Pri channel (sangoma 102 and Elastix 2.0.2

Discussion in 'General' started by zurqui, Sep 27, 2010.

  1. zurqui

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    Hello team

    How are you?

    I need your help.....finally I was able to install Elastix 2.0.2 in my server. Everything is ok. The server has and sangoma card (A102) and a Pri channel from mi ISP installed.

    When i check the hardware detection, the 30 PRI channles appear on green saying that the channels are not in use.

    However, when I try do do a call to the Pri number from My cel phone or any landline phone i just receive a busy tone.

    Some idea? I really will appreciated any help on this.

    Thanks!!

    BTW: I'm located at Costa Rica

    Regards

    Zurqui
     
  2. trymes

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    Have you configured an Inbound Route to handle calls that come in to the server?

    You also need to configure Wanrouter and DAHDI, though I presume that you have...

    Tom
     
  3. zurqui

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    Hi Trymes

    First, thanks a lot for your reply.

    I want to be honest, i'm a completely newbie on this. I have the resources, I mean, the server, the sangoma card and the pri landline to this project.

    As i sais before the elastix recognized the sangoma card and the dashboard says the card is up on green color with the label " is not ion use" on every of the channels.

    Days before I created a inbound route but does not worked. I', not able to receive or make calls. In the other side I did create a virtual trunk and it's working very well.

    Could you guide me on what a nedd to do to have that Pri connection up and running? I feel very frustrated.

    To configure the card I ran "setup-sangoma" only

    Thanks in advance for any help that you or anyone can give me.

    Regards

    Allan
     
  4. trymes

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    My first recommendation would be that you go into Inbound Routes and verify that there is a route there that will catch calls coming in on the Sangoma card and direct them to where you want them to go.

    The simplest Inbound Route is one that has the DID and CID fields blank, so it will match any incoming call, regardless of CID or DID. Then set a destination (like an IVR) that FreePBX will send the call to when an incoming call matches those criteria.

    Let me know what you find!

    Tom
     
  5. zurqui

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    Well,,,, this is a frustration to me!!! :(

    I created the inbound route and nothing, I did some calls to the phone number assgined and just recived a busy tone.

    However, Elastix have recognized the sangoma card saying that has 30 channels (It's a Pri connection) "not in use" and with "green" status.

    I already confirm with my provider and everything is ok, for they the line is well configured and up. The isuse seems to me located in my Elastix box.

    Any help on hits will be very appreciated.

    Regards

    Zurqui
     
  6. trymes

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    Zurqui,

    Can you post the contents of the /etc/asterisk/chan_dahdi.conf file?

    Did you tell Elastix to replace that file in the Hardware detector?

    Does FreePBX have a Trunk defined for ZAP Channel g0?

    If you log in to your box and run the command "asterisk -r", do you see anything happen when you make in inbound call? If not, run "core set verbose 3" at the asterisk (not linux) command prompt and try again.

    You can't be too far from getting it right!

    Tom
     
  7. zurqui

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    Tom

    Here you go the answers

    BTW Thanks a lot for your help, I really appreciated it.

    Can you post the contents of the /etc/asterisk/chan_dahdi.conf file?


    ; Autogenerated by /usr/sbin/setup-sangoma 2010-09-25
    ; If you edit this file and execute /usr/sbin/setup-sangoma again,
    ; your manual changes will be LOST.
    ; Dahdi Channels Configurations (chan_dahdi.conf)
    ;
    ; This is not intended to be a complete chan_dahdi.conf. Rather, it is intended
    ; to be #include-d by /etc/chan_dahdi.conf that will include the global settings
    ;



    ;Sangoma A102 port 1 [slot:5 bus:15 span:1] <wanpipe1>
    switchtype=euroisdn
    context=from-zaptel
    group=0
    echocancel=yes
    signalling=pri_cpe
    channel =>1-15,17-31



    #include chan_dahdi_additional.conf
    #include dahdi-channels.conf


    Did you tell Elastix to replace that file in the Hardware detector? NO

    Does FreePBX have a Trunk defined for ZAP Channel g0? YES

    If you log in to your box and run the command "asterisk -r", do you see anything happen when you make in inbound call? If not, run "core set verbose 3" at the asterisk (not linux) command prompt and try again.

    I ran both commands but when I did the test, nothing appear.

    Thanks Tom...if you need more information let me know!

    Allan
     
  8. trymes

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    GOD I HATE THIS BLASTED FORUM!!!!!! IT JUST ATE ANOTHER POST UPON SUBMISSION!!!!! I clicked "Submit" and it complained "Code not correct", and then ate the friggin post.

