3cx receiving calls but outgoing calls forbidden

[vichar]

Hello, we are using an unsupported sip trunk from a local provider in india (Tata Communications). They allow communication from only 1 allocated IP provided by them.
We have configured that IP in our router. And connected 3cx machine to the router via LAN cable. Also created one-to-one NATting with the router.

We are able to receive calls. But all outgoing calls are shown as forbidden in 3cx activity log.
We communicated with our sip provider and got the response that our requests to them are not in correct format.

The correct format which they allow is:

INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 10.0.90.182:5060;branch=z9hG4bK5870c68d
Max-Forwards: 70
From: <sip:[email protected]>;tag=as717caedd Here please send the DID number
To: <sip:[email protected]>
Contact: <sip:xx[email protected]:5060> Here please send the DID number
Call-ID: [email protected]:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.11.0(11.20.0)
Date: Mon, 11 Feb 2019 07:59:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
P-Preferred-Identity:<sip:[email protected]> Here please send the SIP Pilot number
Content-Type: application/sdp
Content-Length: 305
v=0
o=root 1127501856 1127501856 IN IP4 10.0.90.182
s=Asterisk PBX 11.20.0
c=IN IP4 10.0.90.182
t=0 0
m=audio 10490 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

The format of our requests is:

Request-Line: INVITE sip:[email protected]:5060;user=phone SIP/2.0
Method: INVITE
Request-URI: sip:[email protected]:5060;user=phone
[Resent Packet: False]
Message Header
Via: SIP/2.0/UDP 192.168.1.204:5060;branch=z9hG4bK20788193252639623004
From: varun singh <sip:[email protected]:5060>;tag=1635516337
To: "9xxxxx4546" <sip:[email protected]:5060;user=phone>
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected]:5060>
Max-Forwards: 70
Supported: replaces, join, path
User-Agent: Fanvil X3S 2.8.0.6251 0c383e1b36f1
P-Early-Media: supported
Allow-Events: talk,hold
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE
MAC: 0c383e1b36f1
[Expert Info (Note/Undecoded): Unrecognised SIP header (mac)]
Content-Type: application/sdp
Content-Length: 384


We tried to set the outbound parameters in 3cx SIP trunk settings, but we found no change even after changing the settings.

Kindly advise what are we missing.

 

[JohnS_3CX]

Hello,

Your provider is telling you to change 3 specific fields to match theirs:


From: <sip:your8digitDID@10.0.90.2>;tag=as717caedd Here please send the DID number
Contact: <sip:your8digitDID@10.0.90.182:5060> Here please send the DID number
P-Preferred-Identity:<sip:yourmaintrunknumber@10.0.90.2> Here please send the SIP Pilot number

Currently you are sending something else so they are probably rejecting that calls due to that. Your invite was the one from the Fanvil to the PBX, you should look at the one from the PBX to the provider..


So I think you may need to change your fields in the outbound parameters to show what the provider asked for and then run a capture to see if that is indeed what is going out.

From: User Part = OutboundCallerID
Contact: User Part = OutboundCallerID
P-Preferred Identity : User Part = LineNumber



If you have verified it to be correct and it still fails, show them the capture and ask for further advice.

 

[johnvam]

We have same problem with local Greek provider...

Our provider needs ptime=20 parameter on first invite!

Any idea?

 

[YiannisH_3CX]

We have same problem with local Greek provider...

Our provider needs ptime=20 parameter on first invite!

Any idea?

Please note that you cannot force the PBX to add the ptime value into the Invite message and you should not have to as the field is not mandatory.
I would recommend using a supported provider from our supported providers list
https://www.3cx.com/partners/sip-trunks/

 

[johnvam]

Please note that you cannot force the PBX to add the ptime value into the Invite message and you should not have to as the field is not mandatory.
I would recommend using a supported provider from our supported providers list
https://www.3cx.com/partners/sip-trunks/

I thing that to force me to use the supported providers is not the solution... Each country has specific providers and you must be able to support them all! It's so easy to push ptime=20 on first invite request!!!

 

[YiannisH_3CX]

I thing that to force me to use the supported providers is not the solution... Each country has specific providers and you must be able to support them all! It's so easy to push ptime=20 on first invite request!!!

I am not trying to force you to use a supported provider. It was a suggestion that will save you time and effort because we actually took the time and effort to test them, support them and include them in the PBX. We have a big list of supported providers (some of them are pretty big) that work without the ptime attribute in the SDP as it is not mandatory to be included. We cannot support all available providers as they do not all work the same and not all them follow the same standards. You can of course chose any provider you want and use the generic template to add them to the PBX but then you need to take the time to test them and adjust any required settings.

 

[johnvam]

I am not trying to force you to use a supported provider. It was a suggestion that will save you time and effort because we actually took the time and effort to test them, support them and include them in the PBX. We have a big list of supported providers (some of them are pretty big) that work without the ptime attribute in the SDP as it is not mandatory to be included. We cannot support all available providers as they do not all work the same and not all them follow the same standards. You can of course chose any provider you want and use the generic template to add them to the PBX but then you need to take the time to test them and adjust any required settings.

Dear Yiannis,
I fully understand what you say but I thing that is more easy ust to add the ability to add some attributes in SDP!
Thank you anyway