spa 3102 setting with elastix pbx

Discussion in 'General' started by syedtafazull, Feb 20, 2011.

  1. syedtafazull

    Joined:
    Jan 2, 2011
    Messages:
    13
    Likes Received:
    0
    I have connected spa3102 voip getway as ATA for my elastix PBX, i have done all the setting descried in elastix with out tears guide, but when i make call i can here the ringing but at the destination there is no response (no ring noise is there ). what i shoud do for that to solve the problum, please help me soon, THANK U. :S
     
  2. danardf

    Joined:
    Dec 3, 2007
    Messages:
    8,069
    Likes Received:
    12
    Hi syedtafazull and welcome to our Elastix Forum.

    Could you discribe exactly your config? (side Elastix and side SPA) :huh:
     
  3. syedtafazull

    Joined:
    Jan 2, 2011
    Messages:
    13
    Likes Received:
    0
    this is the setting i have done
    Line 1 tab
    Proxy: 192.168.1.201
    Register Expires: 60
    Display Name: stc
    User ID: 10
    Password: 10
    Silence Threshold: medium
    DTMF Tx Method: INFO
    Hook Flash Tx Method: INFO

    PSTN Line tab
    Proxy: 192.168.1.201
    Register: no
    Make Call Without Reg: yes
    Ans Call Without Reg: yes
    Display Name: No name
    User ID: PSTN
    Password: password
    Silence Supp Enable: no
    Echo Canc Enable: no
    Echo Canc Adapt Enable: no
    Echo Supp Enable: no
    FAX CED Detect Enable: yes
    FAX CNG Detect Enable: yes
    FAX Passthru Codec: G711u
    FAX Codec Symmetric: no
    FAX Passthru Method: None
    DTMF Tx Method: INFO
    FAX Process NSE: no
    Dial Plan 1: (S0<:myphone number>)
    VoIP Caller Default DP: none
    PSTN Ring Thru Line 1: no
    PSTN CID For VoIP CID: yes
    Elastix Without Tears Page 94 of 299
    PSTN Answer Delay: 2
    PSTN Ring Thru Delay: 3
    PSTN Ring Timeout: 4
    PSTN Hook Flash Len: .1
    Disconnect Tone: 425@-30,425@-30;1(.375/.375/1+2)
    FXO Port Impedance: 220+820||120nF
    On-Hook Speed: 26ms


    Outbound Caller ID: <my phone number> (
    Maximum Channels: 1
    Dial Rules: 0+NXXXXXXXX (for example)
    0011+ZXXXXXXXXXX.
    Trunk Name: stc
    Peer Details:
    canreinvite=no
    context=from-pstn
    host=192.168.1.200 (spa IP)
    insecure=very
    nat=no
    port=5061 (for example)
    qualify=yes
    type=peer
    username=PSTN
    User Context: stc-incoming
    User Details:
    canreinvite=no
    context=from-pstn
    host=192.168.1.200
    insecure=very
    nat=no
    port=5061 for example
    type=user
    username=PSTN
     
  4. danardf

    Joined:
    Dec 3, 2007
    Messages:
    8,069
    Likes Received:
    12
    Side SPA, it seems right.
    Side Elastix, in my case, this trunk is made like that:

    Code:
    canreinvite=no
    context=from-trunk
    dtmfmode=rfc2833
    host=ip_SPA
    insecure=very
    nat=no
    port=5061 
    qualify=yes
    type=peer
    username=pstn
    disallow=all
    allow=alaw
    fromdomain=ip_elastix
    But nothing into user context and details.
    It works well for me.

    About SIP port, Line and PSTN should be different. For example:
    5060 and 5061, 5061 and 5062....etc

    You could apply also RFC2833 instead of INFO about dtmfmode.

    There's some years, i made a little document, but it's in French. However, there's some screen captures. ;)

    Look at this HERE, and let me know.



    Regards
     
  5. syedtafazull

    Joined:
    Jan 2, 2011
    Messages:
    13
    Likes Received:
    0
    i done setting according to elastix with out tear guide as well as u described, but when i make call the destination is ringing but at the destination there is no answer, but for sum destination after one ring it is show in connected but there is no reply from destination.
     
  6. danardf

    Joined:
    Dec 3, 2007
    Messages:
    8,069
    Likes Received:
    12
    Hi.

