asterisk users - Apr 2007 - SLA Appearance between 2 Cisco 7960's (SIP)

Has anyone had any success with getting SLA going between 2 SIP phones? 
(Particularly a set of Cisco 79xx's) The SLA document that comes with 
the asterisk source is about as clear as mud.

Does anyone have a working sip.conf, sla.conf, and extensions.conf that 
I can use for reference?

The part I'm most confused about is how to build the lines in sip.conf 
and how the phones should behave. It seems apparent that the phones 
should not register with asterisk, otherwise all the phones will try to 
register to be THE phone for a given extension. should these lines be 
built like a trunk/peer? if I could be an example of how lines for SLA 
should look in sip.conf, that would be helpful.

Also I'm somewhat annoyed that I have to compile zaptel drivers that I 
don't use in order to compile the app_meetme.so module so I can have the 
SLA functions available to the dialplan...

Any feedback is greatly appreciated!

-- 
-------------------------------------------
John C. Wolosuk Jr.
Unix/Linux Systems Administrator
Academic Computing & Communications Center
University of Illinois @ Chicago

E-Mail: jwolosuk at uic dot edu
-------------------------------------------

John C. Wolosuk Jr. wrote:
> Also I'm somewhat annoyed that I have to compile zaptel drivers that I
> don't use in order to compile the app_meetme.so module so I can have
the
> SLA functions available to the dialplan...
If you're using SLA, you're using zaptel drivers, yes -- without the
timing source from ztdummy your SLA won't work (that's really what SLA
is -- a fancy Meetme conference).

Are you having trouble compiling Zaptel?

-Stephen-

John C. Wolosuk Jr. wrote:
> Has anyone had any success with getting SLA going between 2 SIP phones? 
> (Particularly a set of Cisco 79xx's) The SLA document that comes with 
> the asterisk source is about as clear as mud.
Mud, huh?  I guess I should work on that at some point, then ...

You say two phones.  What do you intend to use on the trunk side?  I 
assume you want a SIP trunk.
> Does anyone have a working sip.conf, sla.conf, and extensions.conf that 
> I can use for reference?
sip.conf:

This is configured just like any other SIP device.  In your scenario of 
two SIP phones and one SIP trunk, sip.conf would contain three entries. 
  For example:

[station1]
type=friend
secret=station1
host=dynamic
context=sla_stations
dtmfmode=rfc2833
disallow=all
allow=ulaw

[station2]
type=friend
secret=station2
host=dynamic
context=sla_stations
dtmfmode=rfc2833
disallow=all
allow=ulaw

[providerA]
type=friend
secret=something
host=providerA.com
context=line1
dtmfmode=rfc2833
disallow=all
allow=ulaw


sla.conf: (From sla.pdf, page 7)

Here you create a definition for a single line and two stations.

[line1]
type=trunk
device=Local/disa@line1_outbound

[station](!)
type=station
trunk=line1

[station1](station)
device=SIP/station1

[station2](station)
device=SIP/station2


extensions.conf:

[line1]
; This is used for incoming calls from SIP/providerA because providerA
; has context=line1 in sip.conf.  Incoming calls immediately go into the
; SLATrunk application.  Then, the appropriate stations will start
; ringing.
exten => s,1,SLATrunk(line1)

[line1_outbound]
; This context is used by the SLA code.  line1 in sla.conf was
; configured to use a device called Local/disa@line1_outbound.
; That means that when someone presses the line button for line1,
; it will get connected to Disa.  Disa will provide dialtone and
; allow the caller to dial any other extensions that live in this
; context.  In this case, there is only one available pattern.  When
; it gets dialed, the call goes out to SIP/providerA.
exten => disa,1,Disa(no-password|line1_outbound)
exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@providerA)

[sla_stations]
; These extensions are  called by the stations .

; This extension should be called when the the phone for
; SIP/station1 is taken off hook without pressing a line button.
exten => station1,1,SLAStation(station1)
  ; This extension should be called when the user presses the
; line1 key on the phone.
exten => station1_line1,1,SLAStation(station1_line1)
; The line1 key on the phone for station1 should be configured
; to subscribe to the state of the extension "station1_line1".
; This will allow Asterisk to control the light to make it turn
; on, off, or blink, as appropriate.
exten => station1_line1,hint,SLA:station1_line1

exten => station2,1,SLAStation(station2)
exten => station2_line1,hint,SLA:station2_line1
exten => station2_line1,1,SLAStation(station2_line1)
> The part I'm most confused about is how to build the lines in sip.conf 
> and how the phones should behave. It seems apparent that the phones 
> should not register with asterisk, otherwise all the phones will try to 
> register to be THE phone for a given extension. should these lines be 
> built like a trunk/peer? if I could be an example of how lines for SLA 
> should look in sip.conf, that would be helpful.
Actually, the phones *do* register to Asterisk.  But, the line 
appearance buttons themselves are not registrations to Asterisk.  They 
are simply subscribers to the state of extensions.  You set these up 
just like you would for any other hint in Asterisk.
> Also I'm somewhat annoyed that I have to compile zaptel drivers that I 
> don't use in order to compile the app_meetme.so module so I can have
the
> SLA functions available to the dialplan...
It uses MeetMe internally, and MeetMe requires Zaptel.  That's just the 
way it is.

-- 
Russell Bryant
Software Engineer
Digium, Inc.

>From reading the SLA docs, SIP hints are use to get the lights on the
phone to show the "correct state".  I was under the impression that
the
SIP firmware on the 7960's didn't support the SIP hints properly (or at
all), which means that SLA won't work properly on a 7960.

If anyone has gotten this to work, I'd like to hear about it.

--Jason.

John C. Wolosuk Jr. wrote:
> Has anyone had any success with getting SLA going between 2 SIP phones?
> (Particularly a set of Cisco 79xx's) The SLA document that comes with
> the asterisk source is about as clear as mud.
> 
> Does anyone have a working sip.conf, sla.conf, and extensions.conf that
> I can use for reference?
> 
> The part I'm most confused about is how to build the lines in sip.conf
> and how the phones should behave. It seems apparent that the phones
> should not register with asterisk, otherwise all the phones will try to
> register to be THE phone for a given extension. should these lines be
> built like a trunk/peer? if I could be an example of how lines for SLA
> should look in sip.conf, that would be helpful.
> 
> Also I'm somewhat annoyed that I have to compile zaptel drivers that I
> don't use in order to compile the app_meetme.so module so I can have
the
> SLA functions available to the dialplan...
> 
> Any feedback is greatly appreciated!
>