Sending calls to another Asterisk machine

eijob

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#1
Folks,

Im in a pickle, this is the last portion of my elastix deployment.

I have 1500 DID's. My telco is sending the last 4 digits of the US number. a portion of that DID block should be sent to my remote office that has an asterisk box.

Inter-office calling via extension is working.

When my PBX receives any number from 3900 to 3999 that should go to my remote office.

Im trying to create that dial plan thru "inbound routes". Am i missing anything?

I need the caller ID preserved as well when the call is transferred to the asterisk machine.

Thanks for your assistance.
eijob
 

dicko

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#2
Inter office trunking should be in the context from-internal.

The outbound route that matches the far end extensions should be marked as "intra company" to preserve the CID and use that trunking.

Given that, the "inbound route" that matches is maybe over complicated and unnecessary, those defined "far-end office" destinations should be resolved within the from-internal context.

And I very much doubt that this is the "last portion" of your deployment :)

dicko
 

eijob

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#3
Hi Dicko,

Thanks for your speedy response.

How do I mark that as "intra company"?

I dont see that when I do "show dialplan" inside the CLI.

I appended my extensions_custom.conf and added the following lines to make it work.

[from-pstn-custom]
exten => _39XX,1,Dial(IAX2/pbx02/${EXTEN})
exten => _39XX,n,Goto(from-trunk,${EXTEN},1)

You are right with the last portion, enhancement requests will definitely follow after. :woohoo:

Is there an easier way to do this from freepbx?

Cheers!
Boj
 

dicko

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#4
Yes I can see how that would be a little complicated,

in the "outbound route" page if you look closely there is I believe a box that says (I included the hover-over help for completeness) :-

"Intra Company Route"

Optional: Selecting this option will treat this route as a intra-company connection, preserving the internal Caller ID information and not use the outbound CID of either the extension or trunk.

how about you put a check mark against that?

as to your

[from-pstn-custom]

as I said that might be overcomplicated, try it my way (the from-internal context for the trunk thingy) first maybe?

dicko
 

eijob

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#5
I removed the lines in from-pstn-custom and reloaded asterisk

I had that checked on my Outgoing routes that goes to the other PBX and have these in the dial pattern. Thats how I got the internal calling to work. But if the DID is coming from the PSTN (SIP connect from my carrier) it does not send the call to the other asterisk box. But when I added the lines on my earlier post the calls got routed from PSTN to my 2nd asterisk box.

2712
4357
87730
87740
87750
87760
239[023456789]
23[012345678]X
274[012]
32XX
33XX
38XX
39XX
90[012345]X

This is what I get from the CLI:


Verbosity is at least 3
-- Executing [3926@from-trunk:1] Set("SIP/Wiline_SIP-00000022", "__FROM_DID=3926") in new stack
-- Executing [3926@from-trunk:2] NoOp("SIP/Wiline_SIP-00000022", "Received an unknown call with DID set to 3926") in new stack
-- Executing [3926@from-trunk:3] Goto("SIP/Wiline_SIP-00000022", "s|a2") in new stack
-- Goto (from-trunk,s,2)
-- Executing [s@from-trunk:2] Answer("SIP/Wiline_SIP-00000022", "") in new stack
-- Executing [s@from-trunk:3] Wait("SIP/Wiline_SIP-00000022", "2") in new stack
-- Executing [s@from-trunk:4] Playback("SIP/Wiline_SIP-00000022", "ss-noservice") in new stack
-- <SIP/Wiline_SIP-00000022> Playing 'ss-noservice' (language 'en')
== Spawn extension (from-trunk, s, 4) exited non-zero on 'SIP/Wiline_SIP-00000022'
-- Executing [h@from-trunk:1] Hangup("SIP/Wiline_SIP-00000022", "") in new stack
== Spawn extension (from-trunk, h, 1) exited non-zero on 'SIP/Wiline_SIP-00000022'
uspbx01*CLI>
 

dicko

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#6
If your "Telco" is actually a SIP provider (IMHO that would not normally be considered a Telco, which is generally reserved for RBOC's ILEC's and other TDM/analog vendors) , then you will need to allow anonymous sip calls in "general settings" or specifically redirect the Inbound routes.

(To do this stuff, you really gotta to go get those FM's out and read them dude!, I just read your concurrent post in another forum and it's obvious to me that you are way short of being a "sharp knife" yet when it comes to VOIP, but anyway good luck and I'm sure you will get sharp after your sorely needed "bit of effort".)

Maybe we'll catch you later and after you do that "due diligence" so necessary in the VOIP world.

dicko
 

eijob

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#7
I agree with your opinion, my carrier also provides those, i was just stating how the calls are arriving to me (via SIP connect). Sorry if I wasnt clear on the terms. My bad on that one.

You are also correct on being a sharp knife, i have a long way to go. But I sure am learning along the way. :)

I tried what you suggested ticking allow anonymous sip calls in "general settings" still "no go".

I will continue to do research. Hopefully we can make this work from the freepbx gui.

Ill let you know.

Thanks Dicko

Cheers!
Eijob
 

dicko

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#8
Keep on trucking my friend, I can assure you that it works, for me just as I explained, and others also to my knowledge.

regards and keep us informed . . .

dicko
 

eijob

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#9
Hi Dicko,

I misunderstood your statements. I only changed the trunks to "from-internal" of my asterisk machines.

I had the SIP trunk from my provider set to "from-trunk"

When I changed that, everything worked to what I wanted.

Thank you very very much! B)

Eijob
 

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