[asterisk-users] SLA Appearance between 2 Cisco 7960's (SIP)

Aaron Daniel aarond at digium.com
Wed Apr 25 13:08:38 MST 2007



Russell Bryant wrote:
> 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 at 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 at 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.
> 

Just an FYI, Cisco phones running SIP do *not* do shared line 
appearances, on *any* system.

-- 
Aaron Daniel
Community Relations Specialist
aarond at digium.com
(256) 428-6010


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