D-Link DVG-1120S - Ok - I give up

Discussion in '3CX Phone System - General' started by SmokeyCarr, Jan 2, 2008.

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  1. SmokeyCarr

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    I think I am overlooking something very simple here. I have tried a couple of VoIP gateways but I am having an issue with the D-Link device I need to get working on my test bench. I have tried changing the PSTN prot from 5060 to 5061 to no avail. I am simply trying to call from ext. 1000 to the VoIP gateway ext. 10003 via the D-Link. I even tried calling from the PSTN device to ext. 1000 to no avil. Either I get a fast busy tone from the Polycom (ext. 1000) or I get an empty sound from the PSTN device (ext. 10003). I t seems I do not have a route to the host - so, this is telling me I am missing something various obvious which I cannot see at the moment. I am getting this:

    23:57:23.985 Call::Terminate [CM503008]: Call(154): Call is terminated
    23:57:23.970 Call::RouteFailed [CM503014]: Call(154): Attempt to reach [sip:10003@10.1.10.21;user=phone] failed. Reason: Not Found
    23:57:23.954 CallLeg::eek:nFailure [CM503003]: Call(154): Call to sip:0003@10.1.10.205:5061 has failed; Cause: 404 Not Found; from IP:10.1.10.205
    23:57:23.393 LineMgr::eek:nRegUpdAor Registered line: L:10003(DLINK DVG 1120S)
    23:57:23.128 MediaServerReporting::SetRemoteParty [MS210002] C:154.2:Offer provided. Connection(transcoding mode): 10.1.10.21:7418(7419)
    23:57:23.112 CallCtrl::eek:nSelectRouteReq [CM503004]: Call(154): Calling: PSTNline:10003@[Dev:sip:10003@10.1.10.205:5061]
    23:57:23.066 CallCtrl::eek:nSelectRouteReq [CM503010]: Making route(s) to [sip:10003@10.1.10.21;user=phone]
    23:57:23.066 MediaServerReporting::SetRemoteParty [MS210000] C:154.1:Offer received. RTP connection: 10.1.10.203:2230(2231)
    23:57:23.066 CallLeg::setRemoteSdp Remote SDP is set for legC:154.1
    23:57:23.066 Extension::printEndpointInfo [CM505001]: Ext.1000: Device info: Device Identified: [Man: Polycom;Mod: SoundPoint IP 330;Rev: General] Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [PolycomSoundPointIP-SPIP_330-UA/2.2.0.0047] Transport: [sip:10.1.10.21:5060]
    23:57:23.050 CallCtrl::eek:nIncomingCall [CM503001]: Call(154): Incoming call from Ext.1000 to [sip:10003@10.1.10.21;user=phone]
    23:57:22.941 CallLeg::eek:nNewCall [CM500002]: Info on incoming INVITE:
    INVITE sip:10003@10.1.10.21:5060;user=phone SIP/2.0
    Via: SIP/2.0/UDP 10.1.10.203:5060;branch=z9hG4bKdfecf003790A969C
    Max-Forwards: 70
    Contact: [sip:1000@10.1.10.203:5060]
    To: [sip:10003@10.1.10.21;user=phone]
    From: "Home Office"[sip:1000@10.1.10.21];tag=87E5CCD9-E5880B82
    Call-ID: 502f436e-6d0aae0f-44ca2dc8@10.1.10.203
    CSeq: 2 INVITE
    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
    Proxy-Authorization: Digest username="1000",realm="3CXPhoneSystem",nonce="12843734242:0632cb43d65e4c40988a38eed81c786d",uri="sip:10003@10.1.10.21:5060;user=phone",response="ea0102a7e5487f76cbf25c97bf4d4a59",algorithm=MD5
    Supported: 100rel, replaces
    User-Agent: PolycomSoundPointIP-SPIP_330-UA/2.2.0.0047
    Allow-Events: talk, hold, conference
    Content-Length: 0

    Cordially,

    Smokeycarr
     
  2. Henk

    Henk Member

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    Looks like your ext 1000 is listening to sip port 5060
    and your extension 10003 is listening to sip port 5061

    Change the sip ports to be the same on both devices, and tell 3cx about the sip port of your gateway.


    3CX needs to know the SIP port of you D-link, keep the on 5060.
    The polygocm phone needs register server port to be 5060 (same as 3cx server) If you put all the SIP ports on the same eg 5060 it should work.

    You might have to reboot your devices (and 3cx) as the port might "fixed" and only resets as part of a reboot.

    H.
     
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  3. SmokeyCarr

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    Hello Henk,

    Thank you very much for your input. Yeah, I tried having it on all ports so I decided to try different combinations. This is probably what you are seeing in my previous post. So, every device is on the same port and I am still having the problem.

