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sail to sail config problems lost extensions

Offline rrkelly

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sail to sail config problems lost extensions
« on: March 04, 2008, 09:49:38 AM »
after setting up a sibling trunk i have lost the ability to call between extensions and the sibling trunk only half/somewhat  works.
*56* does not work  nor can i dial from 5000 to 5003 or what not . The local extensions appear OK  every thing is registered and i have restarted asterisk several times to make sure every thing has been reread. -- i can however make a call out via my sip provider just fine. (the magic _ works for dail plans)

I can route over IAX2 to the other sail PBX but when it gets there instead of ringing an extension it tries to call out.  I guess the biggest problem is i don't understand sail's nomenclature or how it does certain things.   How is a ring group attached to an incoming
route?  how are in and out bound routes differentiated? What is an open inbound route as applied to a trunk?
thank you
rob

Offline SARK devs

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Re: sail to sail config problems lost extensions
« Reply #1 on: March 04, 2008, 12:57:59 PM »
Can we see what you have in your route dial plans please (particularly for the IAX link).

Regards

S

Offline rrkelly

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Re: sail to sail config problems lost extensions
« Reply #2 on: March 04, 2008, 01:16:02 PM »
route dial plans

_2XXX from me to canada his extn's are 2000- 2005 (my route)

_50XX from canada to me my extens are 5000 5003 (micheals route)

primary path  in both cases is billsgwsme  active = yes  and the rest blank


left panel of the sibling trunk

type=peer
host=watchorn.homeip.net
qualify=3000
canreinvite=no
username=billsgwsme
fromuser=billsgwsme
secret=asterisk
trunk=yes
disallow=all
allow=alaw
allow=ulaw

right panel

type=user
secret=&&&&&&&&   (real secret replaced)
context=internal

what should Open Inbound Route: be set to?






Offline rrkelly

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Re: sail to sail config problems lost extensions
« Reply #3 on: March 04, 2008, 01:39:25 PM »
cli in canada  -- Accepting AUTHENTICATED call from 24.197.201.104:
       > requested format = alaw,
       > requested prefs = (alaw|ulaw),
       > actual format = alaw,
       > host prefs = (),
       > priority = caller
    -- Executing [2001@internal:1] AGI("IAX2/billsgwsme-3", "selintra|OutCluster|2001") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/selintra
    -- AGI Script selintra completed, returning 0
    -- Executing [2001@default:1] AGI("IAX2/billsgwsme-3", "selintra|OutRoute|MAIN") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/selintra
    -- AGI Script Executing Application: (Dial) Options: (SIP/2001@VitelityOutbound)
    -- Called 2001@VitelityOutbound
    -- Got SIP response 484 "Address Incomplete" back from 64.2.142.22

my cli making the call

   -- Executing [2001@internal:1] AGI("SIP/5000-0839ece0", "selintra|OutCluster|2001") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/selintra
    -- AGI Script selintra completed, returning 0
    -- Executing [2001@default:1] AGI("SIP/5000-0839ece0", "selintra|OutRoute|mag2intouch") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/selintra
    -- AGI Script Executing Application: (Dial) Options: (IAX2/billsgwsme@billsgwsme/2001)
    -- Called billsgwsme@billsgwsme/2001
    -- Call accepted by 24.226.63.103 (format alaw)
    -- Format for call is alaw
    -- IAX2/billsgwsme-16384 answered SIP/5000-0839ece0
    -- Hungup 'IAX2/billsgwsme-16384'
    -- AGI Script selintra completed, returning 0
  == Auto fallthrough, channel 'SIP/5000-0839ece0' status is 'ANSWER'
    -- Executing [h@default:1] Hangup("SIP/5000-0839ece0", "") in new stack
  == Spawn extension (default, h, 1) exited non-zero on 'SIP/5000-0839ece0'






Offline SARK devs

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Re: sail to sail config problems lost extensions
« Reply #4 on: March 04, 2008, 04:48:47 PM »
OK - first things first..  There is no inbound route on a privileged IAX2 sibling call.  This is becasue it is accepted on the internal context (i.e. the same context that handles extensions).  Thus an extension to extension call will dial straight to the phone.  It also means that you can, if you wish, have the sibling server handle your outbound calls using its trunks if you wish.

Here is a clean inbound IAX2 sibling call to extension 6002 


Code: [Select]
   -- Accepting AUTHENTICATED call from 194.164.246.30:
       > requested format = alaw,
       > requested prefs = (alaw|ulaw),
       > actual format = alaw,
       > host prefs = (),
       > priority = caller
    -- Executing [6002@internal:1] AGI("IAX2/mtukswitch-26", "selintra|OutCluster|6002") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/selintra
    -- AGI Script selintra completed, returning 0
    -- Executing [6002@default:1] AGI("IAX2/mtukswitch-26", "selintra|InCall|") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/selintra
    -- AGI Script Executing Application: (Dial) Options: (SIP/6002|10|tT)
    -- Called 6002
    -- SIP/6002-0917dfb8 is ringing
    -- SIP/6002-0917dfb8 is ringing
    -- SIP/6002-0917dfb8 is ringing
    -- SIP/6002-0917dfb8 is ringing
    -- SIP/6002-0917dfb8 answered IAX2/mtukswitch-26

You'll notice that your friend's system doesn't process selintra|InCall.  Instead it processes selintra|OutRoute|MAIN.

This tells me that he almost ceratainly has a route (MAIN) with _X. in the dial plan (or _XX.)  and it is catching everything and throwing it out the SIP channel.

I suspect the same is true of your system if *56* and extension dials don't work.


Kind Regards

S

 
 

« Last Edit: March 04, 2008, 04:50:32 PM by selintra »