Least Cost Router - Help Request

Discussion in 'General' started by ismed, Dec 2, 2009.

  1. ismed

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    Dear Experts,
    I want to build and placed a Least Cost Router between existing PABX and ISDN E1 PRA PSTN, the LCR server will do a splitation between the long distance call and local calls where the long distance will go thru the IP (internet) and terminated by the VoIP gateway and the locall call will go to the PSTN. When possible the LCR server can do the dtmf dial to the voip gateway thru the pstn as well. All ANI/Caller ID from PABX should be able to be passed thru to the PSTN so that the mobile destination can read it and the all ANI/ Caller ID from PSTN can be passed thru to the PABX so that can be read at the PABX extensions.

    Can you help for the solution?

    Awaiting for your respond soon and thank you.

    Best regards,
    Ismed
     
  2. dicko

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    That's quite simple, you will need two E1 interfaces, one signaling PRI-CPE (customer premise equipment) to connect to your provider, one signaling PRI-NET (network) to connect to your PBX (all other E1 and ISDN parameters should match the ones you currently use on the PBX, the clock setup should slave the PBX E1 to the Provider E1 (the master clock source), the context of the Provider facing trunk should be "from-pstn", that of the PBX facing E1 should be "from-internal" to allow outbound calls (and access to all the other FreePBX goodies) through the TDM/VOIP providers. Don't try and over-ride any CID that the PBX currently provides with that which the the provider might reject.

    Otherwise I believe all your other answers are in "Elastix Without Tears" for how to route your calls (in-bound routes ( to the PBX connected trunk), out-bound routes to your outbound trunks, either to E1 (local) or your chosen VOIP provider(s) (LD) ) .

    http://www.voip-info.org has a whole bunch more specifics. as to trunking and ISDN.

    Good luck

    dicko

    p.s. You can even add SIP hard/soft phones to the mix to supplement the restrictions of your current PBX.
     
  3. ismed

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    Thanks a lot dicko for your quick respond.
    Can you guide me please for each step?
    I already got the elastix without tears but it will take much time to go through, so i think i prefer to get direct guidance from you, especialy for PSTN parameter including ISDN PRI, setting up the inbound and outbound routes, and so on. I already have the 2xE1 port card from OpenVox now and ready to implement it.

    Awaiting your respond soon.

    Best regards,
    Ismed
     
  4. dicko

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    :) :) , nice try!!

    Sorry but if you won't spare the time to do your homework, I won't spend the time to do it for you.

    Surely, if you don't know how it works how will you ever troubleshoot/fix it.

    (My normal consulting charge is $190.00 per hour, here I contribute for free, time is however money and you Reading TFM is a prerequisite for me to get further involved)

    regards, dicko

    Please come back after doing that RTFM thingy.
     
  5. ismed

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    Sorry Dicko,
    Frankly speaking I already read them some and I promise and believe that I will be able to troubleshoot later on. And when necessary I will read them again all from first page :). I will do all the homework. So please guide me how to start it. Do you want to talk through email, YM or Skype?

    Thanks
    Ismed
     
  6. ismed

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    Question for Dicko,

    1) How many consulting hours to be consumed to finish this project in Elastix?
    2) What is the TFM (The Fxxxxxg Manual) and RTFM stand for?

    Awaiting for your respond soon.

    Thanks
    Ismed
     
  7. dicko

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    You don't have to apologize, I guess you haven't read many of my posts, As far as I'm concerned, if you haven't Read The manual (all of them, at least twice), then you posted here way too soon.

    Start by reading the manual of your PBX, familiarize yourself with how that works, translate all that stuff into Asterisk/FreePBX/Elastix stuff, Spend a few hours at http://voip-info.org to understand how E1's and ISDN work. and then follow my first post's suggestion, which are just the most common gotchas for first-timers, I've been caught by all of them myself. It's not brain surgery, but neither is it something you want to get into unless you have a deeper understanding than your . . .but it will take much time to go through. . . suggests you are prepared to invest.

