[asterisk-users] 1.4 - SLA

Tue Mar 6 12:09:41 MST 2007

I think it has something to do with hints...I can't seem to subscribe to
anything now with 1.4 vs 1.2, even with a normal non SLA setup.

My phone/config that works with 1.2, so I know hints work with the phone
and firmware and with NAT at least on 1.2.

I did a fresh 1.4 install (and I did a "make samples" so I had something
to work off of)

"sip show subscriptions" shows 0 active

"show hints":
2404366402 at default             : SIP/2404366402        State:Idle
Watchers  0

If I run the default demo app, show hints still shows Idle.

My "Buddies" key in the Polycom, which is watching the proper sip hint
(works in 1.2) shows the extension to be Offline.

Sip.conf
[general]
allowsubscribe=yes
subscribecontext=default
notifyringing=yes
notifyhold=yes
limitonpeers=yes
(I tried with and without the above values commented out, as well as
specifically in my device peer definition)

[2404366402]
type=friend
secret=blahededah
nat=yes
host=dynamic
canreinvite=no
context=default
qualify=yes

extensions.conf
[default]
exten => 2404366402,hint,SIP/2404366402
...etc...

My mac-directory.xml 
..snip...
                <item>
                        <ln>myself</ln>
                        <fn></fn>
                        <ct>2404366402</ct>
                        <sd></sd>
                        <rt></rt>
                        <dc/>
                        <ad>0</ad>
                        <ar>0</ar>
                        <bw>1</bw>
                        <bb>0</bb>
                </item>
...snip...
I also tried in the <ct>2404366402 at default</ct>

Let's pretend 1.1.1.1 is my firewall that the Polycom is behind
2.2.2.2 is my 1.4.1 test Asterisk server

<--- SIP read from 1.1.1.1:60671 --->
SUBSCRIBE sip:2404366402 at 2.2.2.2:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.116;branch=z9hG4bK256e271aEC1CEA7B
From: "Line 1" <sip:2404366402 at 2.2.2.2>;tag=447AE7-653FB66C
To: <sip:2404366402 at 2.2.2.2>
CSeq: 1 SUBSCRIBE
Call-ID: 5b934a10-ae1d4a69-d503e556 at 192.168.1.116
Contact: <sip:2404366402 at 192.168.1.116>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE,
NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_430-UA/1.6.7.0094
Max-Forwards: 70
Expires: 3600
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
Creating new subscription
Sending to 192.168.1.116 : 5060 (no NAT)
Found peer '2404366402'

<--- Transmitting (NAT) to 1.1.1.1:60671 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
192.168.1.116;branch=z9hG4bK256e271aEC1CEA7B;received=1.1.1.1
From: "Line 1" <sip:2404366402 at 2.2.2.2>;tag=447AE7-653FB66C
To: <sip:2404366402 at 2.2.2.2>;tag=as3123a96d
Call-ID: 5b934a10-ae1d4a69-d503e556 at 192.168.1.116
CSeq: 1 SUBSCRIBE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk",
nonce="7a544b2b"
Content-Length: 0

<--- SIP read from 1.1.1.1:60671 --->
SUBSCRIBE sip:2404366402 at 2.2.2.2:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.116;branch=z9hG4bK3f8d777548EC8ED2
From: "Line 1" <sip:2404366402 at 2.2.2.2>;tag=447AE7-653FB66C
To: <sip:2404366402 at 2.2.2.2>
CSeq: 2 SUBSCRIBE
Call-ID: 5b934a10-ae1d4a69-d503e556 at 192.168.1.116
Contact: <sip:2404366402 at 192.168.1.116>
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE,
NOTIFY, PRACK, UPDATE, REFER
Event: presence
User-Agent: PolycomSoundPointIP-SPIP_430-UA/1.6.7.0094
Authorization: Digest username="2404366402", realm="asterisk",
nonce="7a544b2b", uri="sip:2404366402 at 2.2.2.2:5060",
response="404b224f5abbdc3793d4df45ee2ffa59", algorithm=MD5
Max-Forwards: 70
Expires: 3600
Content-Length: 0

<------------->
--- (14 headers 0 lines) ---
Creating new subscription
Sending to 1.1.1.1 : 60671 (NAT)
Found peer '2404366402'
Looking for 2404366402 in default (domain 2.2.2.2)

<--- Transmitting (NAT) to 1.1.1.1:60671 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP
192.168.1.116;branch=z9hG4bK3f8d777548EC8ED2;received=1.1.1.1
From: "Line 1" <sip:2404366402 at 2.2.2.2>;tag=447AE7-653FB66C
To: <sip:2404366402 at 2.2.2.2>;tag=as3123a96d
Call-ID: 5b934a10-ae1d4a69-d503e556 at 192.168.1.116
CSeq: 2 SUBSCRIBE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

Bill

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Russell
Bryant
Sent: Tuesday, March 06, 2007 1:35 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] 1.4 - SLA

Bill Gibbs wrote:
> I have been using 2 Polycom 430s so far.  I can get incoming calls
just
> fine (both phones ring on line 1).  However it doesn't appear to seize
> the line, so if a call is on the one phone, I can still pick up line 1
> on the other and dial - and it's reflected in the connected call.  I
> assume that's related to the hint/subscription issue Lacy indicated as
> well.  "sip show subscriptions" shows nothing.

If you see no subscriptions, then the phones will not dispaly the state 
of the line at all.

In regards to still allowing you to dial when all lines are busy, do you

have your phones set up to automatically dial when you go off-hook?  In 
this SLA setup, you should not allow any dialing on the phone before a 
call is made.  If the phone is taken off hook without pressing a 
specific line button, the phone should immediately dial the "station1" 
(or whatever the station is named) extension.  This will connect the 
station to the first available trunk if there is one, and then provide 
dialtone for making a call.

-- 
Russell Bryant
Software Engineer
Digium, Inc.
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