IVR and direct dial issues - URGENT

Discussion in 'General' started by coyoty, Mar 19, 2010.

  1. coyoty

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    Hello to all

    I have an issue with IVR and direct dial. When someone calls at the office the IVR will answer the call and because i have "enable direct dial" checked on the IVR menu if someone knows the internal he can dial it and the extension rings.

    So far so good but this only works well when the person calling is using a mobile phone. If someone tries to do that from a land line he won't be able to reach the dialed extension.

    To be more specific when i tried to test this from home this is what happens.

    Call from the mobile:

    I dial the office -> IVR answers -> message is played and before the message is over i start to dial my extension. The moment i press the first button on my mobile (143 is my extension and as soon as i press the 1) i can no longer hear the message. I dial the remaining digits (43) and i got transfered to my extension

    Call from my landline

    I dial the office -> IVR answers -> message is played and before the message is over i start to dial my extension. The moment i press the first button on the phone (same 143 extension - as soon as i press the 1) i can stil hear the message but the voice is cracked. I dial the remaining digits (43) and i still hear the rest of the message played but the sound is with gaps and cracks. At this point the system failed to forward the call to the desired extension.

    I need to know if there is a bug on asterisk or on elastix and if there is a fix to it. It is of vital importance as the office accepts on average more than 300 calls per day.

    I have been reported this issue plenty of times and no matter how much i tried to google an answer i haven't been able to find a solution.


    I am running Elastix 1.6-12 64bit on an HP ML350G5 Server

    Thank you for reading!
     
  2. Chilling_Silence

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    How are your landline calls coming in? It could be a way in which the DTMF tones are being handled?

    I'm presuming your IVR menu works fine from a landline, for example if you choose Option 1) to go through to Sales (Just for example), it works?
     
  3. Amphibian

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    coyoty,

    Try setting up a inbound route, with a standard IVR, without using the "enable direct dial", and see if everything works like it is supposed to both landline and cell.

    If it doesn't then you have an issue with the DTMF levels either within your card or with your service provider and I would suspect your card (because we all know that phone companies have their stuff together).
    One thing to remember is that in most cases (in the USA, don't know where you are at) the DTMF level coming from a cell phone is at a lower db level then the exchange direct is.

    Additionally, have you try doing an inbound call from landline through VoIP?

    I suspect that you have a issue with your PSTN Interface card. I would probably change that out first if you have a spare one and save yourself a lot of grief.

    Hope this helps
    Amphibian
     
  4. coyoty

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    There are 5 adsl pirelli routers and each router has 2 phone jacks at the back which are connected to the Elastix server. There is a total of 10 phone lines (5 routers x 2 phone jacks). The Interface card is an Openvox A1200 with 10FXO modules installed.

    [/quote]I'm presuming your IVR menu works fine from a landline, for example if you choose Option 1) to go through to Sales (Just for example), it works?[/quote]

    Sometimes it works sometimes it doesn't. At my home i have 1 pirelli router and a standard wireless phone connected. When i tried to call my company i failed each and every time to get through at the dialed extension


    Any ideas??
     
  5. coyoty

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    No i haven't tried to call my office from a voip provider. I will try that and come back to you.

    My hardware is an Openvox A1200 with 10fxo modules installed (for 10 pstn lines)

    This is how my IVR is designed:

    IVR1 message played:
    welcome to my company for english please press 1

    settings:
    timeout = 7
    Enable direct dial: yes
    repeat loops: 1

    if 1 is pressed it goes to the english IVR
    for I (invalid extension) goes to secretary ring group
    for t goes to next IVR in greek

    assuming that 1 was pressed:

    IVR2 message played:
    if you know the extension number please dial it know otherwise please hold

    settings:
    timeout = 7
    Enable direct dial: yes
    repeat loops: 0

    for I call is routed to a specific extension
    for t call is routed to a specific extension


    Now if at IVR1 the caller would hold and didn't press 1 and assuming he doesn't know the extension the call is routed to IVR3

    IVR3 message played:
    if you know the extension number please dial it know otherwise please hold (the message now is in greek)

    settings:
    timeout = 7
    Enable direct dial: yes
    repeat loops: 0

    for I (invalid extension) goes to secretary ring group
    for t (timeout) goes to secretary ring group
     
  6. ramoncio

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    It looks as if your telco provider is already using VoIP to give you telephony services. They will be giving you g711 audio with inband DTMF, so you need to get the highest audio quality for incoming DTMFs to be properly detected. Have you run fxotune in your lines? This should improve your audio quality. You can also play with gains to see if the detection is improved.

    It would be ideal if you could configure your Elastix as an endpoint for their VoIP services, instead of the router. But that's very strange, they usually will not give you the configuration data, nor support to configure your Elastix end.

    In this case I would suggest you to migrate your numbers to full SIP (you will need to find a provider who can offer you migrate your current DIDs), or to get real digital lines (BRI or maybe some T1 channels?, I've heard of people using just 10 digital channels in a E1).
     
