No ringback - last try

[bullan]

OK, I'm evaluating this software and looks none is able to help me with a simple questions.

I have 3 generic providers but have no ringback on the outband calls. I search and found same problem when tunnels are involved;
I AM NOT USING TUNNELS. Just the simple setup.

I'm testing a HT286 and eybeam and on all 3 trunks I have the same problem.

Can someone guide me where to look? Is it a settings problem? NAT, Firewall?

thanks

 

[bluetel2]

Hi Bullan,

post your logs please.

describ your configuration (serveur router tel devices etc..)

 

[mfm]

Hi,

Go to settings > Advanced > and advise what the "Local SIP domain" is. Also besides logs a wireshark capture would be required for a problem like this but let us start with the local sip domain.

 

[bullan]

Hi,

Go to settings > Advanced > and advise what the "Local SIP domain" is. Also besides logs a wireshark capture would be required for a problem like this but let us start with the local sip domain.

local sip domain: 192.168.1.27 (the server IP)


and the log for an external call is:

18:08:39.818  [CM503008]: Call(36): Call is terminated
18:08:35.427  Session 12274 of leg C:36.1 is confirmed
18:08:35.256  [CM503007]: Call(36): Device joined: sip:[email protected]:5060
18:08:35.240  [CM503007]: Call(36): Device joined: sip:[email protected]:40000
18:08:35.224  [CM505003]: Provider:[freephoneline] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Sippy] PBX contact: [sip:[email protected]:5060]
18:08:35.224  [CM503002]: Call(36): Alerting sip:[email protected]:5060
18:08:23.646  [CM503025]: Call(36): Calling VoIPline:514yyyyyyyyy@(Ln.10002@freephoneline)@[Dev:sip:[email protected]:5060]
18:08:23.615  [CM503004]: Call(36): Route 2: VoIPline:514yyyyyyyyy@(Ln.10004@videotron)@[Dev:sip:[email protected]:5060]
18:08:23.615  [CM503004]: Call(36): Route 1: VoIPline:514yyyyyyyyy@(Ln.10002@freephoneline)@[Dev:sip:[email protected]:5060]
18:08:23.568  [CM503010]: Making route(s) to "514yyyyyyyyy"<sip:[email protected]>
18:08:23.568  [CM505001]: Ext.102: Device info: Device Identified: [Man: Counterpath;Mod: eyeBeam;Rev: General] Capabilities:[reinvite, no-replaces, unable-no-sdp, recvonly] UserAgent: [eyeBeam release 1102q stamp 51814] PBX contact: [sip:[email protected]:5060]
18:08:23.552  [CM503001]: Call(36): Incoming call from Ext.102 to "514yyyyyyyyy"<sip:[email protected]>
18:08:23.537  [CM500002]: Info on incoming INVITE:
  INVITE sip:[email protected] SIP/2.0
  Via: SIP/2.0/UDP 192.168.1.20:40000;branch=z9hG4bK-d8754z-0629855b6f329314-1---d8754z-;rport=40000
  Max-Forwards: 70
  Contact: <sip:[email protected]:40000>
  To: "514yyyyyyyyy"<sip:[email protected]>
  From: "3cx"<sip:[email protected]>;tag=b938ec73
  Call-ID: YzdlNzE2YTU4MmU5ZmRjNjRmY2YxMDNiY2ExZThmMDM.
  CSeq: 2 INVITE
  Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
  Proxy-Authorization: Digest username="102",realm="3CXPhoneSystem",nonce="414d535c01e942d781:d52af0b05bd1576eead08ffb1476a053",uri="sip:[email protected]",response="3325bfb0639b7e73ce8008e62ee0210c",algorithm=MD5
  User-Agent: eyeBeam release 1102q stamp 51814
  Content-Length: 0

 

[bullan]

so? no one? I'm the only one with this problem?

 

[Cjay]

You might want to search on 'early media' - might possibly have something to do with this...?

 

[mfm]

Hi,

Try with the 3CX softphone out of tunnel mode and check the results with that. I would need a wireshark to diagnose this further if you have a support contract then open a ticket and I will guide you from there.

 

[bullan]

with 3cx softphone all is looking right; I can hear the ringback. that is why I asked if can be a setup problem.
NAT or firewall or something else.
I dont have a contract and I dont intend to buy one for simple things like this; If none can help me I'll just stop testing and looking for something else.

thanks

 

[smb1]

3cX support asked if you could perform a Wireshark capture and post those to help.

