[asterisk-users] Beggining asterisk

[Esteban Cacavelos]

2011/9/6 Esteban Cacavelos <estebancacavelos at gmail.com>

>
>
> 2011/9/6 Leif Madsen <leif.madsen at asteriskdocs.org>
>
>> On 04/09/11 02:51 PM, Tamer Higazi wrote:
>>
>>> the 3rd edition is available, but that book covers every thing to run
>>> the asterisk PBX.
>>>
>>
>> You can read the 3rd edition online at http://ofps.oreilly.com/**
>> titles/9780596517342/ <http://ofps.oreilly.com/titles/9780596517342/>
>>
>> HTH!
>> Leif.
>>
>> --
>> Leif Madsen
>> http://www.oreilly.com/**catalog/asterisk<http://www.oreilly.com/catalog/asterisk>
>>
>>
>> --
>> ______________________________**______________________________**_________
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>
>
> Thanks for all the responses !. I will try with ubuntu bundleded packages
> first.
>
> I will post my results.
>
>
>
> Esteban
> --
> Esteban L. Cacavelos de Amoriza
> Cel: 0981 220 429
>


finally i decided to install from source because the documentation suggest
that.

I've installed successfully asterisk+dahdi+libpri. I tested a basic SIP
configuration and there were no problems.

Now i have problems with pstn termination and origination. I have one fxo
module from witch i want to make and receive calls. Can I do that ?. I'll
post my configuration files.

I want to make calls from my android phone (where i have a SIP client) and
recieve calls from my analog line through my androi.

My country code is 595, city code 21, number , xxx xxx


chan_dahdi.conf

[channels]

;
; To apply other options to these channels, put them before "channel".
;
signalling=fxs_ks  ; in Asterisk, FXO channels use FXS signaling
                     ; (and yes, FXS channels use FXO signaling)
context=from-pstn
channel => 1       ; apply all the previously defined settings to this
channel


extensions.conf
[LocalSets]

exten => 100,1,Dial(SIP/android-esteban) ; Replace 0000FFFF0001 with your
device name

exten => 101,1,Dial(SIP/recepcion) ; Replace 0000FFFF0002 with your device
name


exten => 200,1,Answer()
    same => n,Playback(hello-world)
    same => n,Hangup()

; TERMINATION
[from-voip-network]
exten => _X.,1,Verbose(2, Call from VoIP network to ${EXTEN})
   same => n,Dial(DAHDI/g0/${EXTEN})

ORIGINATION
[from-pstn]
; This is the context that would be listed in the config file
; for the circuit (i.e. chan_dahdi.conf)

exten => _X.,1,Dial(SIP/android-esteban)

[number-mapping]
; This context is not strictly required, but will make it easier
; to keep track of your DIDs in a single location in your dialplan.
; From here you can pass the call to another part of the dialplan
; where the actual dialplan work will take place.

exten => 59521xxxxxx,1,Dial(SIP/android-esteban)

exten => i,1,Verbose(2,Incoming call to invalid number)




Dahdi system.conf

# Autogenerated by /usr/sbin/dahdi_genconf on Tue Sep  6 14:40:03 2011
# If you edit this file and execute /usr/sbin/dahdi_genconf again,
# your manual changes will be LOST.
# Dahdi Configuration File
#
# This file is parsed by the Dahdi Configurator, dahdi_cfg
#
# Span 1: WCTDM/4 "Wildcard TDM400P REV I Board 5" (MASTER)
fxsks=1
echocanceller=mg2,1
# channel 2, WCTDM/4/1, no module.
# channel 3, WCTDM/4/2, no module.
# channel 4, WCTDM/4/3, no module.

# Global data

loadzone        = us
defaultzone     = us



Thanks in advance !

-- 
Esteban L. Cacavelos de Amoriza
Cel: 0981 220 429
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