    Also I don't skype, email or even twitter. I do read and sometimes respond to posts here though.

    [edit]

    RTFM = Read The Fucking Manual
    TFM = THE Fucking Manual
    EWT = Elastix Without Tears

    20 hours if you haven't RTM, N/C if you have and I can help clear up a few loose ends

    :) :)
     
  8. ismed

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    thanks again dicko, i thought those are kamasutra :) however that is something like EWT.
    What I mean for the existing PABX is actually the Cisco Call Manager, Siemen PABX and another one is Avaya. Do you think this is possible?
    What are the parameter to be done for Cisco Call Manager and the Elastix? Because I want to implement for this PABX firstly.

    Awaiting your respond soon.

    Thanks and regards,
    Ismed
     
  9. dicko

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    PABX: Private Automatic Branch Exchange,

    In the traditional world this was limited to TDM/analog circuits, and unless otherwise noted is generally still assumed to be just that. In your case you talk Cisco, Siemens and Avaya, All these can be VOIP enabled.

    I believe the Cisco talks TDM, H323, SCCP and SIP depending on the software load, hardware configuration and licenses you have purchased, it is perfectly possible, as it is should be be with anything that claims to be a PBAX, (including Asterisk) . That said however, there is no particular answer to your question, as each trunking needs a different treatment, SCCP is Cisco proprietary and is a PITA to get working with non Cisco stuff, H323 is kind of a "red-headed step child" in VOIP so also problematic, SIP is easy but expensive in Ciscoland, TDM (E1/T1) are nearly as old as AT&T and just work, again I suggest you RT(heir)FM, and have knowledge of what you actually have hard/software wise. Similar arguments with the Seimens and Avaya (AT&T)

    Elastix is more than capable to handle 99% of most users needs, however you will need to learn to ride without training wheels when you play with these big guys, no GUI's here or auto provisioning. It's all down to "knowledge is king" I suggest you absorb more of that knowledge before you get your fingers burnt.

    I stand by my original post as to the basics, but substitute your choice of trunking as appropriate.
     
  10. ismed

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    Hi Dicko,
    Since we are not allowed to reconfigurate either Cisco Call Manager, Siemens or the Avaya then connection from the said PABX to Elastix is only to be done by using E1 line connection not thru the SIP connection and the same to the PSTN.
    So do you think we need to enter some parameter into the Elastix configuration in order to recognize the call coming or going to the PABX?

    Awaiting your respond soon and thank you.

    Best regards,
    Ismed
     
  11. dicko

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    If your two E1's on the Elastix box are (as I suggested in my original post) in effect an in-line pass-through, then no, what I posted was the rudimentary necessities to make the trunk that points to the PBX look like the original TDM providers trunk, and conversely the Network facing E1 look like the PBX E1 interface. Under those circumstances everything will just work, just match parameter for parameter and get your timing/signaling right, get your trunks contexts right and the rest is just Reading TFM.

    p.s. you owe me 190 bucks, because you still haven't RTM (I do one hour minimum) please send it to palosanto, out gracious host/sponsor. B)
     
  12. ismed

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    Hi dicko don't worry please send the invoice or can I send you a bottle of beer?:cheer:
    Another question is how can the existing PABX/PBX will detect whenever the operator/ provider ISDN E1 PSTN line that connected to Elastix trunk line is drop or offline/cutoff and give the trouble signal to the PABX extension?
    Is that possible?

    Thanks
    Ismed
     
  13. dicko

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    Each trunk has a field for "Monitor Trunk Failures" it's FM says:

    If checked, supply the name of a custom AGI Script that will be called to report, log, email or otherwise take some action on trunk failures that are not caused by either NOANSWER or CANCEL.:

    So the answer is a reserved yes but you will have to write that script.
     
  14. ismed

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    Hi Dicko,
    Is the the belows dahdi_channels config correct?

