Context for Dial [Solved]

[nicolas]

set the "context" in your button definition to whatever you want.

[sollostech]

I specified the right context for all the extensions I configured, but Dial is just an input text area at the top of the screen.

[nicolas]

Actions will work with the configuration of the logged in extension, if you have a button:

[SIP/100]
type=extension
context=some-funky-context
extension=100

[SIP/200]
type=extension
context=my-other-tennant-context
extension=200

And you login to fop2 with extension 100, when you dial out, it will originate the call using "some-funky-context"

If you login with extension 200, it will originate using "my-other-tennant-context"

fop2 never uses explicity the "default" context. If that happens is because your context is missing in the button definition and asterisk defaults to "default" when the context is empty.

[sollostech]

Got it! Thanks for the clarification I will test my settings further.

[sollostech]

I have the context issue seemingly resolved, but now my ITSP is rejecting the call as failing to authenticate.

Its not passing my caller ID as seen below. Calling between extensions this information is properly passed.

 Executing [s@macro-ael-druid-trunkdial:71] NoOp("SIP/phone-aastra-1-0828ae58", "Finish if-ael-druid-trunkdial-946") in new stack
    -- Executing [s@macro-ael-druid-trunkdial:72] Set("SIP/phone-aastra-1-0828ae58", "exten=""") in new stack
    -- Executing [s@macro-ael-druid-trunkdial:73] Set("SIP/phone-aastra-1-0828ae58", "name=""") in new stack
    -- Executing [s@macro-ael-druid-trunkdial:74] Set("SIP/phone-aastra-1-0828ae58", "cid=""") in new stack
    -- Executing [s@macro-ael-druid-trunkdial:75] GotoIf("SIP/phone-aastra-1-0828ae58", "0?76:83") in new stack
    -- Goto (macro-ael-druid-trunkdial,s,83)
    -- Executing [s@macro-ael-druid-trunkdial:83] GotoIf("SIP/phone-aastra-1-0828ae58", "0?84:86") in new stack
    -- Goto (macro-ael-druid-trunkdial,s,86)
    -- Executing [s@macro-ael-druid-trunkdial:86] Set("SIP/phone-aastra-1-0828ae58", "options="WTKo"") in new stack
    -- Executing [s@macro-ael-druid-trunkdial:87] NoOp("SIP/phone-aastra-1-0828ae58", "Finish if-if-ael-druid-trunkdial-947-949") in new stack
    -- Executing [s@macro-ael-druid-trunkdial:88] NoOp("SIP/phone-aastra-1-0828ae58", "Finish if-ael-druid-trunkdial-947") in new stack
    -- Executing [s@macro-ael-druid-trunkdial:89] Verbose("SIP/phone-aastra-1-0828ae58", "3|- Checking if trunk is available") in new stack
    -- - Checking if trunk is available
    -- Executing [s@macro-ael-druid-trunkdial:90] ChanIsAvail("SIP/phone-aastra-1-0828ae58", "SIP/Vitelity") in new stack
    -- Executing [s@macro-ael-druid-trunkdial:91] GotoIf("SIP/phone-aastra-1-0828ae58", "0?92:99") in new stack
    -- Goto (macro-ael-druid-trunkdial,s,99)
    -- Executing [s@macro-ael-druid-trunkdial:99] Verbose("SIP/phone-aastra-1-0828ae58", "3|- Normal trunk operation") in new stack
    -- - Normal trunk operation
    -- Executing [s@macro-ael-druid-trunkdial:100] Dial("SIP/phone-aastra-1-0828ae58", "SIP/Vitelity/4048867603||WTKo") in new stack
    -- Called Vitelity/4048867603
[Apr  1 19:12:20] NOTICE[2495]: chan_sip.c:12471 handle_response_invite: Failed to authenticate on INVITE to '"asterisk" <sip:asterisk@64.2.142.93>;tag=as42b9b750'
    -- SIP/Vitelity-082d9360 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)

[nicolas]

I am sorry, I cannot debug issues with your provider. If your provider authenticates based on the callerid then it is super insecure. You cannot tweak fop2 to fix it, you have to tweak your dialplan. The origination goes to a context, extension and priority. It does not perform a Dial. you have to fix your dialplan.

Best regards,

[sollostech]

I just saw you had 2.1 out so I upgraded to that and used the new fop2cfg file. One of those fixed the problem for me as it works now. However its dialing on all the available lines on my Aastra so I need to work that out. Progress comes in steps.

[sollostech]

Restarted everything and the problem with ringing all extensions is now gone.

[thiagoalecio]

How can i change the contex of an extencion if i use auto-configuration?

[nicolas]

You can't change the context unless you modify the autoconfig-buttons-freepbx.sh script. By default it uses the context defined in the sip peer definition.