    What's the result of: sip show peers ? (only your trunk line).
     
  7. syedtafazull

    Joined:
    Jan 2, 2011
    Messages:
    13
    Likes Received:
    0
    this is the result of (sip show peer PSTN) command

    * Name : PSTN
    Secret : <Set>
    MD5Secret : <Not set>
    Remote Secret: <Not set>
    Context : from-trunk
    Subscr.Cont. : <Not set>
    Language :
    AMA flags : Unknown
    Transfer mode: open
    CallingPres : Presentation Allowed, Not Screened
    Callgroup :
    Pickupgroup :
    Mailbox :
    VM Extension : *97
    LastMsgsSent : 32767/65535
    Call limit : 0
    Dynamic : No
    Callerid : "" <>
    MaxCallBR : 384 kbps
    Expire : -1
    Insecure : no
    Nat : No
    ACL : No
    T.38 support : No
    T.38 EC mode : Unknown
    T.38 MaxDtgrm: -1
    DirectMedia : No
    PromiscRedir : No
    User=Phone : No
    Video Support: No
    Text Support : No
    Ign SDP ver : No
    Trust RPID : No
    Send RPID : No
    Subscriptions: Yes
    Overlap dial : Yes
    DTMFmode : rfc2833
    Timer T1 : 500
    Timer B : 32000
    ToHost : 192.168.1.200
    Addr->IP : 192.168.1.200 Port 5061
    Defaddr->IP : 0.0.0.0 Port 5060
    Prim.Transp. : UDP
    Allowed.Trsp : UDP
    Def. Username: PSTN
    SIP Options : (none)
    Codecs : 0x4 (ulaw)
    Codec Order : (ulaw:20)
    Auto-Framing : No
    100 on REG : Yes
    Status : OK (5 ms)
    Useragent :
    Reg. Contact :
    Qualify Freq : 60000 ms
    Sess-Timers : Accept
    Sess-Refresh : uas
    Sess-Expires : 1800 secs
    Min-Sess : 90 secs
    Parkinglot :
     
  8. danardf

    Joined:
    Dec 3, 2007
    Messages:
    8,069
    Likes Received:
    12
    Ok.

    Now, you could try to make a trace from your call.
    Stay on CLI mode, and make a call.
    You could see step by step the way to do. For example, if your call take the trunk, the called number ..Etc
     
  9. syedtafazull

    Joined:
    Jan 2, 2011
    Messages:
    13
    Likes Received:
    0
    How 2 do call through cli mode plese help me
     
  10. danardf

    Joined:
    Dec 3, 2007
    Messages:
    8,069
    Likes Received:
    12
    If you know to do sip show peer, you know how make a trace.
    Simply:
    # asterisk -rvvvvvvvvvvvvvvvv
    CLI>

