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Thread: permanently engaged tones

  1. #1

    Default permanently engaged tones

    Hi All

    I am trying (failing in the main) to install a Quadro4X. I have about a million questions however the most pressing is that when I connect my two telephone lines they become engaged.

    BT (Openreach) have installed the two sockets which, when a POTS phone is connected allow me to ring between themselves on the same number (Aux lines). However when I connect to the incoming ports on the 4X they become engaged....

    Any idea's ?

  2. #2

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    Quote Originally Posted by miktdish View Post
    Hi All

    I am trying (failing in the main) to install a Quadro4X. I have about a million questions however the most pressing is that when I connect my two telephone lines they become engaged.
    Any idea's ?
    That was a show stopper then ...

    So here's another. I was under the impression that I could (for example) set up an account with SiPGate or Vonage and access the provider from the PBX, rather than from each of the phones ?

    For example dial '8' for SiPGate and '9' for a pots line

    Have I missed something ?
    p.s.
    still got the engaged tones with aux lines.
    Last edited by miktdish; 11-18-2009 at 04:08 AM.

  3. #3
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    Default

    Quote Originally Posted by miktdish View Post

    So here's another. I was under the impression that I could (for example) set up an account with SiPGate or Vonage and access the provider from the PBX, rather than from each of the phones ?

    For example dial '8' for SiPGate and '9' for a pots line

    Have I missed something ?
    The PBX can work both ways. You can have each extension have its own account at ITSP, or have a common accounts for everyone.
    To configure the latter, you would need to use "VoIP Carrier Wizard" (easy way), or manually tweak "Call Routing Table" (advanced way).

    To get better understanding, you can look first at your Call Routing Table (CRT), then pass through the "VoIP Carrier Wiazrad" and look at the CRT again. After you understand the structure of the CRT, you can manually add rules there.
    But basically to add a common account for all extension, the voip carrier wizard is enough.

    Quote Originally Posted by miktdish View Post
    still got the engaged tones with aux lines.
    What do you get when connecting just one FXO line, Mik?

    Best regards,
    David

  4. #4

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    Quote Originally Posted by davrays View Post
    The PBX can work both ways. You can have each extension have its own account at ITSP, or have a common accounts for everyone.
    To configure the latter, you would need to use "VoIP Carrier Wizard" (easy way), or manually tweak "Call Routing Table" (advanced way).
    David. Thanks for offering help. I ran the VoIP Wiz and set up the service although I am still a little confused as at the end it asks me to define where calls are routed to, and I only get a singe extension choice...
    It does work though so, after about 3 days I feel I am getting somewhere


    Quote Originally Posted by davrays View Post
    What do you get when connecting just one FXO line, Mik?
    Best regards,
    David
    I haven't revisited this as I brought the 4X into my office, away from the phone lines.

    I have three analogue lines I want to connect. Line 1 and 2 will be the Aux pair and Line 3 the "house" phone. Line 3 worked OK before and when I called it I got into the AA however Lines 1 and 2 were engaged when connected to the 4X but rang freely when disconnected.

    Now I am managing to get extensions registered and have the SiPGate account set up I need to get the unit back into the rack and connected into the LAN.

    Therein lies another story .....

    At first I tried to connect everything via the WAN port (without success, but I did learn to scream and run in circles). Having perfect the ever diminishing circular scream I brought the 4X down into the office (well it will be my office once I get these phones sorted out) and bought a small hub/switch so I could connect more than 1 phone (SNOM 370's) to the LAN port.

    After more screaming, a fair bit of gurgling and several factory resets I now have 3 working extensions on the normal 172.30 LAN and am connected to the Draytek/router via a Netgear switch.

    Plan A is to set up 2 VLans on the switch and connect the 4X between the 2 of them.
    This should allow me to continue to get phone services (on 172.30) and allow me to operate a 192.168 LAN on which will be (at some time in the near future) my Domain/Exchange/File Server (Win 2003) will reside.


    Hopefully (if I haven't exhausted my will to live) I will be au-fait with the Auto Assistant, have installed the rest of the phones (inc some Snom M3's) and will be able to route incoming calls from different phone lines to different extensions and be able to make outgoing calls on predetermined lines by employing least cost routing rules or simply by preceding the pstn number with a certain digit.

    Any help and assistance to attaining this lifetime achievement is gratefully accepted.
    Bestest
    Last edited by miktdish; 11-18-2009 at 06:41 PM. Reason: spelling and adding bits

  5. #5

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    The fog cleareth ...

