[asterisk-users] 1.4 - SLA
Bill Gibbs
bgibbs at edurotech.com
Mon Mar 5 12:33:54 MST 2007
Here is the debug output of the SUBSCRIBE request
I am sure it has something to do with the way I am attempting to setup
the Polycom for shared appearances...
Nat=yes is set in the peer. I don't get these weird messages when
connecting with a "private" line appearance.
<--- SIP read from x.x.x.x:60671 --->
SUBSCRIBE sip:103 at x.x.x.x:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.116;branch=z9hG4bKf2bac598B416612D
From: "Line 1" <sip:station1_line1 at x.x.x.x>;tag=259528A1-76B251C6
To: <sip:103 at x.x.x.x>
CSeq: 1 SUBSCRIBE
Call-ID: 180a0f12-40e98447-433cc5cc at 192.168.1.116
Contact: <sip:2404366402-2 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
<------------->
[Mar 5 14:25:02] VERBOSE[9835] logger.c: --- (13 headers 0 lines) ---
[Mar 5 14:25:02] VERBOSE[9835] logger.c: Sending to 192.168.1.116 :
5060 (no NAT)
[Mar 5 14:25:02] VERBOSE[9835] logger.c: Found no matching peer or user
for 'x.x.x.x:60671'
[Mar 5 14:25:02] VERBOSE[9835] logger.c: Looking for 103 in default
(domain x.x.x.x)
[Mar 5 14:25:02] VERBOSE[9835] logger.c:
<--- Transmitting (no NAT) to 192.168.1.116:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP
192.168.1.116;branch=z9hG4bKf2bac598B416612D;received=x.x.x.x
From: "Line 1" <sip:station1_line1 at x.x.x.x>;tag=259528A1-76B251C6
To: <sip:103 at x.x.x.x>;tag=as4d77da56
Call-ID: 180a0f12-40e98447-433cc5cc at 192.168.1.116
CSeq: 1 SUBSCRIBE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Bill Gibbs
Sent: Monday, March 05, 2007 1:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] 1.4 - SLA
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.
I just started playing with it this morning however...still playing
around w/ the configs.
One odd thing, I keep seeing some weirdness:
[Mar 5 13:10:52] VERBOSE[9381] logger.c: Looking for 105 in default
(domain x.x.x.x)
And also Looking for 103
Yet I have no idea where those values are coming from!
I am running 1.6.7.
Here is a snippet of the phone config from one of the phones:
<reg reg.1.displayName="Line 1" reg.1.address="station2_line1"
reg.1.label="Line 1" reg.1.type="shared"
reg.1.thirdPartyName="2404366402-2" reg.1.auth.userId="2404366402-2"
reg.1.auth.password="1234" reg.1.server.1.address=""
reg.1.server.1.port="" reg.1.server.1.transport="DNSnaptr"
reg.1.server.2.transport="DNSnaptr" reg.1.server.1.expires="60"
reg.1.server.1.register="1
I noticed that I had to set the reg.x.address field to the
stationX_lineX value or the phone wouldn't fill in the "icon"
image...but it would accept cals. Still not completely clear but I am
making progress!
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: Friday, March 02, 2007 6:58 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] 1.4 - SLA
Lacy Moore - Aspendora wrote:
> Russell, I don't have any specifics at this time. I need to dig a
> little further. I'm thinking the autocontext is what is giving me
> fits. I can receive calls and place calls, but the hint status is not
> working. It currently registers as a hint showing not in use. It
> does not show in use.
If you aren't seeing any lights change on the phones when calls are
going on, check "sip show subscriptions" at the CLI. If the phones have
not properly subscribed to the right extensions, you won't see anything.
> I ended up using some of the config from the bottom of the sla.txt
> file. The sample file may be missing the template section. The
> sample config does not match the config in the sla.txt. I couldn't
> get the sample config to work at all. Again, hopefully over the
> weekend I'll be able to get more information.
You are correct. The sample configuration is missing the template. I
will add it now. However, I just made the tarballs for 1.4.1, so this
config fix didn't make it in.
> Using the config in the sample file, the hint status was working. I
> could see the line ringing, but I could not answer the lines or place
> calls. Using the config from the sla.txt file, I could place calls
> and receive calls, but the hints never showed any activity, just
> always not in use.
As I noted earlier, check your "sip show subscriptions" to make sure the
phones are subscribed to the right thing.
Another helpful thing that you can use for debugging is to look at the
output of "sla show stations". You can see the state of each line
appearance on each station. This should correspond with what you see on
the phone ... unless there is a problem, of course.
> If possible, could you provide the config that you've used for
> testing? I'm testing using Polycom phones to try to keep things
> simple. I'm assuming you are using a Polycom.
I have been testing with a variety of different phones. I have not
tested all of the Polycom models, yet. This is one of the things we're
going to have to work through. I would like to document issues with
specific phones in sla.txt as we come across them.
The configuration I'm using for testing looks just like the stuff in
configs/sla.conf.sample. Essentially, it is:
[line1]
type=trunk
device=Zap/3
autocontext=line1
[line2]
type=trunk
device=Zap/4
autocontext=line2
[station](!)
type=station
autocontext=sla_stations
trunk=line1
trunk=line2
[station1] (station)
device=SIP/station1
[station2](station)
device=SIP/station2
[station3](station)
device=SIP/station3
Thanks for providing some feedback on this. You are the first one to
say anything about it. :) I am very eager to get everything working
well so that everyone is happy. Just please be patient as I work
through the initial flood of reports since it is just now getting out in
the field.
--
Russell Bryant
Software Engineer
Digium, Inc.
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