John C. Wolosuk Jr.
2007-Apr-25 08:47 UTC
[asterisk-users] 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 -------------------------------------------
Stephen Bosch
2007-Apr-25 09:54 UTC
[asterisk-users] SLA Appearance between 2 Cisco 7960's (SIP)
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-
Russell Bryant
2007-Apr-25 13:01 UTC
[asterisk-users] SLA Appearance between 2 Cisco 7960's (SIP)
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.
Jason Howk
2007-Apr-25 16:32 UTC
[asterisk-users] SLA Appearance between 2 Cisco 7960's (SIP)
>From reading the SLA docs, SIP hints are use to get the lights on thephone 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! >