    So I now have a SiP account which rings all the phones anytime a call comes in and any phone can press '0' and they get out on the SiP line. Progress .....

    Or so I thought ... but the screaming has started again.
    New thread started "MERg does and doesn't"
    Last edited by miktdish; 11-19-2009 at 04:13 AM. Reason: update

  6. #6
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    Nice, though a bit harrowing story..
    This is how the progress generally goes forward... I'm glad that things are getting better, the fog is clearing, and the sun is starting to emerge from behind the clouds..

    I'll have a look at your new thread to see how it goes..

    Best regards,
    David

  7. #7

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    Quote Originally Posted by miktdish View Post
    I have three analogue lines I want to connect. Line 1 and 2 will be the Aux pair and Line 3 the "house" phone. Line 3 worked OK before and when I called it I got into the AA however Lines 1 and 2 were engaged when connected to the 4X but rang freely when disconnected.

    Now I am managing to get extensions registered and have the SiPGate account set up I need to get the unit back into the rack and connected into the LAN.

    I have relocated the 4X back up to the rack room and connected the phone lines.
    These appear to work although I did notice that the 4X refused to drop the incoming aux-line call despite the fact that I had replaced the handset.

    I rebooted and that erratisism seems to have stopped ...

    All is not 100% well though as when I press the "help" button I just get an empty daughter window, I haven't rebooted in 5 hours, must be overdue ...


    Like the VoIP wizard is there a POTS wizard ??

    I see in the System Config Wizard the initial FXO access rules are set (dial 9 etc).

    What I want to do is set up FXO1 and FXO2 to ring a number of extensions as they are the main business lines (same number). The third FXO is a private line and I don't want this to be picked up by dialling '9'.

    Is it possible to allocate another number '7' for example ?

    Thanks in advance for any replies

  8. #8

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    "private line"s can be achieved by using call routing pointing to the fxo port with pbx and the extension as the identifier ...

    All other extensions would be excluded...

    So if you wanted lets say extension 15 to dial out on fxo4 as a preference...

    You could in call routing have 2 entries like the following :

    I dial 0 for an outside number here in Australia make it what you want...

    Routing Call Type - Add Entry


    Destination Number Pattern: 0,Number of Discarded Symbols: 1, Destination Type: FXO, Filter on Source / Modify Caller ID: Ticked

    Routing Call Settings - Add Entry


    Select the FXO port for the private line

    Source Filter / Modify Caller ID - Add Entry

    Select Extension number instead of * ie 15 and ensure that PBX is selected

    Finish...

    Then create another call route except this time change the following variable only.

    Under :

    Routing Call Settings - Add Entry


    Select the FXO ports and do not select the private line FXO port, make sure this is unchecked.

    What the above achieves is :

    Anyone dials 0, all calls will call out on the FXO ports excluding port 4 being the private line. If extension 15 dials 0, he will be offered FXO port 4 first before dialling out on the remainder ports.


    Substitue 0 with a 9 if you dial 9 for an outside line.

    Make sure that these rules are above any others with the same dial pattern otherwise it will not work.


    Kevin
    Last edited by KSComs; 11-25-2009 at 05:49 AM.

  9. #9

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    Quote Originally Posted by KSComs View Post
    "private line"s can be achieved by using call routing pointing to the fxo port with pbx and the extension as the identifier ...
    Make sure that these rules are above any others with the same dial pattern otherwise it will not work.
    Kevin
    Kevin - thank you. With your and Davrays help I am now starting to glean meaning and results from, what 2 weeks ago, may well have been a foreign language

    I appear to be almost there (with routing etc) in that I can dial "0" to get the SiP, "9" to get the PSTN lines and "7" to get the house line.

    I have the incoming ringing all phones although i'm a little surprised that the calls are coming from "anonymouse" - is there a setting I have missed ?
    In the FXO lines I have filled in the number field etc .. anything else I need to do or is this normal ?

  10. #10
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    Quote Originally Posted by miktdish View Post
    I have the incoming ringing all phones although i'm a little surprised that the calls are coming from "anonymouse" - is there a setting I have missed ?
    In the FXO lines I have filled in the number field etc .. anything else I need to do or is this normal ?
    This should not be the case - if you enabled the Caller ID service from telco, you should see the correct callerID passed to the called phones.

    If you open "Telephony"->"Call Statstics" page, do you see the correct caller ID there? If not, then what is shown? (please copy and show here the caller details from the table)

    Best regards,
    David

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