  7. coyoty

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    Hi!

    Thank you for your advice.

    This is what i did:

    amportal stop

    fxotune -i

    It successfully tested and configured all dahdi channels but at the end it gave me several messages saying:
    /dev/dahdi/1 absent: no such file or directory
    /dev/dahdi/2 absent: no such file or directory
    /dev/dahdi/3 absent: no such file or directory
    *
    *
    *
    /dev/dahdi/252 absent: no such file or directory

    Is this normal?

    I will test the system (and i am sure that the users will let me know if it works ok or we still have a problem) and let you know if we cracked this one.


    When you say about playing with the gains. How do i do that?

    Yes you are correct again. It would be ideal to configure elastix as an endpoint to their services but they are not willing to give any info at all....

    We are stuck with the voip solution behind routers because it is the most economical one


    Thank you for your time!!
     
  8. fmvillares

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    for using fxotune you need to first stop the asterisk service amportal stop
    then fxotune -i and a number to break the tones
    for instance fxotune -i 4
    then fxotune -s to save configs
    then restart amportal
    but one advice the problem is carried across your voip provider...becasue it has this scenario

    pstn --- provider ----- voip ------ your analog "opencrap" card ---- voip again

    the best scenario will be that your provider send to you the direct sip connection to avoid the dual conversion

    and the scenario will be

    pstn ---- provider ----- voip ----your asterisk in sip native ---- your extensions and voilaaa

    best regards
     
  9. coyoty

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    ok this is what i did:

    amportal stop

    (stops ok)

    fxotune -i 4

    configs each dahdi separately and says done! every time it finishes. When it reaches the last it says done and then the following messages:

    /dev/dahdi/1 absent: no such file or directory
    /dev/dahdi/2 absent: no such file or directory
    /dev/dahdi/3 absent: no such file or directory
    *
    *
    *
    /dev/dahdi/252 absent: no such file or directory

    I then typed

    fxotuned -s

    (seems to be executed)

    Am i doing something wrong?

    The problem still remains. People calling from landlines some times they get the dialed extension but most of the times they don't.

    People calling from mobile phones get the dialed extension just fine.

    Is there anything else i can do??

    I am getting desperate here...


    I just got off the phone with the technical support of my provider and they said that they currently don't support sip. It seems that i am stuck with what i have and i really need to solve this.

    Thank you very much for your time!!
     
  10. coyoty

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    Some updates on the subject

    I have being trying to solve this with our provider.

    They asked me if i can prioritize the codecs asterisk is using. For example if i can set as first codec the G711a and as a second G729. Would that help? Is this possible to prioritize them?

    It seems though that asterisk cannot understand the DTMF tones only from certain providers NOT all of them...

    Any ideas?
     
  11. fmvillares

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    read about disallow and allow sip config lines and dtmfmode....search voip-info.org and you will be very surprised
     
  12. Amphibian

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    What country are you in?
     
  13. coyoty

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    In northern Greece.
     
  14. coyoty

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    Updates on the subject


    After a long discusion with the technicians of my provider and some test we came up to the following conclusion:

    1. Call from a simple phone (from our national telephony provider) connected to a landline = success

    2. Call from a phone from their network (behind a pirreli router) = success

    3. Call from a phone behind an ericsson pbx = success

    4. Call from a phone from behind an alcatel pbx = unsuccessful (looks like the pbx is choking the DTMF tones)

    5. Call from another provider (behind an unknown router) sometimes successful sometimes not.

    6. Call from my home behind a pirelli router same provider as at the office = unsuccessful

    Next step is to try and monitor the DTMF tones that arrive on Asterisk and see what happens from there.


    If add these lines at chan_dahdi.conf will it make a difference?

    dtmfmode = rfc2833
    rfc2833compensate = yes

    The routers connected at the PSTN card of the pbx don't support rfc2833. Could that be causing the problem at first place??

    Thank you
     
  15. coyoty

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    Update on the same day:

    I did change chan_dahdi.conf and i added the following lines:

    dtmfmode=rfc2833
    rfc2833compensate=yes

    It seems that the situation has improved dramatically to the best since now i am able to call from home and successfully reach the dialed extension. I also called some of the people i know and they have reported that problem to try it out again and it seems to be working much better.

    I am not saying it works 100% because out of 5 calls i have made from home 4 were successful and 1 failed. Remember that from home it was 100% fail.

    I am pretty sure that it is the routers fault since it doesn't support rfc2833. My provider said that they will send me another router that is fully compliant with rfc2833 and if it is successful i will change them all.
     
  16. fmvillares

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    well its good to knnow that with proper knowledge and guidance you reach to the objective that i expect...understand the problem and try to use tools at your sight to solve it...
     
  17. coyoty

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    I want to say a big thank you to all the people who read my thread and tried to guide me to a solution. I do not consider this thread closed yet since i am waiting to see how it will perform next week. I might have got some positive answers now but the amount of calls the pbx handles is quite big.

    THANK YOU ALL!!!

    I will keep you posted if something new comes up
     

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