Calls from PSTN on SPA3102

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Asterisk Users 2 Comments

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

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:

include => saliente_pstn
include => entradas_pstn
include => sips

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

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

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

2 thoughts on - Calls from PSTN on SPA3102

  • Hi Josu,

    In the sip.conf you have to put the correct context of your
    SPA3102-peer/friend. So put a line like


    there. That should get it out of the “default” context it is now going
    to. Your “s” extension in entradas_pstn is already dialling to SIP/103
    so that should be OK.

    Best regards,
    Jeroen Eeuwes