the config I am using:

exten => 451,hint,sip/451
exten => 451,1,Dial(SIP/451,20,tr)
exten => 451,2,Voicemail(b451 at other)
exten => 451,102,Voicemail(u451 at other)

exten => 453,hint,sip/453
exten => 453,1,Dial(SIP/453,20,tr)
exten => 453,2,Voicemail(b453 at other)
exten => 453,102,Voicemail(u453 at other)

On the SNOM, the SIP trace shows the initial subscription:

NOTIFY sip:320 at xxx.xxx.xxx.xxx SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK13ea176f
From: <sip:453 at xxx.xxx.xxx.xxx;user=phone>;tag=as77402d3b
To: <sip:320 at xxx.xxx.xxx.xxx>;tag=c0av8f2x4v
Contact: <sip:453 at xxx.xxx.xxx.xxx>
Call-ID: 3c26700c30d4-libo7sf1rff7 at snom320
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
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:320 at xxx.xxx.xxx.xxx">
<dialog id="453">
<state>terminated</state>
</dialog>
</dialog-info>

The SNOM shows the light off for this extension. This is a hardphone, and is
always registered.

NOTIFY sip:320 at 195.27.242.8 SIP/2.0
Via: SIP/2.0/UDP 195.27.242.8:5060;branch=z9hG4bK258fb569
From: <sip:451 at 195.27.242.8;user=phone>;tag=as26ba79ca
To: <sip:320 at 195.27.242.8>;tag=8ioo4i3sp7
Contact: <sip:451 at 195.27.242.8>
Call-ID: 3c26700c2bf2-wfqpeg34g7az at snom320
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Event: dialog
Content-Type: application/dialog-info+xml
Content-Length: 202

<?xml version="1.0"?>
<dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="0"
state="full" entity="sip:320 at 195.27.242.8">
<dialog id="451">
<state>confirmed</state>
</dialog>
</dialog-info>

This is a softphone that is not registered, and the light on the keyboard is
on.

Light is one unavailable, light is off available.

When I make a call from extension 453, and am on the phone, nothing is sent
to the SNOM. I see no SIP packets leaving Asterisk either.

This is what Asterisk shows:

asterisk_test*CLI> sip show subscriptions
Peer             User        Call ID                URI
195.27.242.113   320         3c26700c30d4-libo7sf1
195.27.242.113   320         3c26700c2bf2-wfqpeg34
0 active SIP subscriptions(s)
asterisk_test*CLI>

If anyone has any additional ideas, or a snippet of config that works,
please post it.

I will try to upgrade to 1.2 and see how this works.

Thanks,
Joe