    Anyhow, here is the much abbreviated version of the long, quite helpful post I just made. Sorry it's not quite so good.

    1.) Open the Hardware Detector
    2.) check the "Replace the file chan_dahdi.conf"
    3.) Check "Detect Sangoma Hardware"
    4.) Click "Detect New Hardware"
    5.) Go to the PBX tab and click on Tools (top right)
    6.) Enter "reload" in the text area and click "Execute"
    7.) Wait 30 seconds and then try again and see what you get.

    Tom
     
  9. dicko

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    I question that whatever script did it that a PRI should be in the context from-zaptel, that is for adding pseudo DID info to FXO's that are intrinsically DID by channel and the DID is unknown by anyone but the subscriber, PRI's have the advantage of the D channel which will identify the various DID's so do NOT need that overlaid from-zaptel artifact, adding DID info on a channel by channel basis, as the from-zaptel context allows and requires, to a PRI makes absolutely no sense at all, it just won't work, the context must be from-pstn for it to work.

    Is this a bug either in Elastix or the sangoma script.?

    dicko
     
  10. zurqui

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    Hello Tom Hello Dicko

    Tom, unfortunately I ran the steps you sent me but the issue still there. I still receiving a busy tone when I called from outside and when a tried to do a call to outside I received a messages saying that all channels are busy.

    Some idea?
     
  11. trymes

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    Dicko: I think that chan_dahdi.conf was generated by setup-sangoma.

    Zurqui: can you check the chan_dahdi file again and see if it has changed now that you asked Elastix to replace it?

    Also, you did reload the system, right? (Step #6)

    Next up you have two options to my way of thinking:

    1.) Log onto the Asterisk console and see what scrolls by when you try to make a call in or out. To do this, log into your server via SSH. Run "asterisk -r" and you should see
    Code:
    Asterisk 1.6.2.10, Copyright (C) 1999 - 2010 Digium, Inc. and others.
    Created by Mark Spencer <markster@digium.com>
    Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
    This is free software, with components licensed under the GNU General Public
    License version 2 and other licenses; you are welcome to redistribute it under
    certain conditions. Type 'core show license' for details.
    =========================================================================
    Connected to Asterisk 1.6.2.10 currently running on vox (pid = 3377)
    Verbosity is at least 3
    vox*CLI>
    If it doesn't say "Verbosity is at least 3", type "core set verbose 3" and press return. Then make an inbound call and see if anything scrolls by on the screen. The answer to what the problem is might be there.

    2.) If #1 doesn't help any, maybe set up an incoming SIP or IAX trunk from a provider like TelIAX (http://www.teliax.com). If you can get that working, you know that your incoming routing, etc works fine. Then focus on the PRI.


    Tom
     
  12. zurqui

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    Hello Tom

    yes, the chan_dandhi files has been changed to this:

    ==========================================================================================


    ; Auto-generated by /usr/sbin/hardware_detector
    [trunkgroups]

    [channels]
    context=from-pstn
    signalling=fxs_ks
    rxwink=300 ; Atlas seems to use long (250ms) winks
    usecallerid=yes
    hidecallerid=no
    callwaiting=yes
    usecallingpres=yes
    callwaitingcallerid=yes
    threewaycalling=yes
    transfer=yes
    canpark=yes
    cancallforward=yes
    callreturn=yes
    echocancel=yes
    echocancelwhenbridged=no
    faxdetect=incoming
    echotraining=800
    rxgain=0.0
    txgain=0.0
    callgroup=1
    pickupgroup=1

    ;Uncomment these lines if you have problems with the disconection of your analog lines
    ;busydetect=yes
    ;busycount=3


    immediate=no

    #include dahdi-channels.conf
    #include chan_dahdi_additional.conf

    =======================================================================================

    The step #6 was completed as suggested, the server was reloaded.

    I ran asteriks -r and did some tests calling gthe Pri number but nothing appear. However, calling a DID I already have at Miami USA the console showed the next:

    ========================================================================================