    ;e1 trunk facing pstn
    ; Span 1: TE2/0/1 "T2XXP (PCI) Card 0 Span 1" (MASTER)
    group=0,11
    context=from-pstn
    switchtype = euroisdn
    signalling = pri_cpe
    channel => 1-15,17-31
    context = default
    group = 63

    ;e1 trunk facing pbx
    ; Span 2: TE2/0/2 "T2XXP (PCI) Card 0 Span 2"
    group=0,12
    context=from-internal
    switchtype = euroisdn
    signalling = pri_net
    channel => 32-46,48-62
    context = default
    group = 63

    another question is :
    1. what is the channel group of master or user? is that 63 for master?
    2. if we set the context as from-internal for user than should we threatened this trunk as extension?
    3. if we threatend the e1 trunk facing pbx as extension then how elastix captured the real isdn caller id to be sent to pstn that sent by pbx (as normally done by existing pbx)? (as i know in the analog line elastix will send extension cid)

    Waiting your respond soon and thank you.

    ismed
     
  15. dicko

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    Well, it's probably functional, but as I said the files is parsed sequentially, each keyword = value is stored until it meets a channel => <something> then the latest value of a keyword is used.

    Thus in your file you can use group 11 for calls to the PSTN, group 12 for calls to the PBX, group 0 is basically unusable as it includes all of the above group 63 remains undefined further, context = default is spurious.


    Given those comments, I don't understand your terminology in question 1, If you are asking in Question 2 if there is a security risk, then as it can only connect to extensions or other trunks in your pbx, depending on the PBX programming, only you can answer that. As to Question three, you have to ensure that you do not attempt to over-ride with unacceptable CID info any outbound CID info, if it currently works with the PBX it should be passed through transparently and effectively unless you try and overide, it might be necessary to add:

    callerid=asreceived


    There are other parameters that might need "tuning"

    Your reference here is:

    http://www.voip-info.org/wiki/index.php ... apata.conf

    in the CallerID options.
     
  16. ismed

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    Hi Dicko,
    I modified the config to follow your comment as follows:
    ;e1 trunk facing pstn
    ; Span 1: TE2/0/1 "T2XXP (PCI) Card 0 Span 1" (MASTER)
    group=11
    context=from-pstn
    switchtype = euroisdn
    signalling = pri_cpe
    channel => 1-15,17-31
    ;context = default
    ;group = 63

    ;e1 trunk facing pbx
    ; Span 2: TE2/0/2 "T2XXP (PCI) Card 0 Span 2"
    group=12
    context=from-internal
    switchtype = euroisdn
    signalling = pri_net
    channel => 32-46,48-62
    ;context = default
    ;group = 63

    Will the above config work?

    I want to clarify that what I mean for the channel group of Master is E1 trunk group that facing to PBX and User is E1 trunk group that facing to PSTN. So now I know that the group of Master is group=11 and group of User is group=12.
    And the remaining question is : is that true that we need to assign every channel of trunk group=11 to be zap extension in freepbx so that we can assign a route for inbound or outboud call?
    And the same for channel of trunk group=12 to be assigned as zap trunk?

    Awaiting your respond soon and thank you very much.

    Best regards,
    Ismed
     
  17. dicko

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    It will work if the parameters are right, I can't guarantee that it will as I don't know your trunk parameters.

    I would replace group=11 with group=0 and group=12 with group=1 for elegance.
    I am still unfamiliar with your words, "Group of Master" and "group of User", what exactly do you mean.


