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Endpoint::findSource Can not find source by following specs

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onyunn

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I have installed system, and am able to make calls to other extensions, as well as make outbound calls, however when trying to make an inbound call I get the following message...please help.

Endpoint::findSource Can not find source by following specs: From: '5852373013'; To: '15852195905'; Rline: '5852195905'; Contact: '5852373013'
 
Re: Endpoint::findSource Can not find source by following sp

onyunn said:
I have installed system, and am able to make calls to other extensions, as well as make outbound calls, however when trying to make an inbound call I get the following message...please help.

Endpoint::findSource Can not find source by following specs: From: '5852373013'; To: '15852195905'; Rline: '5852195905'; Contact: '5852373013'
Click to expand...

Hi,

You should set up your line to have correct numbers. At least one number (External, Internal or Authentication ID) should match one provided in the string above. I can not guess what that numbers means in your case, but one of them (most probably - From) is your account number (auth-id). If there is no auth-id, than there should be external number of your VoIP account. I think it's in To field. If it's true, than you shall put it in 'External Number' field of your line settings, and than in Advanced settings for its VoIP provider you correctly set up section about Line-ID (I don't have fresh version of interface handy, so I can not tell exact text).
In this section you should choose 'External number' ... in ... 'To' field
If it doesn't help - tell me what each of numbers above exactly means, and I will suggest exact settings.
 
Thanks for getting on this for me....

The 5852373013 is a standard PSTN Home number, not connected to the VOIP that I'm using to test inbound connections with.

5852195905 is the inbound trunk number given to me by my VOIP Provider, nexvortex.com. I think he said this is the DID provided for inbound calls.

I had previously tried using Asterisk, and they gave a sample configuration file, not sure if it will help you determine where I went wrong.


[nexvortex.com]
username=sparrowfin
user=phone
type=peer
secret=XXXXXXXXX
nat=yes
insecure=very
host=66.23.129.253
authname=sparrowfin
fromuser=sparrowfin
dtmfmode=rfc2833
dtmf=rfc2833
context=from-trunk
canreinvite=no


If you need any additional information, please let me know.

Thanks in advance!!!!
 
Than you should set External Number for the line to be 5852195905
And, for the provider settings, you should go to Advanced options, section 'Line-ID matching:' select 'External line' and 'Request-Line-URI' field
After that all should work
 
Call terminates, while receiving inbound call, during welcom

I put in the settings that you suppled, and I am able to dial inbound now, however, the call terminates.

I have the system setup to go to the auto attendant, and when I place an inbound call it begins to play the welcome message I recorded, but it terminates the call within a few seconds.

Not sure if it is related to NAT of not. I have configured STUN, and see in the log that it resolves the correct external IP address.

I have 5060 UDP and TCP port forwarded from the Router to the Internal IP of the PBX. I also have a range of UDP RTP ports open and forwarded to the same Internal PBX.

I have no idea where to look next....any insight would be greatly appreciated.

Thanks in advance for your help
 
Hi,

Restart your server (this will clear the logs) and reattempt the call. Then create a support file and send it to our support mailbox.

This way we can check the detailed logs and see who is terminting the call.

Posting the generated log lines from under 'Server status' will also be a quick help.
 
Server ststus capture

Here is the server status info after a fresh reboot, and an inbound call.

10:58:47.671 StratInOut::onHangUp Call from Ln:[email protected] to 100 has been terminated
10:58:26.500 CallLegImpl::onConnected Established media channel for Ln:[email protected]: remote=207.218.174.182:57884; local=192.168.1.6:7000
10:58:26.203 CallConf::onIncoming Incoming call from Ln:[email protected] to sip:[email protected]:5060
10:57:17.625 ??: SL: connected pdc-sparrow:0/PHPExtension_0 at [pdc-sparrow:0]/PHPExtension_0
10:50:29.750 ClientRegs::onSuccess Registration of sip:[email protected] is successful
10:50:29.593 IVRConnected: IVR Server is connected
10:50:29.593 ??: SL: connected pdc-sparrow:5483/IVRServer at [pdc-sparrow:5483]/IVRServer
10:50:29.375 ExtLine::Register_ Send registration for "Sparrow Financial"<sip:[email protected]>
10:50:29.375 ExtLine::Register_ Use External IP for device line registration DN='10000' device='nexvortex.com'
10:50:26.875 StunClient::process STUN resolved external IP=66.67.163.22:5060 by server 80.239.235.209
10:50:26.796 MediaServerConnected: Media Server is connected
10:50:26.796 ??: SL: connected pdc-sparrow:0/MediaServer at [pdc-sparrow:0]/MediaServer
10:50:26.750 StunClient::process Send initial STUN request to 80.239.235.209
10:50:26.671 ??: SL: connected pdc-sparrow:0/VoiceBoxManagerService at [pdc-sparrow:0]/VoiceBoxManagerService
10:50:26.640 CallMgr::Stack::thread ** Enter Stack Loop **
10:50:26.484 DBA: ** Database connection Ok **
10:50:26.359 CallMgr::DumThread::thread ** Enter DUM Thread **
10:50:26.281 CallMgr::Stack::Initialize ** Adding transports **
10:50:26.171 CallMgr::Initialize ** Initializing SIP stack **
10:50:26.171 CallMgr::Initialize Version: 3.0.1699.0
10:50:26.125 CallMgr::Initialize Start 3CX PhoneSystem Call Manager
 
I have an almost identical problem!

I have an almost identical problem!

Can not find source by following specs: From: '8447'; To: '101'; Rline: '101'; Contact: '8447'

Its between 2 systems, the 3CX and another. When I call an extension from the 3CX to the other system (from x101 to x28161) it all works fine.
When I call from x28161 to the x101 on the 3CX, I get the above error.

You can see that the To field contains 101 (the extension I am trying to dial) and the From contains the ID of the system I am calling from. I have put 8447 into the External Number (8447) on the Line Config, and am using an Internal dummy number of 7000. My extension number is 101.

The call hits the 3CX, meaning that the problem is in routing on the 3CX, but all I get as an error is:

Can not find source by following specs: From: '8447'; To: '101'; Rline: '101'; Contact: '8447'

and an unobtainable dial-tone.

Help!
 
10:58:47.671 StratInOut::onHangUp Call from Ln:[email protected] to 100 has been terminated
Click to expand...
Most likely a SIP port re-registration. So the NAT, it the call automatically terminates during the call it can be sip reregistration.

ihq-it
Now that is confusing, external being internal and visa versa?
When you configured the external extension did you tell it to be external?
If so it will use ports 9000 - 9003 instead of internal ports. So it can drop the call due to firewalls.

Do you have any (fire walls that is?)
 
port 9000?

When I create the VOIP Gateway, the default is to use port 5060 - so thats what we use.

Gateway Port (default is 5060)

I tried changing the other end to attempt an outbound call to port 9000, and the 3CX doesn't see the call at all. At least when its using 5060 it see's it. I have no firewall issue as the two boxes are sat next to each other on the same subnet.

any ideas?
 
outbound call to port 9000
Click to expand...

Oops sorry for the confusion, 9000 - 9003 are ports used by 3cx you should not have to configure them.

Default is good.

I have to have a play with this problem, not sure what the answer is.

Henk.
 
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