Call Progress Tone

Discussion in 'Windows' started by virgel07, Apr 10, 2012.

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  1. virgel07

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    Sir,

    good day!

    i recently bought Grandstream GXW4104 and i dont know the parameters to be set for call progress tone.
    i'm living here in the Philippines, can you please provide me the exact tones. Thanks a lot.

    1. Dial Tone:
    2. Ringback Tone:
    3. Busy Tone:
    4. Reorder Tone:

    I'm already tired searching the exact parameter tone.
     
  2. leejor

    leejor Well-Known Member

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    Busy tone should be 480 + 620 Hz 0.5 on/ 0.5 off

    This site has a database but you are restricted to only so many "free" searches within a given period.

    Put in the country then the specific tone, it doesn't seem to like you selecting "all".

    http://www.3amsystems.com/wireline/tone-search.htm
     
  3. virgel07

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    thanks for the reply, i followed the guide regarding the setup for grandstream gxw4104 and i can accept an inbound now, my problem is the outbound rule. i supposed to dial local # 9910551 from my extension sip 136

    Outbound Rule:
    Calls to Numbers starting with (Prefix) :
    Calls from extension(s) : 136
    Calls to Numbers with a length of : 7

    Make Outbound Calls on
    You can configure up to 3 routes for calls to go out on. More >The second and third route will be used if the previous one is unavailable. For each route, you can decide if any digits should be stripped and if any digits should be added. If you have configured a prefix, make sure to strip off the prefix. < Less

    Route 1 Grandstream_GXW4104 Strip Digits 0
    Route 2 Grandstream_GXW4104 Strip Digits 0
    Route 3 Grandstream_GXW4104 Strip Digits 0
     
  4. leejor

    leejor Well-Known Member

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    You should not have to put the same trunk group in more than once, just allow the trunk group to accept more than one call.
    Post the 3CX log of an outgoing call attempt.

    I'm not certain the rule(s) will work with out any prefix, at least, I've never set up any like that.
     
  5. virgel07

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    Good Day!

    Sir pls. elighten me on this, still i cannot make an outbound call eventhough inbound call is pretty much working right now.
    if i make an ountbound call, i got this server log,

    11:43:16.703 [CM503020]: Normal call termination. Reason: Server Failure
    11:43:16.703 [CM503016]: Call(5): Attempt to reach "99910551"<sip:99910551@192.168.18.1:5060> failed. Reason: Server Failure
    11:43:16.703 [CM503003]: Call(5): Call to sip:99910551@192.168.8.2:5060 has failed; Cause: 503 No route to host; warning: ; internal
    11:43:16.640 Can't find source interface to use
    11:43:16.593 [CM503025]: Call(5): Calling PSTNline:99910551@(Ln.10000@SGCPbx)@[Dev:sip:10003@192.168.18.2:5066;transport=udp]
    11:43:16.578 [CM503003]: Call(5): Call to sip:99910551@192.168.8.2:5060 has failed; Cause: 503 No route to host; warning: ; internal


    i set thins in 3cx server;

    Outbound Rule:
    Calls to Numbers starting with (Prefix) : 9
    Calls from extension(s) : 136
    Calls to Numbers with a length of : 8

    Route 1 SGCPbx Strip Digits 1


    I'm here in the philippines, i supposed to call local line 9910551 and i dialed in my sip phone 99910551.
    What's wrong with my outbound rule? or in my grandstream GXW4104 setup? i followed the setup guide posted in the other post. Pls. help me sir. thank you!
     
  6. leejor

    leejor Well-Known Member

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    Check your IP addresses. In one case you have 192.168.8.2 and in the other 192.168.18.1. Unless you have a subnet mask that will accommodate this, the devices won't "talk" to each other.

    I see three different IP address there, so I'm not sure what you have in your datafill.
     
  7. virgel07

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    Good Day!

    Thanks for the reply sir and also for giving me the tip regarding the ip address,

    11:43:16.578 [CM503003]: Call(5): Call to sip:99910551@192.168.8.2:5060 has failed; Cause: 503 No route to host

    i figure out this one and fix it, it seems that i wrongly inputed in the 3cx server the PSTN ip address, i got it now but still have difficulties here regarding the outbound call, i can make a call to other line but their is also line that i cannot connect.
    The 3cx log says that i connected but in my sip phone it only ring once then it give me a ringback which is sounds like a busy line and no body answer. at the other line also their is no alert tone that may hear.