    Try and change your thinking from extensions and trunks in this scenario, think of them as "endpoints", they are all the same concept, just get your routing right.
    It's all a matter of "context", you have Network E1 in "from-pstn" and PBX E1 as "from-internal", so . . ..
    Whatever comes in on the Network facing trunk will be routed as per your "inbound route" setup, wild cards are acceptable (RTFM)
    Whatever comes in on the PBX facing trunk will be routed as per your "outbound routes" setup, wild cards are acceptable (RTFM)


    (Don't forget to set you mastrer slave timings in /etc/dahdi/system.conf)
     
  18. ismed

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    Hi Dicko,
    Further to your suggestions I polished a bit the config as belows :
    ;e1 trunk facing pstn
    ; Span 1: TE2/0/1 "T2XXP (PCI) Card 0 Span 1" (MASTER)
    group=0
    context=from-pstn
    switchtype = euroisdn
    signalling = pri_cpe
    channel => 1-15,17-31
    ;context = default
    ;group = 63

    ;e1 trunk facing pbx
    ; Span 2: TE2/0/2 "T2XXP (PCI) Card 0 Span 2"
    group=1
    context=from-internal
    switchtype = euroisdn
    signalling = pri_net
    channel => 32-46,48-62
    ;context = default
    ;group = 63

    Ok now to avoid confusion I agreed to you to use the term of E1 facing PSTN trunk and E1 facing PBX instead of Master Trunk and User Trunk.

    And now for the routing I just have a look to the unembeded freepbx and open the "Inbound Routes" setting for the network facing trunk as per your suggestion, I tried to follow the TFM EWT but stuck on the setting of "Destination" as I only can see there four choices : (1) Terminate Call (2)Extensions (3) IVR (4)Phonebook Directory, I cannot see there is opportunity to enter the Destination Trunk, I only can see the Destination Extension as per item (2).

    So is there a clue to route the inbound to the Network facing trunk?

    Awaiting for our respond soon and thank you.

    Best regards,
    Ismed
     
  19. ismed

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    Hi Dicko,
    The belows is my system.conf:
    # Autogenerated by /usr/sbin/dahdi_genconf on Thu Dec 10 21:18:52 2009
    # If you edit this file and execute /usr/sbin/dahdi_genconf again,
    # your manual changes will be LOST.
    # Dahdi Configuration File
    #
    # This file is parsed by the Dahdi Configurator, dahdi_cfg
    #
    # Span 1: TE2/0/1 "T2XXP (PCI) Card 0 Span 1" (MASTER)
    span=1,1,0,ccs,hdb3,crc4
    # termtype: te
    bchan=1-15,17-31
    dchan=16
    echocanceller=oslec,1-15,17-31

    # Span 2: TE2/0/2 "T2XXP (PCI) Card 0 Span 2"
    span=2,2,0,ccs,hdb3,crc4
    # termtype: te
    bchan=32-46,48-62
    dchan=47
    echocanceller=oslec,32-46,48-62

    # Global data

    loadzone = us
    defaultzone = us


    I cannot see there is a parameter to set the master slave timing?

    Awaiting your respond soon and thank you.

    Best regards,
    Ismed
     
  20. dicko

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    It is defined in that file (corrected), please actually READ my posts and follow my suggestions before asking what I have already answered

    quote from me in my first post here:

    . . . the clock setup should slave the PBX E1 to the Provider E1 (the master clock source),
    .
    .

    www.voip-info.org has a whole bunch more specifics. as to trunking and ISDN.
    .
    .

    So to save you your valuable time (and mine, less than a month ago I posted here http://www.elastix.org/component/option ... ,en/#40307, does that sound like a similar situation to yours?), please go to voip-info.org wherin, and hopefully after spending some time actually reading what is to be found there will find very specifically:

    http://www.voip-info.org/wiki/view/system.conf
    and very explicitly from that reference:

    ; If you choose 0, the port will never be used as a source of timing. This is
    ; appropriate when you know the far end should always be a slave to you. If the
    ; port is connected to a channel bank, for example, you should always be its
    ; master. Any number of ports can be marked as 0.
    ;
    ; Incorrect timing sync may cause clicks/noise in the audio, poor quality or failed
    ; faxes, unreliable modem operation, and is a general all round bad thing.
    ;

    It is crucial for you to better understand how things work before you try and get them working.
     

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