[asterisk-users] Asterisk Query

[garge rama]

Hi Juan,



Thanks for your inputs, I tried with changes you suggested and find my
observation.



After adding context and extension able to make an outgoing call
[Digium-fxs<3333> to X-lite<2000>].



But not able to make incoming call [X-lite<2000> to Digium-fxs<3333>]. Call
failed with,



(1)  “*Call failed: 503 Service Unavailat 3333*” error message on X-lite



(2) “CHANUNAVAIL” on asterisk CLI.



**CLI>        > Saved useragent "X-Lite release 1105d" for peer 2000*

*  == Using SIP RTP CoS mark 5*

*    -- Executing [3333 at my-phones:1] Dial("SIP/2000-00000000", "Zap/1/3333")
in new stack*

*[May  6 13:02:44] WARNING[20496]: channel.c:4003 ast_request: No channel
type registered for 'Zap'*

*[May  6 13:02:44] WARNING[20496]: app_dial.c:1745 dial_exec_full: Unable to
create channel of type 'Zap' (cause 66 - Channel not implemented)*

*  == Everyone is busy/congested at this time (1:0/0/1)*

*    -- Auto fallthrough, channel 'SIP/2000-00000000' status is
'CHANUNAVAIL'*



Please find conf files below.





chan_dahdi.conf

============

[channels]

context=my-phones

usecallerid=yes

hidecallerid=no

immediate=no

signaling=fxo_ks

echocancel=yes

group=1

channel=1



sip.conf

======

[general]

port=5060

bindaddr=0.0.0.0

context=my-phones



[2000]

type=friend

context=my-phones

secret=1234

host=dynamic



extensions.conf

===========

[my-phones]

exten => 2000,1,Dial(SIP/2000)

exten => 3333,1,Dial(Zap/1/3333)



system.conf

========

fxoks=1

loadzone=us

defaultzone=us





Please let me know any other configuration needs to be done.

On Fri, Apr 30, 2010 at 1:12 AM, Juan David Diaz <juanchonk at gmail.com>wrote:

>
>
> 2010/4/29 garge rama <garge.rama at gmail.com>
>
>>
>>
>> Hi,
>>
>>
>>
>> I am new to asterisk and trying to make calls with TDM400P asterisk digium
>> card.
>>
>>
>>
>> I am using asterisk-1.6.2.4, dahdi-linux-complete-2.3.0+2.3.0 and
>> libpri-1.4.10.2 packages which are downloaded from asterisk website (
>> www.asterisk.org)
>>
>> and able to compile successfully. TDM400P Digium card (having only one FXS
>> connected to J4) has installed successfully in PC.
>>
>>
>>
>> I would like to make calls across SIP [x-lite] to analog phone connected
>> to TDM400P Digium card (fxs-j4).
>>
>> For this the following four conf files are modified as shown below.
>>
>>
>>
>> * chan_dahdi.conf*
>>
>> *==============*
>>
>> [channels]
>>
>> context=test
>>
>> usecallerid=yes
>>
>> hidecallerid=no
>>
>> immediate=no
>>
>>
>>
>> signaling=fxo_ks
>>
>> echocancel=yes
>>
>> group=1
>>
>> channel=1
>>
>>
>>
>> *extensions.conf***
>>
>> *=============*
>>
>> [my-phones] ------------------->*EXTEN 3333  does not exists  for your
>> sip peer context*
>>
>> exten => 2000,1,Dial(SIP/2000)
>>
>>  ; Should look like:
>>
> *exten => 3333,1,Asterisk_Application(Action)* ;Dial(Zap/1/${Phone_Number_you
> want})
>
>>   [test]
>>
>> exten => 3333,1,Dial(Zap/1)
>>
>> exten => 3333,2,HangUp()
>>
>>
>>
>> *sip.conf***
>>
>> *=======*
>>
>> [general]
>>
>> port = 5060
>>
>> bindaddr = 0.0.0.0
>>
>> context = others
>>
>>
>>
>> [2000]
>>
>> type=friend
>>
>> *context=**my-phones *
>>
>> secret=1234
>>
>> host=dynamic
>>
>>
>>
>> *system.conf*
>>
>> *==========*
>>
>> fxoks=1
>>
>> loadzone = be
>>
>> defaultzone = be
>>
>>
>>
>> With those changes x-lite getting registered with asterisk and analog
>> device/phone is getting ring tone with off-hook and also getting debug
>> prints on cli, but not able to make calls.
>>
>>
>>
>> Test Setup:
>>
>> ========
>>
>>  X-lite [configured as 2000, password… other info] running on asterisk PC
>> à registered with asterisk.
>>
>>  Analog phone connected to TDM400P Digium card - FXS-J4 running on same
>> asterisk PC à getting ring tone
>>
>>
>>
>> Test Result:
>>
>> =========
>>
>> Tried by calling 3333 from x-lite à getting message on CLI “call from
>> ‘2000’ to ‘3333’ rejected because extension not found”
>>
>> Tried by calling 2000 from analog phone [Digium-FXS-J4] -> getting some
>> engage/disconnected tone while pressing digts [2000] on phone itself.
>>
>>
>>
>> Welcome for your valuable suggestions and comments. Thank You in advance.
>>
>>
>>
>> Regards,
>>
>> Garge.
>>
>>
>>
>> --
>>
>> _____________________________________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
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>>
>
>
>
> --
> Juan.
> Linux User #441131
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>               http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
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