Viewing 6 posts - 1 through 6 (of 6 total)
  • Migrated from old forum
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    Post count: 327
    #5823 |

    Hi after a lot of weeks and still Asterisk noob (or dum)

    My setup:

    FreePBX 2.10.1.9
    Asterisk 1.8
    One extension (named ‘1’)
    USB GSM Dongle working (Hearing “Googbye” only)
    Nothing else (no inbound route, no Voip trunk)

    VDP configured FreePBX/Ast1.8. Using your sample FreePBX/Hello World.

    From extension 1 when dialing 56781 (or anything else by modifying plan to “_.”),i get ‘service unavailable’. See log:

    EDIT OOPS Forgot to mention. In order for the dongle to work i had to re-compile Asterisk. Does this change everything?
    EDIT2 WOW. It works if i make a new context named ‘from-sip-external’ and put the dialplan there. Why???

    —-
    2013-04-21 01:42:41 VERBOSE3832 netsock2.c: == Using SIP RTP TOS bits 184
    2013-04-21 01:42:41 VERBOSE3832 netsock2.c: == Using SIP RTP CoS mark 5
    2013-04-21 01:42:41 VERBOSE8932 pbx.c: — Executing 56781@from-sip-external:1 NoOp(“SIP/192.168.25.141-0000000d”, “Received incoming SIP connection from unknown peer to 56781”) in new stack
    2013-04-21 01:42:41 VERBOSE8932 pbx.c: — Executing 56781@from-sip-external:2 Set(“SIP/192.168.25.141-0000000d”, “DID=56781”) in new stack
    2013-04-21 01:42:41 VERBOSE8932 pbx.c: — Executing 56781@from-sip-external:3 Goto(“SIP/192.168.25.141-0000000d”, “s,1”) in new stack
    2013-04-21 01:42:41 VERBOSE8932 pbx.c: — Goto (from-sip-external,s,1)
    2013-04-21 01:42:41 VERBOSE8932 pbx.c: — Executing s@from-sip-external:1 GotoIf(“SIP/192.168.25.141-0000000d”, “0?checklang:noanonymous”) in new stack
    2013-04-21 01:42:41 VERBOSE8932 pbx.c: — Goto (from-sip-external,s,5)
    2013-04-21 01:42:41 VERBOSE8932 pbx.c: — Executing s@from-sip-external:5 Set(“SIP/192.168.25.141-0000000d”, “TIMEOUT(absolute)=15”) in new stack
    2013-04-21 01:42:41 VERBOSE8932 func_timeout.c: Channel will hangup at 2013-04-21 01:42:56.699 EEST.
    2013-04-21 01:42:41 VERBOSE8932 pbx.c: — Executing s@from-sip-external:6 Answer(“SIP/192.168.25.141-0000000d”, “”) in new stack
    2013-04-21 01:42:41 VERBOSE8932 pbx.c: — Executing s@from-sip-external:7 Wait(“SIP/192.168.25.141-0000000d”, “2”) in new stack
    2013-04-21 01:42:43 VERBOSE8932 pbx.c: — Executing s@from-sip-external:8 Playback(“SIP/192.168.25.141-0000000d”, “ss-noservice”) in new stack
    2013-04-21 01:42:43 VERBOSE8932 file.c: — Playing ‘ss-noservice.ulaw’ (language ‘en’)
    2013-04-21 01:42:44 VERBOSE8932 pbx.c: == Spawn extension (from-sip-external, s, 8) exited non-zero on ‘SIP/192.168.25.141-0000000d’
    2013-04-21 01:42:44 VERBOSE8932 pbx.c: — Executing h@from-sip-external:1 Hangup(“SIP/192.168.25.141-0000000d”, “”) in new stack
    2013-04-21 01:42:44 VERBOSE8932 pbx.c: == Spawn extension (from-sip-external, h, 1) exited non-zero on ‘SIP/192.168.25.141-0000000d’
    —-

    Migrated from old forum
    Participant
    Post count: 327
    #5824 |

    That’s because that’s the context asterisk is looking for when an external sip call arrives. You can change that in the asterisk config for inbound routes if wanted. This has more to do with your asterisk config than it does visual dialplan.

    Kevin

    Migrated from old forum
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    Post count: 327
    #5825 |

    Ok this is what i understand so far:

    There is no problem with me having compiled Asterisk for the dongle to work.
    There is no problem with me having no inbound route and no Voip trunk

    BUT i have to make an inbound route to send incoming calls (from?) to …vdp-inbound context?

    Migrated from old forum
    Participant
    Post count: 327
    #5826 |

    That’s correct. If Asterisk doesn’t know WHERE you want to send an inbound call (the context. And there can be many of them or just 1 to handle all calls) then it won’t know what to do with the call and it will fail and usually give a “number not in service” type message.

    Kevin

    Migrated from old forum
    Participant
    Post count: 327
    #5827 |

    For the moment i use the context named ‘from-sip-external’ to write my dialplan until i figure out what to change in the configuration.

    Anyway, thanks for your help.

    Kyriakos

    Migrated from old forum
    Participant
    Post count: 327
    #5828 |

    No problem. Enjoy!

    Kevin

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