    Make your call.
    Look at this

    CLI>[ctrl] + [C] or exit

    The result will be something like that.
    Code:
    -- Executing [s@macro-dialout-trunk:29] Goto("SIP/39014-0000000f", "s-CHANUNAVAIL,1") in new stack
        -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
        -- Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] Set("SIP/39014-0000000f", "RC=1") in new stack
    ....
        -- Executing [continue@macro-dialout-trunk:4] Set("SIP/39014-0000000f", "CALLERID(number)=39014") in new stack
        -- Executing [35000@from-internal:7] Macro("SIP/39014-0000000f", "outisbusy,") in new stack
        -- Executing [s@macro-outisbusy:1] Progress("SIP/39014-0000000f", "") in new stack
        -- Executing [s@macro-outisbusy:2] GotoIf("SIP/39014-0000000f", "0?emergency,1") in new stack
        -- Executing [s@macro-outisbusy:3] GotoIf("SIP/39014-0000000f", "0?intracompany,1") in new stack
        -- Executing [s@macro-outisbusy:4] Playback("SIP/39014-0000000f", "all-circuits-busy-now&pls-try-call-later, noanswer") in new stack
        -- <SIP/39014-0000000f> Playing 'all-circuits-busy-now.gsm' (language 'fr')
      == Spawn extension (macro-outisbusy, s, 4) exited non-zero on 'SIP/39014-0000000f' in macro 'outisbusy'
      == Spawn extension (from-internal, 35000, 7) exited non-zero on 'SIP/39014-0000000f'
        -- Executing [h@from-internal:1] Macro("SIP/39014-0000000f", "hangupcall") in new stack
        -- Executing [s@macro-hangupcall:1] GotoIf("SIP/39014-0000000f", "1?noautomon") in new stack
        -- Goto (macro-hangupcall,s,3)
        -- Executing [s@macro-hangupcall:3] NoOp("SIP/39014-0000000f", "TOUCH_MONITOR_OUTPUT=") in new stack
        -- Executing [s@macro-hangupcall:4] GotoIf("SIP/39014-0000000f", "1?noautomon2") in new stack
        -- Goto (macro-hangupcall,s,6)
        -- Executing [s@macro-hangupcall:6] NoOp("SIP/39014-0000000f", "MONITOR_FILENAME=") in new stack
        -- Executing [s@macro-hangupcall:7] GotoIf("SIP/39014-0000000f", "1?skiprg") in new stack
        -- Goto (macro-hangupcall,s,10)
        -- Executing [s@macro-hangupcall:10] GotoIf("SIP/39014-0000000f", "1?skipblkvm") in new stack
        -- Goto (macro-hangupcall,s,13)
        -- Executing [s@macro-hangupcall:13] GotoIf("SIP/39014-0000000f", "1?theend") in new stack
        -- Goto (macro-hangupcall,s,15)
        -- Executing [s@macro-hangupcall:15] Hangup("SIP/39014-0000000f", "") in new stack
      == Spawn extension (macro-hangupcall, s, 15) exited non-zero on 'SIP/39014-0000000f' in macro 'hangupcall'
      == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/39014-0000000f'
     
  11. danardf

    Joined:
    Dec 3, 2007
    Messages:
    8,069
    Likes Received:
    12
    A way, you must find your called number with the good format.
    Remember that you have a dial rules:
    0+NXXXXXXXX
    0011+ZXXXXXXXXXX

    This way could modify your called number, so .... a wrong number sent.

    You could try without these dial rules, just for test.
     
  12. syedtafazull

    Joined:
    Jan 2, 2011
    Messages:
    13
    Likes Received:
    0
  13. syedtafazull

    Joined:
    Jan 2, 2011
    Messages:
    13
    Likes Received:
    0
    yes it is showing that , it is making use of trunk, and the dile pattern that i defined in that trunk.but still not working. is there some other technique to do this.
     
  14. danardf

    Joined:
    Dec 3, 2007
    Messages:
    8,069
    Likes Received:
    12
    Ok.
    From Elastix is sending: 054757...etc
    So at first step, you must have a dial pattern from outgoing call route.
    05xxxxxxxx (for example).
    Next, point your PSTN trunk on with this route.

    Leave blank dial rules in your trunk.

    Make your call. 054757...

    Next, you'll see taking up the PSTN line , and more, in the SPA web page, the last called number.
    If you could see the last called numbre on this page, i guess that's working.
     
  15. syedtafazull

    Joined:
    Jan 2, 2011
    Messages:
    13
    Likes Received:
    0
    it is not showing last called number on spa Graphical web interface, bu on cli of elastix it show , savin useragent "Linksys/SPA3102-3.3.6(GW)"for peer 10.
     
  16. danardf

    Joined:
    Dec 3, 2007
    Messages:
    8,069
    Likes Received:
    12
    By default, the dtmfmode is RFC2833. Try to harmonised this mode.
    Include this method in your config trunk.
     
  17. syedtafazull

    Joined:
    Jan 2, 2011
    Messages:
    13
    Likes Received:
    0
    ya i have included dtmfmode is RFC2833 in trunk, but thing is that in my spa dtmf mode i set to AUTO,because there is no option of rfc2833
     
  18. danardf

    Joined:
    Dec 3, 2007
    Messages:
    8,069
    Likes Received:
    12
    Right, you could select AUTO.
     
  19. syedtafazull

    Joined:
    Jan 2, 2011
    Messages:
    13
    Likes Received:
    0
    after setting auto also it is not working, ia there any another method to do this.
     
  20. syedtafazull

    Joined:
    Jan 2, 2011
    Messages:
    13
    Likes Received:
    0
    after setting auto also it is not working, is there any another method to do this.
     

Share This Page