[asterisk-users] Calls from PSTN on SPA3102

[Josu Lazkano]

Hello list, this is my first post on this list.

I have a server with Asterisk and a Linksys SPA3102 with 3 SIP phones.
I have configured the SPA PSTN line as trunk to receive and send
calls.

I can call outside from SIP phone throw the PSTN line and all is OK,
the problem is when I receive a call from the PSTN, on the out caller
phone there is a demo playback. I want to redirect the call to a
internal SIP phone.

This is the extensions.conf:

[spa]
include => saliente_pstn
include => entradas_pstn
include => sips

[saliente_pstn]
exten => _9ZXXXXXXX,1,Dial(SIP/${EXTEN}@pstn,60,)
exten => _9ZXXXXXXX,n,Hangup

[entradas_pstn]
exten => s,1,Dial(SIP/103,20,tm)
exten => s,2,VoiceMail(103)
exten => s,3,Hangup

[sips]
exten => 100,1,Dial(SIP/100,20,Ttm) ; extensión 100
exten => 100,2,Voicemail(100)
exten => 100,3,Hangup
exten => 101,1,Dial(SIP/101,20,Ttm) ; extensión 101
exten => 101,2,Voicemail(101)
exten => 101,3,Hangup
exten => 102,1,Dial(SIP/102,20,Ttm) ; extensión 102
exten => 102,2,Voicemail(102)
exten => 102,3,Hangup
exten => 103,1,Dial(SIP/103,20,Ttm) ; extensión 103
exten => 103,2,Voicemail(103)
exten => 103,3,Hangup

When I receive a call from outside this is the asterisk console log:

  == Using SIP RTP CoS mark 5
    -- Executing [s at default:1] wait("SIP/pstn-00000004", "1")
    -- Executing [s at default:1] answer("SIP/pstn-00000004", "")
    -- Digit timeout set to 5.000
    -- Response timeout set to 10.000
    -- Executing [s at default:1] background("SIP/pstn-00000004", "demo-congrats")
    -- <SIP/pstn-00000004> Playing 'demo-congrats.slin' (language 'en')
[Oct 31 20:55:55] NOTICE[4001]: channel.c:3066 __ast_read: Dropping
incompatible voice frame on SIP/pstn-00000004 of format ulaw since our
native format has changed to 0x8 (alaw)
    -- Executing [s at default:1] background("SIP/pstn-00000004", "demo-instruct")
    -- <SIP/pstn-00000004> Playing 'demo-instruct.slin' (language 'en')
    -- Executing [s at default:1] waitexten("SIP/pstn-00000004", "")
    -- Timeout on SIP/pstn-00000004, going to 't'
    -- Executing [t at default:1] playback("SIP/pstn-00000004", "demo-thanks")
    -- <SIP/pstn-00000004> Playing 'demo-thanks.slin' (language 'en')
    -- Executing [t at default:1] hangup("SIP/pstn-00000004", "")
  == Spawn extension (default, t, 1) exited non-zero on 'SIP/pstn-00000004'
  == Using SIP RTP CoS mark 5
    -- Executing [s at default:1] wait("SIP/pstn-00000005", "1")
    -- Executing [s at default:1] answer("SIP/pstn-00000005", "")
    -- Digit timeout set to 5.000
    -- Response timeout set to 10.000
    -- Executing [s at default:1] background("SIP/pstn-00000005", "demo-congrats")
    -- <SIP/pstn-00000005> Playing 'demo-congrats.slin' (language 'en')
[Oct 31 20:59:33] NOTICE[4015]: channel.c:3066 __ast_read: Dropping
incompatible voice frame on SIP/pstn-00000005 of format ulaw since our
native format has changed to 0x8 (alaw)
    -- Executing [s at default:1] background("SIP/pstn-00000005", "demo-instruct")
    -- <SIP/pstn-00000005> Playing 'demo-instruct.slin' (language 'en')

How could I make to redirect the call to the 103 extension?

Thanks for your help, best regards.

-- 
Josu Lazkano