Channel 'DAHDI/1-1' sent into invalid extension

Asterisk Asterisk Support
1

Hi I am setting up a very simple asterisk system with 1 analog card TDM410P and a few SIP account extensions in my home. I use a Debian 7.

When calls are incoming on the TDM410P, they should be forwarded to SIP/3020(ATA SPA 8000).

I have setup a basic and without any doubt incorrect config, and I am getting the following errors:

    > Saved useragent "Zoiper r30798" for peer 3028
    -- Starting simple switch on 'DAHDI/1-1'
    [May 20 15:34:57] WARNING[5944][C-00000000]: pbx.c:6646 __ast_pbx_run: Channel 'DAHDI/1-1' sent to invalid extension but no invalid handler: context,exten,priority=ramais,s,1
    -- Hanging up on 'DAHDI/1-1'
    -- Hungup 'DAHDI/1-1'

My config is as follows:

cat chan_dahdi.conf

    ; General options
    [channels]
    usecallerid=yes
    hidecallerid=no
    callwaiting=yes
    threewaycalling=yes
    transfer=yes
    vechocancel=yes
    echocancelwhenbridged=yes
    rxgain=0.0
    txgain=0.0

    ;FXO Modules
    group=0
    vechocancel=yes
    signalling=fxs_ks
    context=ramais
    channel=1

cat sip.conf

cat /etc/asterisk/sip.conf

     [general]
     udpbindaddr=0.0.0.0:5089
    context=ramais
    disallow=all
    allow=ulaw,alaw,gsm
    externhost=myhost.noip.us:5089
    localnet=192.3.1.2/255.255.255.0
    ;;
    ;SPA8800 Changes
    ;define SPA8800 analog phone 1 
    [3020]
    type=friend
    username=3020
    secret=100567
    qualify=yes
    allow=ulaw
    host=dynamic
    canreinvite=no
    regext=3020
    ;;
    [3021]
    type=friend
    username=3021
    secret=100567
    qualify=yes
    host=dynamic
    canreinvite=no
    regext=3021
    ;;
    [3022] 
    type=friend
    username=3022
    secret=100567
    qualify=yes
    nat=no
    host=dynamic
    canreinvite=no
    regext=3022
    ;;
    [3023]
    type=friend
    username=3023
    secret=100567
    qualify=yes
    nat=no
    host=dynamic
    canreinvite=no
    regext=3023
    ;;
    [3024]
    type=friend
    username=3024
    secret=100567
    qualify=yes
    nat=no
    host=dynamic
    canreinvite=no
    regext=3024
    ;;
    [3025]
    type=friend
    username=3025
    secret=100567
    qualify=yes
    nat=no
    host=dynamic
    canreinvite=no
    regext=3025
    ;;
    [3026]
    type=friend
    username=3026
    secret=100567
    qualify=yes
    nat=no
    host=dynamic
    canreinvite=no
    regext=3026
    ;;
    [3027]
    type=friend
    username=3027
    secret=100567
    qualify=yes
    nat=no
    host=dynamic
    canreinvite=no
    regext=3027
    ;;
    [3028]
    type=friend
    username=3028
    secret=100567
    qualify=yes
    host=dynamic
    nat=force_rport,comedia
    canreinvite=no
    regext=3028
    allow=ulaw    
    ;;

cat extensions.conf

      [globals] ;Define as variáveis Globais
      TELEFONISTA=SIP/3020 ;Declaração de variável

      [general] ;Opções gerais do Dialplan
      writeprotect=no ;Modo somente leitura
      static=yes ;Modo estático

      [ramais]
      exten => _XXXX,1,Dial(SIP/${EXTEN},60,tT)

I have a few questions:

What context should I use for the incoming calls? Is necessary to fix in my extensions.conf for make route from PSTN to my SIP/2030?

Finally, any idea why my config doesnt work and i get bombed out with Channel ‘DAHDI/1-1’ sent into invalid extension ‘s’ in context ‘default’, but no invalid handler

Appreciate the help in getting my very first test install

Many thanks in advance.

2

I would suggest adding s extension in your incoming call context like below

[ramais]
exten => s,1,Dial(SIP/3020,60,tT)

and then reading through
asteriskdocs.org/ (which seems to be down at this moment!)
wiki.asterisk.org/wiki/display/AST/Home

–Satish Barot

3

[quote=“satish4asterisk”]I would suggest adding s extension in your incoming call context like below

[ramais]
exten => s,1,Dial(SIP/3020,60,tT)

and then reading through
asteriskdocs.org/ (which seems to be down at this moment!)
wiki.asterisk.org/wiki/display/AST/Home

–Satish Barot[/quote]

Hi Satish, thanks very much for your reply!