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

[SmokeyCarr]

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:[email protected];user=phone] failed. Reason: Not Found
23:57:23.954 CallLeg:nFailure [CM503003]: Call(154): Call to sip:[email protected]:5061 has failed; Cause: 404 Not Found; from IP:10.1.10.205
23:57:23.393 LineMgr: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:nSelectRouteReq [CM503004]: Call(154): Calling: PSTNline:10003@[Dev:sip:[email protected]:5061]
23:57:23.066 CallCtrl:nSelectRouteReq [CM503010]: Making route(s) to [sip:[email protected];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:rintEndpointInfo [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:nIncomingCall [CM503001]: Call(154): Incoming call from Ext.1000 to [sip:[email protected];user=phone]
23:57:22.941 CallLeg:nNewCall [CM500002]: Info on incoming INVITE:
INVITE sip:[email protected]:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.1.10.203:5060;branch=z9hG4bKdfecf003790A969C
Max-Forwards: 70
Contact: [sip:[email protected]:5060]
To: [sip:[email protected];user=phone]
From: "Home Office"[sip:[email protected]];tag=87E5CCD9-E5880B82
Call-ID: [email protected]
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:[email protected]: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

 

[Henk]

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.

 

[SmokeyCarr]

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

 

[SmokeyCarr]

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:[email protected];user=phone] failed. Reason: Not Found
08:45:34.226 CallLeg:nFailure [CM503003]: Call(156): Call to sip:[email protected]: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:nSelectRouteReq [CM503004]: Call(156): Calling: PSTNline:10003@[Dev:sip:[email protected]: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:nSelectRouteReq [CM503010]: Making route(s) to [sip:[email protected];user=phone]
08:45:34.038 CallLeg::setRemoteSdp Remote SDP is set for legC:156.1
08:45:34.038 Extension:rintEndpointInfo [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:nIncomingCall [CM503001]: Call(156): Incoming call from Ext.1000 to [sip:[email protected];user=phone]
08:45:34.007 CallLeg:nNewCall [CM500002]: Info on incoming INVITE:
INVITE sip:[email protected]:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.1.10.203:5060;branch=z9hG4bKf6dfa4e5B0822CAE
Max-Forwards: 70
Contact: [sip:[email protected]:5060]
To: [sip:[email protected];user=phone]
From: "Home Office"[sip:[email protected]];tag=26BC64DB-808D8FB4
Call-ID: [email protected]
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:[email protected]: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:umThread::thread [CM100002]: *!* Exception detected: Missing header Contact *!*
08:44:33.143 LineMgr:nRegUpdAor Registered line: L:10003(DLINK DVG 1120S)

 

[N_G]

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

 

[SmokeyCarr]

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

 

[N_G]

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

 

[SmokeyCarr]

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

 

[Pentangle]

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.