Voip provider account problems signaled as busy line

  • Dear Starface team,


    We are using voipstunt as voip provider for international calls.
    We have several accounts and they are all set up in the cost optimized routing so that if first one is occupied the second call goes over second account etc.
    However it seems if all accounts are busy if we try to place another call, we get a busy line.
    This is a bit of a problem as then we are not sure if the accounts are exhausted or the remote party is indeed busy.
    In the asterisk log we also see a difference, if the remote party is busy then in the logs explicitey says something like "remote pary busy" or such, whereas then the accounts are exhausted then there is no such entry.
    So asterisk knows the difference so the starface should know it also.


    Is there any way to replace the busy signal with a meaningful message?


    Thanks in advance,
    Zsolt

  • Hi Zsolt,


    which device returns "busy"? A SIP-Client or your ISDN-Phones? The default return code should be "Declined" or something like that, but not "busy". Perhaps your ISDN-Phones misinterprets this return status...


    Regards,
    Philipp

    Gruß / Regards
    Philipp

  • We tested with HW phones only, I will repeat the test with SIP client.


    Now, sorry for mixing up the things a bit, but I will post it here, maybe it is not worth to start a new topic...
    If I call a reguler phone over ISDN and the remote party is busy, when I call from SIP client, the connection starts, aborts with a klick and I get
    Call rejected: 503 Service Unavailable
    but if I call from starface call gui or from the SF adressbook then I get a busy signal...
    Is that OK?
    Thanks
    Zsolt

  • so to the matter of getting busy line on teh ISDN phone if the line is indeed not busy but has some other problems like exhausted number of accounts or there is no credit on the account...
    thru SIP phone we get a message, like "there is no credit....", but on the ISDN phone we get only a busy line after a short delay.
    I'm not sure why can't you hand this message over to the PBX, but you will know better.
    I have the asterisk logs for both when calling thru SIP and when calling thru ISDN...
    Rgds
    zsolt

  • There is one more interesting thing, as I said, with SIP phone the call attempt ends with a message from voip provider, in the logs it ends with:
    -- Called orchri_1/00491705416754
    -- SIP/orchri_1-08dd1408 is making progress passing it to SIP/zsoltXllite-08d958e0
    == Spawn extension (international, 00491705416754, 1) exited non-zero on 'SIP/zsoltXllite-08d958e0'


    with HW phone, the call ends with busy line after a delay, but from the logs it is visible that it tries first with the second voip account, i.e. it fails over once...?? the first call ends with this:
    -- Called orchri_1/00491705416754
    -- SIP/orchri_1-08dd1860 is making progress passing it to Zap/11-1
    -- Got SIP response 480 "Temporarily not available" back from 194.120.0.198
    -- SIP/orchri_1-08dd1860 is circuit-busy
    == Everyone is busy/congested at this time (1:0/1/0)


    then it fails over to second account and that call ends with:
    -- Called zsmagyar/00491705416754
    -- SIP/zsmagyar-08de6d50 is making progress passing it to Zap/11-1
    -- Channel 0/2, span 4 got hangup request, cause 102
    == Spawn extension (international, 00491705416754, 1) exited non-zero on 'Zap/11-1'
    -- Hungup 'Zap/11-1


    note that the second voip account the zsmagyar DOES have enough credit to make the call...


    any idea
    thanks
    Zsolt

  • Zitat

    -- Called orchri_1/00491705416754
    -- SIP/orchri_1-08dd1408 is making progress passing it to SIP/zsoltXllite-08d958e0
    == Spawn extension (international, 00491705416754, 1) exited non-zero on 'SIP/zsoltXllite-08d958e0'


    Here is some kind of status missing from your Provider. Something like "answered" or "busy".
    This could also be the normal way of your provider telling you that the remote party is busy...


    In the second log your really get a busy back from your provider, it could also be a problem at your provider because of this message

    Zitat

    -- Got SIP response 480 "Temporarily not available" back from 194.120.0.198


    I'm not quite sure who is 194.120.0.198 - perhaps your provider...


    Then STARFACE takes the second provider of your COR-Rule and gets

    Zitat

    Channel 0/2, span 4 got hangup request, cause 102


    back, cause 102 means "recovery of timer expiry".


    Because of the posted logfiles I suppose that the remote party is really not available...


    Regards
    Philipp

    Gruß / Regards
    Philipp

  • Zitat

    Here is some kind of status missing from your Provider. Something like "answered" or "busy".
    This could also be the normal way of your provider telling you that the remote party is busy...


    As I said, we got a message that there is not enough money on the account, and this message can be heard on SIP phone, but if we call from HW phone then we do not hear this message, just get busy line after a break of about 5 seconds. So the question is why is this voice message not forwarded to the HW phone...?


    The second case is perfectly legal, the remote party is busy and we got busy line both on SIP client and on HW phone, so it is only the first case which is the problem
    Thanks
    Zsolt

  • Ok, in the first call I can't see a real progress of this call. The Provider only sends


    Zitat

    -- SIP/orchri_1-08dd1408 is making progress passing it to SIP/zsoltXllite-08d958e0


    In a normal call you see e.g. something like


    Zitat

    SIP/orchri_1-08dd1408 ist ringing


    and after answering the call you see


    Zitat

    SIP/orchri_1-08dd1408 answered SIP/zsoltXllite-08d958e0


    and then

    Zitat

    Attempting native bridge of SIP/orchri_1-08dd1408 and SIP/zsoltXllite-08d958e0


    So I guess because it's not a "real" call the audio streams are not being forwarded to your ZAP-Phones.


    BTW: A nice overview of ISDN+SIP hangup causes can be found here: http://www.voip-info.org/wiki/…risk+variable+hangupcause


    Regards
    Philipp

    Gruß / Regards
    Philipp

  • Hi Phillip, thanks for answers.
    At one point I would like to test this with you and see if we can come up with a solution, then I will post the outcome to the forum.
    Next week we should go productive and then I will contact you.
    Rgds
    Zsolt

Jetzt mitmachen!

Sie haben noch kein Benutzerkonto auf unserer Seite? Registrieren Sie sich kostenlos und nehmen Sie an unserer Community teil!