Author Topic: [Dianemo S] Phone system configuration  (Read 821 times)

Domodude

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[Dianemo S] Phone system configuration
« on: March 13, 2012, 08:33:25 am »
Hi (particularly, Dianemo users),

The phone system is installed a bit differently on Dianemo S than on LMCE, presumably because of pnp being turned off. On my installation, I am having a hard time installing the phone system. I seem to remember that on the Phone Lines page, you would see a "registered" notification. It does not happen now. But once I go into FreePBX, I can see that I do have a trunk.

If that's not confusing enough: I thought I'd use the trunk and place a call from an orbiter phone to my mobile. No luck, phone does not even ring.

So the bottom line is this. What is the best way to set up the phone system on Dianemo S?

Thanks,
Mark

bulek

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Re: [Dianemo S] Phone system configuration
« Reply #1 on: March 13, 2012, 10:20:54 am »
Hi,

I remember that I had to fix few things and got telephony in some kind of "working state". I think that some of basic Asterisk debugging skills should be used... First of all, enter asterisk shell with "asterisk -vvvvvvvvvvvvvvvrgc" and from there you can check "sip show registry" state of your SIP trunks, then "sip show peers" your extensions, etc...

Then also in shell you can get debug messages when you try to make a call - then try to locate warnings, errors. You can even get more messages by "tail -f /var/log/asterisk/messages"

I've found my notes and instructions what I needed to fix my system - hope they will help, although they should get into updates soon :
Quote
had to change :
[19.2.2012 3:12:08] tiniahouse: add this below to /etc/asterisk/extensions_custom.conf
[19.2.2012 3:12:11] tiniahouse: [incoming-sip-calls]
include => ext-did

#include extensions_pluto_dial.conf

; ###########################################################################
; PLUTOs "trusted" context
; ###########################################################################
[trusted]
include => from-internal
[19.2.2012 3:12:25] tiniahouse: comments :
[19.2.2012 3:14:05] tiniahouse: I have SIP trunk to my provider on dedicated network card... For some reason, all incoming calls (even from trunk) end in from-sip-external context that is basically meant for anonymous incoming calls... so it's pretty restricted and all I got was "out of service" annoucement...
[19.2.2012 3:15:47] tiniahouse: So I created my own context (first two lines) and also made this context default. That lowers security level, but we already have passwords being the same as extension number... Now, when call comes in, if it's for DID, then it goes to coresponding incoming route - that's first step I made
[19.2.2012 3:17:15] tiniahouse: Now, my telephony is organized in this manner : I route all incoming calls to one dummy Dianemo user (housephone)... Why ? Cause I don't need to have more users and then I have to switch statuses for all of them... So we have housephone and calls are routed according to its state...
[19.2.2012 3:19:18] tiniahouse: so at the end of incoming route, there is a dial for Local channel 307@.... and 307 is virtual extension for user housephone... That was not working cause 307 and all other user extensions were not visible in dialplan... For some reason file with them is not included in any other config files.... So that explains 3rd line
[19.2.2012 3:21:18] tiniahouse: Then I got to proper housephone user, but calling more users at once (according to routing setup in admin page) was not working, cause calls are made as 202@trusted - trusted context was unknown in my system (it's only synonim for context from-internal - so that explains last lines
[19.2.2012 3:22:33] tiniahouse: I also had to change /etc/asterisk/sip_general_custom.conf ,where I changed from-sip-external to my new context incoming
[19.2.2012 3:22:38] tiniahouse: -sip-calls....
[19.2.2012 3:23:34] tiniahouse: Also, there are files missing in /usr/share/asterisk/agi-bin  - those with pluto-... in front of them (I compared to my 7.10 system)...
[19.2.2012 3:23:57] tiniahouse: Now also incoming calls are working ok in my house... Did you follow me ?

Sometimes I just wonder how Dianemo users use telephony... :)

HTH,

regards Bulek.
« Last Edit: March 13, 2012, 10:29:15 am by bulek »
Thanks in advance,

regards,

Bulek.

totallymaxed

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Re: [Dianemo S] Phone system configuration
« Reply #2 on: March 14, 2012, 08:53:40 am »
Yep there are still parts of the the FreePBX/Asterisk installation that are 'works in progress' and some of the old 0710 and earlier Pluto code just is not compatible with FreePBX any longer. I hope we will have a section in the Dianemo S wiki page soon on setting up/configuring telephony too.

Andrew
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Domodude

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Re: [Dianemo S] Phone system configuration
« Reply #3 on: March 15, 2012, 08:48:08 am »
Can't wait!

Domodude

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Re: [Dianemo S] Phone system configuration
« Reply #4 on: March 15, 2012, 11:18:20 am »
I remember that I had to fix few things and got telephony in some kind of "working state".
My biggest problem is getting pluto/dianemo to understand what I did in FreeBPX! I'll wait for the wiki, hopefully that will help ...

bulek

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Re: [Dianemo S] Phone system configuration
« Reply #5 on: March 15, 2012, 11:25:06 am »
My biggest problem is getting pluto/dianemo to understand what I did in FreeBPX! I'll wait for the wiki, hopefully that will help ...

You shouldn't change too much in Freepbx as parts are generated by Dianemo/LMCE scripts. I remember Radu said not to touch extensions (although passwords are trivial and make system vulnerable)... What I changed in Freepbx is only trunk settings and took care so incoming phone calls come to proper extension. The rest is up to Dianemo....

HTH,

regards,

Bulek.
Thanks in advance,

regards,

Bulek.