    Sir, pls. guide me again on this, where i supposed to fix?

    Thanks a lot, i really appreciated your support.
     
  8. leejor

    leejor Well-Known Member

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    I'm not really clear on what you are attempting. Call to other line? Line you can't connect to? please post a log of the call(s) with more of an explanation.
     
  9. virgel07

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    16:38:29.687 Call::Terminate :[CM503008]: Call(6): Call is terminated
    16:38:20.187 CallCtrl::eek:nLegConnected :[CM503007]: Call(6): Device joined: sip:10003@192.168.3.2:5064;transport=udp
    16:38:20.156 CallCtrl::eek:nLegConnected :[CM503007]: Call(6): Device joined: sip:136@192.168.3.136:5060
    16:38:20.140 Line::printEndpointInfo :[CM505002]: Gateway:[Grandstream_GXW4104] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Grandstream GXW4104 (HW 2.0, Ch:10) 1.3.4.10] Transport: [sip:192.168.3.1:5060]
    16:38:17.187 CallCtrl::eek:nSelectRouteReq :[CM503004]: Call(6): Calling: PSTNline:10004@[Dev:sip:10004@192.168.3.2:5066;transport=udp, Dev:sip:10001@192.168.3.2:5060, Dev:sip:10002@192.168.3.2:5062;transport=udp]
    16:38:17.187 CallCtrl::eek:nSelectRouteReq :[CM503004]: Call(6): Calling: PSTNline:10004@[Dev:sip:10004@192.168.3.2:5066;transport=udp, Dev:sip:10001@192.168.3.2:5060, Dev:sip:10002@192.168.3.2:5062;transport=udp]
    16:38:17.093 CallCtrl::eek:nSelectRouteReq :[CM503004]: Call(6): Calling: PSTNline:10003@[Dev:sip:10003@192.168.3.2:5064;transport=udp, Dev:sip:10004@192.168.3.2:5066;transport=udp, Dev:sip:10001@192.168.3.2:5060, Dev:sip:10002@192.168.3.2:5062;transport=udp]
    16:38:16.953 CallCtrl::eek:nIncomingCall :[CM503001]: Call(6): Incoming call from Ext.136 to "9910607"[sip:9910607@192.168.3.1:5060]

    This is what i got sir, if i dialled e.g 9902174 their is no problem but if i call e.g 9910607 then it will ring once and then my sip phone is talking at the same time it sound a busy tone. Their is also no dial tone to the pstn line whom i call e.g 9910607
     
  10. virgel07

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    Good Day!

    Sir, i already got it, inbound and outbound call is working now but i just want to ask for the help also, the caller id is not really working, when i received inbound call, only the pstn no. assigned in the 3cx server appeared in my sip phone, not the actual pstn. no. Where i will be going to update sir? And also the audio is not stable, sometimes its not clear and echo.
     
  11. virgel07

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    Good Day!

    Sir, i already got it, inbound and outbound call is working now but i just want to ask for the help also, the caller id is not really working, when i received inbound call, only the pstn no. assigned in the 3cx server appeared in my sip phone, not the actual pstn. no. Where i will be going to update sir? And also the audio is not stable, sometimes its not clear and echo.
     
  12. leejor

    leejor Well-Known Member

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    First of all, be sure that you do indeed have caller ID on the line, it's amazing how many people expect it to work even if they do not pay for the feature.

    Once you've confirmed that another caller ID enabled phone, works (shows the CID), when plugged into the same line, then you may need to make some adjustments on your gateway.


    Caller ID is sent in different ways depending upon which country you live/which provider you use, I'm not sure which method is used in the Philippines. The setting in the gateway must match the method that your PSTN provider is sending .
    A common problem in North America (and other areas that use Bellcore/Telcordia CID), where the CID data is send between the first and second ring, is that the gateway is not set to wait long enough, before alerting 3CX of an incoming call. the Answer Delay (it's called by various names) must be at least 3 or 4 seconds.
     
  13. virgel07

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    Ok sir, i will check of it. Thank you!
     
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