    Carl
     
  4. SmokeyCarr

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    Here is another listing using all on 5060:

    08:45:34.257 Call::Terminate [CM503008]: Call(156): Call is terminated
    08:45:34.226 Call::RouteFailed [CM503014]: Call(156): Attempt to reach [sip:10003@10.1.10.21;user=phone] failed. Reason: Not Found
    08:45:34.226 CallLeg::eek:nFailure [CM503003]: Call(156): Call to sip:0003@10.1.10.205:5060 has failed; Cause: 404 Not Found; from IP:10.1.10.205
    08:45:34.101 MediaServerReporting::SetRemoteParty [MS210002] C:156.2:Offer provided. Connection(transcoding mode): 10.1.10.21:7426(7427)
    08:45:34.101 CallCtrl::eek:nSelectRouteReq [CM503004]: Call(156): Calling: PSTNline:10003@[Dev:sip:10003@10.1.10.205:5060]
    08:45:34.054 MediaServerReporting::SetRemoteParty [MS210000] C:156.1:Offer received. RTP connection: 10.1.10.203:2234(2235)
    08:45:34.054 CallCtrl::eek:nSelectRouteReq [CM503010]: Making route(s) to [sip:10003@10.1.10.21;user=phone]
    08:45:34.038 CallLeg::setRemoteSdp Remote SDP is set for legC:156.1
    08:45:34.038 Extension::printEndpointInfo [CM505001]: Ext.1000: Device info: Device Identified: [Man: Polycom;Mod: SoundPoint IP 330;Rev: General] Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [PolycomSoundPointIP-SPIP_330-UA/2.2.0.0047] Transport: [sip:10.1.10.21:5060]
    08:45:34.038 CallCtrl::eek:nIncomingCall [CM503001]: Call(156): Incoming call from Ext.1000 to [sip:10003@10.1.10.21;user=phone]
    08:45:34.007 CallLeg::eek:nNewCall [CM500002]: Info on incoming INVITE:
    INVITE sip:10003@10.1.10.21:5060;user=phone SIP/2.0
    Via: SIP/2.0/UDP 10.1.10.203:5060;branch=z9hG4bKf6dfa4e5B0822CAE
    Max-Forwards: 70
    Contact: [sip:1000@10.1.10.203:5060]
    To: [sip:10003@10.1.10.21;user=phone]
    From: "Home Office"[sip:1000@10.1.10.21];tag=26BC64DB-808D8FB4
    Call-ID: dc9959e0-fff12b31-9c55d21a@10.1.10.203
    CSeq: 2 INVITE
    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
    Proxy-Authorization: Digest username="1000",realm="3CXPhoneSystem",nonce="12843765933:d991271dbafcc0a365957a8bb0c9c2e0",uri="sip:10003@10.1.10.21:5060;user=phone",response="4800d12fb2a7af86d5273d9855d14af3",algorithm=MD5
    Supported: 100rel, replaces
    User-Agent: PolycomSoundPointIP-SPIP_330-UA/2.2.0.0047
    Allow-Events: talk, hold, conference
    Content-Length: 0


    08:45:02.863 CallMgr::DumThread::thread [CM100002]: *!* Exception detected: Missing header Contact *!*
    08:44:33.143 LineMgr::eek:nRegUpdAor Registered line: L:10003(DLINK DVG 1120S)
     
  5. Nick Galea

    Nick Galea Site Admin

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    You should use a supported modern gateway, such as the grandstream, patton or audiocodes....
     
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  6. SmokeyCarr

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    Yeah, perhaps you are right. Unfortunately, I have to try out various equipment types for our installation manual.

    I managed to get an IP phone to call a PSTN phone through the gateway I now have to figure out how to get the PSTN to ring. Same issue with the RTP300-NV I am testing.

    Thanks for your input.

    Cordially,

    Carl
     
  7. Nick Galea

    Nick Galea Site Admin

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    considering the cost of a new modern gateway, its just not worth your time...... A grandstream can be had for about $200
     
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  8. SmokeyCarr

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    Ok. We agree with you. We will just go to market with a packaged offering based on your VoIP gateway tested products. Sometimes you create more of a support nightmare trying to please all.

    One last question on thispost - Why do you not have any internal devices listed e.g., TDM type cards? It would be nice to have your product loaded on 1U's with everything sold in one box.

    Carl
     
  9. Pentangle

    Pentangle Member

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    I think we'd all agree there SmokeyCarr, but the issue is in getting the add-in cards to exist as though they are separately IP addressable units accessed via SIP. Most if not all of them don't act this way and expect you to use drivers etc.
     
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