SIP Trunk with STC (KSA)

[siscom]

Good Afternoon Colleagues
I hope you are doing well . May i know if anyone has experience setting up SIP Trunk with STC (One of provider at KSA) ?

We ae trying to setup it but failed.

Thanks in advance.

Best Regards

 

[Alejandro_3CX]

Hi, I would suggest to enter in contact with your voip provider for assistance.

You may also check this document regarding the trunk registration that can help you learn more the process.

Have a great day

 

[siscom]

Thx for your replay

Well Noted

 

[Alejandro_3CX]

You are welcome!

 

[siscom]

Good Morning Alejandro
We have contacted STC and kindly find my below notes:

• They informed us that the trunk should be peer trunk.
• Attached is Email from STC with the required setup information.
• Attached also is 3CX SIP configuration based on email from STC.

The call didn't established . Below is Activity log:



10/24/2023 9:28:53 AM - [CM503025]: Call(C:2): Calling T:Line:10000>>0122166666@[Dev:sip:[email protected]:5060] for L:2.1[Extn:101]
10/24/2023 9:28:53 AM - Route to L:2.2[Line:10000>>0122166666] sends Invite-OUT Send Req INVITE from 10.154.15.1:0 tid=6427e829ec1ab462 Call-ID=KH7iRjPsjM6a8biJ5Lb6GQ..:
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/ ;branch=z9hG4bK-524287-1---6427e829ec1ab462;rport
Max-Forwards: 70
Contact: <sip:[email protected]:5060>
To: <sip:[email protected]:5060>
From: "8068800"<sip:[email protected]:5060>;tag=2210927b
Call-ID: KH7iRjPsjM6a8biJ5Lb6GQ..
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE, UPDATE
Content-Type: application/sdp
Supported: replaces, timer
User-Agent: 3CXPhoneSystem 18.0.7.312 (312)
Remote-Party-ID: "8068800"<sip:[email protected]:5060>;party=calling
Content-Length: 207

v=0
o=3cxPS 353328168960 255366004737 IN IP4 10.170.5.182
s=3cxPS Audio call
c=IN IP4 10.170.5.182
t=0 0
m=audio 7002 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
10/24/2023 9:28:53 AM - [Flow] Call(C:2): making call from L:2.1[Extn:101] to T:Line:10000>>0122166666@[Dev:sip:[email protected]:5060]
10/24/2023 9:28:53 AM - [CM503027]: Call(C:2): From: Extn:101 ("Agent 1" <sip:[email protected]:0>) to T:Line:10000>>0122166666@[Dev:sip:[email protected]:5060]
10/24/2023 9:28:53 AM - [CM503004]: Call(C:2): Route 1: from L:2.1[Extn:101] to T:Line:10000>>0122166666@[Dev:sip:[email protected]:5060]
10/24/2023 9:28:53 AM - Line limit check: Current # of calls for line Lc:10000(@stc SIP Trunk[<sip:[email protected]:5060/UDP>]) is 0; limit is 10


So kindly advice.

Best Regards

 

[TheodorosG_3CX]

I believe the guide below will give you all the information you need:
• https://www.3cx.com/blog/docs/voip-provider-template/

 

[siscom]

Good Afternoon Theodoros
Thanks for the great guide . In my case , The other end is IP Based / SIP Trunk VoIP providers .

I have followed the guide but i have the below notes:

• For "Source Identification" , Does it required to be set in spite of the fact that the other end is IP Based / SIP Trunk VoIP providers or i can ignore it.
• Also for "Inbound/Outbound Parameter" , Is it required for IP Based / SIP Trunk VoIP providers ? If so , Should i follow the attached which is listed in the guide ?

Thx in advance.

Best Regards

 

[TheodorosG_3CX]

Maybe you need to check those options as well:

 

[siscom]

Good Morning Theodoros
Thx for your replay . I am little bit confused now .

May i know your feedback regarding the below two points ?

Good Afternoon Theodoros
Thanks for the great guide . In my case , The other end is IP Based / SIP Trunk VoIP providers .

I have followed the guide but i have the below notes:

• For "Source Identification" , Does it required to be set in spite of the fact that the other end is IP Based / SIP Trunk VoIP providers or i can ignore it.
• Also for "Inbound/Outbound Parameter" , Is it required for IP Based / SIP Trunk VoIP providers ? If so , Should i follow the attached which is listed in the guide ?

Thx in advance.

Best Regards

- Also for the below , for the field " Put Public IP in SIP VIA Header" , As shared in my previous email , Our customer received email from the VOIP provider with the IP Address that should be configured on 3CX machine which is "10.170.5.182" . Do you mean to put this IP Address "10.170.5.182" in field " Put Public IP in SIP VIA Header" ?

- Also for the field "Select which IP to use in 'Contact' (SIP) and 'Connection'(SDP) fields" , DO you mean to select the option "Use this IP Address" and add the IP received from VOIP provider which is "10.170.5.182" ?

Maybe you need to check those options as well:
View attachment 37968

Thx in advance

 

[TheodorosG_3CX]

If the provider accepts the information in "Put Public IP in SIP VIA Header" then you can use that option.

As for the Inbound/Outbound parameters they are not related with the registration.

 

[siscom]

Hello
Well Noted .

• For "Source Identification" , Does it required to be set in spite of the fact that the other end is IP Based / SIP Trunk VoIP providers or i can ignore i?

Best Regards

 

[TheodorosG_3CX]

Also check guideline here for more information.

 

[siscom]

Thx for the great document . What i understand that source identification is used to check the source of INVITE message to be sure it is from trusted domain . In spite that fact that our VOIP provider shared with us username & password , He shared with us the below information also:

"Note: No need to send registration message with username & password just respond on STC option with 200 ok message accordingly in order to get the trunk registered."

So does source identification is required in my case ?

Best Regards

 

[TheodorosG_3CX]

Correct, in your case it is not requested.

 

[siscom]

Great . Another question , Does 3CX use peer trunk "This is one the requirements from the SIP provider" ?

 

[TheodorosG_3CX]

Can you please elaborate further on this?

 

[siscom]

Good Afternoon Theodoros
I have doubled checked with them and they mean by peer trunk that no username/password is required for authentication . So this for sure supported by 3CX when choosing the option "Do not require - IP Based".

I will test the template with the customer and will update you.

Thx in advance.

Best Regards

 

[siscom]

Good Morning Theodoros
I hope you are doing well . I have implement the template based on the link you shared with me and based on our discussion in this email and i would like to share with you the below:

• Attached is the template after finalize it and upload it to 3CX.
• Inbound calls successfully established .
• Outbound called still not successful.
• Attached is a log for outbound call that failed.

SO kindly advice.

Best Regards

 

[TheodorosG_3CX]

The outbound call was routed correctly. If it is failing this has to be clarified with the provider.

 

[siscom]

Hello Theodoros
Thx for your fast response . We will check with the provider and will let you know if any further support required.

Best Regards