Without a support contract you can only expect help on a best effort basis.

Another suggestion would be to try and stick with 3cX supported phones and VOIP providers if possible when learning the system and then branch out when you have some experience with 3cX under your belt.

 

[bullan]

I'm testing that for selling purposes, and I have to work with voip providers used in my region, as is Montreal. I have a telecom company and had clients who ask for solutions to replace their old nortel. I sell and install talkswitch but having alternatives are always a good ideea.

There are little chances for my client to cancel his voip providers videotron and to switch to one from US, Belgium, UK or else.

And the adaptor is in supported list as Handytone 286.

Anyway my problem is not from providers cause 3cx softphone works very well, the ringback is not audible for HT286.
And for the paid support, i'm very OK with that for some fancy features who require difficult settings. But, let's be serious, ring back is not a feature is a basic must. Make part of the first step of settings. If I cannot pas by, I'll never arrive at the fancy features. And it looks like.

 

[mfm]

Hi Bullan,

There is a number of things to consider here before assuming that we are not giving out the basic ringing tone. When using a phone over a network there a number of factors to consider such as : Your network on the phone side, your router on the phone side, the network inbetween then two, and finally your router on the PBX side.

The next step I would take from here, is to grab hold of the 3cx tunnel proxy manager and try using that:

http://www.3cx.com/downloads/3CXSIPProxyManager8.msi

 

[bullan]

All phones and softphones extensions are in the same LAN. Do I need that SIP proxy manager ????
I tried today with an Linksys Sipura 2100 adapter and I all it's the same. No ringback on outbound calls. So I eliminate the adapters.
Must be something on the VOIP providers or some ports settings. 3cxsoftphone works perfectly.
I saw in the working voip provider list that freephone line is working with standard template. Not my case.

edit: just saw in log that stun request are timed out. Could that be the problem?

21:32:35.265  [CM506004]: STUN request to STUN server 193.16.148.245:3478 has timed out; used Transport: 192.168.1.27:5060
21:32:32.249  [CM506004]: STUN request to STUN server 193.16.148.245:3478 has timed out; used Transport: 192.168.1.27:5060
21:32:29.202  [CM506004]: STUN request to STUN server 193.16.148.245:3478 has timed out; used Transport: 192.168.1.27:5060
21:32:26.140  [CM506004]: STUN request to STUN server 193.16.148.245:3478 has timed out; used Transport: 192.168.1.27:5060
21:32:24.405  Active calls counted toward license limit: []
21:32:23.077  [CM506001]: STUN request to resolve SIP external IP:port mapping is sent to STUN server 193.16.148.245:3478 over Transport 192.168.1.27:5060

the calls are working just fine, I receive and make calls without problems

 

[Destonomos]

I'm having a similar issue. Phone calls can be made to a hardphone that I have through the 3cx box and vice versa but when a call is made outbound there is no ringing tone, just dead air and after about 5 to 7 seconds the phone that you are calling rings and once it is picked up audio is transfered flawlessly.

 

[bullan]

so I'm not the only one.

what voip providers are you using?

 

[Destonomos]

I was talking to another technician and he said that it was in reference to STUN not correctly sending RTP traffic. He said it was an issue with the phone itself, not 3cx. What model phone are you using?

 

[bullan]

I use HT286 and Sipura 2100 as adapters, and Eyebeam as softphone.

I don't think is phone/adapter related cause only happens on freephoneline.ca
I have another test account with netmaster and works just fine.

If its stun related, can we try another stun server? Maybe a more reliable one?

 

[Destonomos]

well for me I was having problems just getting my phone to connect to the server because it was doing some remotely. The 3cx box was behind a router that was wanting to translate the source port. I would say 5060 on the phone as source and to and the router would give it 4180. I changed it to 4180 and it went to 60732 and I didn't know how the router was configured so I just started choosing random ports and for some reason 4198 worked. After I got it to connect the dial tone issue popped up but I've kind of put the issue on the back burner because for me the tech that told me configured the router and he said it was stun.

 

[leejor]

I was talking to another technician and he said that it was in reference to STUN not correctly sending
RTP traffic.

You may want to read up on how STUN works and exactly what it does.

http://www.the-voip-systems.com/stun-protocol-how-it-works.htm

If its stun related, can we try another stun server? Maybe a more reliable one?

Yes,there are many STUN servers available (do a Net search, you will come up with quite a few) and you can use any one that you want. However,iIf they do their job properly, then they should all come up with the same results and report that info to the set.