3cx only tcp voip provider

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ImpLosioN

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Hello,

A VoIP provider can be successful only as TCP calls.
How can I make a call just as TCP?

Thanks.
 
Hi there,
You can try adding ;transport=Tcp after the host of provider, then apply (this is not a supported configuration).
 
Where add this?

I add IP;transport failure?

01-Eki-2014 15:53:39.634 Leg L:117955.1[Extn] is terminated: Cause: BYE from 192.168.19.31:50441
01-Eki-2014 15:53:35.425 [CM503007]: Call(C:117955): Extn:192 has joined, contact <sip:[email protected]:50441>
01-Eki-2014 15:53:35.422 L:117955.3[EndCall] has joined to L:117955.1[Extn]
01-Eki-2014 15:53:35.313 [CM503025]: Call(C:117955): Calling T:EndCall:EndCall@[Dev:sip:[email protected]:40600;rinstance=90e4cb9371e23587] for L:117955.1[Extn]
01-Eki-2014 15:53:35.305 L:117955.1[Extn] failed to reach Line:10005>>05552379265, reason Server Failure
01-Eki-2014 15:53:35.305 Call to T:Line:10005>>05552379265@[Dev:sip:[email protected];transport=Tcp:5060] from L:117955.1[Extn] failed, cause: Cause: 503 Service Unavailable/INVITE from local
01-Eki-2014 15:53:35.298 [CM503025]: Call(C:117955): Calling T:Line:10005>>05552379265@[Dev:sip:[email protected];transport=Tcp:5060] for L:117955.1[Extn]
01-Eki-2014 15:53:35.273 [CM503027]: Call(C:117955): From: Extn:192 ("keremg" <sip:[email protected]:5060>) to T:Line:10005>>05552379265@[Dev:sip:[email protected];transport=Tcp:5060]
01-Eki-2014 15:53:35.273 [CM503004]: Call(C:117955): Route 1: from L:117955.1[Extn] to T:Line:10005>>05552379265@[Dev:sip:[email protected];transport=Tcp:5060]
 
Yes apologizes this trick seems to not work anymore, the correct syntax from hostpart should be 84.44.127.6:5060;transport=Tcp but you can't manipulate the fields in this way apparently it will make it 84.44.127.6;transport=Tcp:5060 which is not valid.

Therefore you will have to post this as a feature request on 3CX ideas : http://www.3cx.com/ideas/ there is no workaround at the moment on this matter.
 
Hi,

I am having same issue.

Is there any solution now about using only TCP with SIP Trunk?

or

Is there any option to set server mode or client mode for Sip Trunks?


Thanks.
 
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