    [root@voip ~]# asterisk -r
    Asterisk 1.6.2.10, Copyright (C) 1999 - 2010 Digium, Inc. and others.
    Created by Mark Spencer <markster@digium.com>
    Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
    This is free software, with components licensed under the GNU General Public
    License version 2 and other licenses; you are welcome to redistribute it under
    certain conditions. Type 'core show license' for details.
    =========================================================================
    Connected to Asterisk 1.6.2.10 currently running on voip (pid = 3033)
    Verbosity is at least 3
    -- Accepting AUTHENTICATED call from 74.63.41.218:
    > requested format = ulaw,
    > requested prefs = (ulaw|alaw|gsm|g729|g726|ilbc),
    > actual format = ulaw,
    > host prefs = (ulaw),
    > priority = mine
    -- Executing [7862285572@from-trunk:1] NoOp("IAX2/voipms-838", "Catch-All DID Match - Found 7862285572 - You probably want a DID for this.") in new stack
    -- Executing [7862285572@from-trunk:2] Goto("IAX2/voipms-838", "ext-did,s,1") in new stack
    -- Goto (ext-did,s,1)
    -- Executing [s@ext-did:1] Set("IAX2/voipms-838", "__FROM_DID=s") in new stack
    -- Executing [s@ext-did:2] Gosub("IAX2/voipms-838", "app-blacklist-check,s,1") in new stack
    -- Executing [s@app-blacklist-check:1] GotoIf("IAX2/voipms-838", "0?blacklisted") in new stack
    -- Executing [s@app-blacklist-check:2] Set("IAX2/voipms-838", "CALLED_BLACKLIST=1") in new stack
    -- Executing [s@app-blacklist-check:3] Return("IAX2/voipms-838", "") in new stack
    -- Executing [s@ext-did:3] ExecIf("IAX2/voipms-838", "0 ?Set(CALLERID(name)=50625086000)") in new stack
    -- Executing [s@ext-did:4] Set("IAX2/voipms-838", "__CALLINGPRES_SV=allowed_not_screened") in new stack
    -- Executing [s@ext-did:5] Set("IAX2/voipms-838", "CALLERPRES()=allowed_not_screened") in new stack
    -- Executing [s@ext-did:6] Goto("IAX2/voipms-838", "ext-meetme,3000,1") in new stack
    -- Goto (ext-meetme,3000,1)
    -- Executing [3000@ext-meetme:1] Macro("IAX2/voipms-838", "user-callerid,") in new stack
    -- Executing [s@macro-user-callerid:1] Set("IAX2/voipms-838", "AMPUSER=50625086000") in new stack
    -- Executing [s@macro-user-callerid:2] GotoIf("IAX2/voipms-838", "0?report") in new stack
    -- Executing [s@macro-user-callerid:3] ExecIf("IAX2/voipms-838", "1?Set(REALCALLERIDNUM=50625086000)") in new stack
    -- Executing [s@macro-user-callerid:4] Set("IAX2/voipms-838", "AMPUSER=") in new stack
    -- Executing [s@macro-user-callerid:5] Set("IAX2/voipms-838", "AMPUSERCIDNAME=") in new stack
    -- Executing [s@macro-user-callerid:6] GotoIf("IAX2/voipms-838", "1?report") in new stack
    -- Goto (macro-user-callerid,s,10)
    -- Executing [s@macro-user-callerid:10] GotoIf("IAX2/voipms-838", "0?continue") in new stack
    -- Executing [s@macro-user-callerid:11] Set("IAX2/voipms-838", "__TTL=64") in new stack
    -- Executing [s@macro-user-callerid:12] GotoIf("IAX2/voipms-838", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,19)
    -- Executing [s@macro-user-callerid:19] NoOp("IAX2/voipms-838", "Using CallerID "50625086000" <50625086000>") in new stack
    -- Executing [3000@ext-meetme:2] Set("IAX2/voipms-838", "MEETME_ROOMNUM=3000") in new stack
    -- Executing [3000@ext-meetme:3] Set("IAX2/voipms-838", "MEETME_MUSIC=") in new stack
    -- Executing [3000@ext-meetme:4] GotoIf("IAX2/voipms-838", "0?READPIN") in new stack
    -- Executing [3000@ext-meetme:5] Answer("IAX2/voipms-838", "") in new stack
    -- Executing [3000@ext-meetme:6] Wait("IAX2/voipms-838", "1") in new stack
    -- Executing [3000@ext-meetme:7] Set("IAX2/voipms-838", "PINCOUNT=0") in new stack
    -- Executing [3000@ext-meetme:8] Read("IAX2/voipms-838", "PIN,enter-conf-pin-number,,,,") in new stack
    -- <IAX2/voipms-838> Playing 'enter-conf-pin-number.gsm' (language 'en')
    -- User disconnected
    -- Executing [h@ext-meetme:1] Hangup("IAX2/voipms-838", "") in new stack
    == Spawn extension (ext-meetme, h, 1) exited non-zero on 'IAX2/voipms-838'
    -- Hungup 'IAX2/voipms-838'
    -- Remote UNIX connection
    -- Remote UNIX connection disconnected
    -- Remote UNIX connection
    -- Remote UNIX connection disconnected

    =======================================================================================

    So this means the inbound routing is working and the issues is related internally between the sangoma card and Elastix....
     
