Solved Unable to transfer external VOIP to external number via SIP Trunk

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McDowellster

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The system just disconnects the call.

When using an internal number calling ext 702 (which forwards to an external number) it connects, no problem. When a call comes in on the SIP Trunk, forwarding to 702 (I need this for our IVR / Digital Receptionist) the call fails. It appears to be related as to how 3cx handles the call to the SIP Trunk.

02/16/2017 4:07:50 PM - L:129.6[Line:10000>>1866xxxxxxx] got Failure: Failure Recv 500/INVITE from 208.43.27.75:5060 tid=781853790f4cfa29 Call-ID=x-bbNT07vdcp0qYC7cEYmg..: SIP/2.0 500 Body-less INVITE is not supported

As per the error above it appears that the transferred call is missing the body. Any idea's?
 
Have you looked at the failed message with Wireshark, and compared to a working call. Could this be a re-invite issue, or a setting on the outgoing trunk parameters that require tweaking? Is your provider of any help?

Does this happen with all calls, or only those involving toll free numbers?
 
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I spoke to the VOIP provider and they say it's 3CX at fault. Sending a SIP invite with an empty body. Not sure where I can configure this any differently with transfers...
 
who is the provider?
 
IXICA Communications Inc.
 
Ask this unsupported provider "Do you support re-invites" like all the sip trunks in the universe do?
Also you are from Canada. Out of all sip trunks we have in canada why do you go for an unsupported (therefore an untested) provider?

It's not like you come from a country where we have no supported providers. Then you would be right because you would have no choice. But I just checked now and we support and interop with 5 Canadian Sip Trunk services. Apart from the fact that most of the us providers we support also offer service in canada too.

The provider should handle this. 3CX was blamed because whoever you spoke to has large gaps in knowledge when it comes to sip signalling.

And in case you receive a blank face from the provider after asking him the re-invites question, you can proceed with a variation of the question: "How do you handle re-invites without SDP which are perfectly normal and required in transfer scenarios?"

The Provider you chose cannot make transfers basically. And they point the finger at 3CX. This is why we constantly preach to USE a supported Voip provider. If these guys come to make the interop test for example they will not pass.
This is why we have supported and not supported providers.

Do the following - get a free callcentric account and try the same thing. Mentioned callcentric because it is in the USA and if IM not mistaken offers service in Canada.
 
And in case you receive a blank face from the provider after asking him the re-invites question, you can proceed with a variation of the question: "How do you handle re-invites without SDP which are perfectly normal and required in transfer scenarios?"
Or this provider should upgrade their high-performance, carrier grade B2BUA?:confused:
  • RFC3261 compliance
  • Seamless compatibility with majority of popular SIP software and hardware on the market today
Add support for the case when no SDP is present in the INVITE
 
IXICA keeps redirecting my questions to our 3CX vendor. They recommended IXICA so they must have dealt with this scenario in the past.
 
Well believe it or not... support re-invite needs to be turned off. This work fine now. Funny because IXICA wanted me to turn it on in the first place.
 
Glad to hear the issue is resolved and thank you for sharing your solution
 
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