  13. trymes

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    Excellent. At least the inbound routing is working. Now to figure out what's up with Sangoma. To begin with, it looks like the Elastix hardware detector is not fully seeing your card, as it did not include anything in the chan_dahdi.conf file for it. My file looks like this (It won't work for you, as mine is a US setup, but...):

    Code:
    [root@vox ~]# cat /etc/asterisk/chan_dahdi.conf
    ;autogenerated by /usr/sbin/wancfg_dahdi do not hand edit
    ;autogenrated on 2010-09-10
    ;Dahdi Channels Configurations
    ;For detailed Dahdi options, view /etc/asterisk/chan_dahdi.conf.bak
    
    [trunkgroups]
    
    [channels]
    context=default
    usecallerid=yes
    hidecallerid=no
    callwaiting=yes
    usecallingpres=yes
    callwaitingcallerid=yes
    threewaycalling=yes
    transfer=yes
    canpark=yes
    cancallforward=yes
    callreturn=yes
    echocancel=yes
    echocancelwhenbridged=yes
    relaxdtmf=yes
    rxgain=0.0
    txgain=0.0
    group=1
    callgroup=1
    pickupgroup=1
    immediate=no
    
    ;Sangoma A101 port 1 [slot:2 bus:10 span:1] <wanpipe1>
    switchtype=national
    context=from-pstn
    group=0
    echocancel=yes
    faxdetect=incoming
    signalling=pri_cpe
    channel =>1-23
    
    Also weird is that your file specifies signaling=fxs_ks, which would be for an FXO port, IIRC. Can you confirm that the Sangoma card is the ONLY Dahdi card in the system?

    I would start by:

    1.) Restart the system (why not?).
    2.) Re-run setup-sangoma
    3.) Go back into the hardware detector and again check both the "replace chan_dahdi.conf" box AND the "Detect Sangoma hardware" box and detect the hardware again.
    4.) Check the contents of your /etc/asterisk/chan_dahdi.conf file and see if it now has a section for the card in it.
    5.) From the Linux Command prompt, run "service wanrouter status" I get output like this:
    Code:
    Devices currently active:
            wanpipe1
    
    
    Wanpipe Config:
    
    Device name | Protocol Map | Adapter  | IRQ | Slot/IO | If's | CLK | Baud rate |
    wanpipe1    | N/A          | A101/1D/A102/2D/4/4D/8| 233 | 2       | 1    | N/A | 0         |
    
    Wanrouter Status:
    
    Device name | Protocol | Station | Status        |
    wanpipe1    | AFT TE1  | N/A     | Connected     |
    
    Worth a shot...

    Tom
     
  14. zurqui

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    Hello Tom

    First, apologyzes for my absent the last days, I had a flue,,,,,so.... you know!! :(

    Look, I did the last steps you sent me and nothing, the Pri still saying that the channels are busy.

    i did check with my provider aboyt the configuration, and this is the info we have:

    LINECODE: HDB3
    FRAMING: NO-CRC4
    SWICHT TYPE: EUROISDN OR PRIMARY NET5 OR PRIMARY EDSS1
    CLOCK : From line (I assume it's the "normal") (CLOCK MASTER is used by the PSTN)
    SIGNALING: PRI ETSI OR SS7(signaling 7)

    I hope this info be important.

    in the other hand I wna to clarify something here:

    Here at CR the PRI connections is delivered using a coaxial equipment, I mean I have to use a Balun and from the balun connect it with the server using a PRI Crossover, so in summary

    PSTN --->PRI coaxial receiver ----> balun-----E1 Crossover----> Sandoma A102


    Regards

    Allan
     
  15. trymes

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    Allan,

    Your setup sounds more complicated than the average run-of-the-mill PRI setup. I would call Sangoma for support, as they have excellent support, and they will be far better able to help you than I am. Be prepared to provide SSH access to the machine for them.

    Tom
     
  16. dicko

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    If you carrier requires ss7 signaling, unfortunately for you Elastix does not support that, you will need to "compile Asterisk from source" (and a little bit more) ss7 is not for the "faint of heart".

    http://www.voip-info.org/tiki-index.php ... terisk+SS7

    Sangoma's wanpipe/wanrouter driver have that support compiled but, that would be for Sangoma to let you know how to integrate it with Elastix/Asterisk
     

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