SIP Trunk Grandstream HT841 FXO no outgoing calls

[Alhakimi]

Hello Everyone,
We are still limited to use PSTN Lines, so I am having Grandstream HT841 FXO , but not able to make outgoing calls, the incoming calls work, but outgoing still not going through , do any has kind of tips for me to make it work.
appreciates your help here.

Rashed

 

[leejor]

It might help if you provided a bit more information...

Is this on a system locally hosted?
Have you looked at the 3CX Activity Log to see if the call is being sent to the correct IP and port, and if there are any error messages?
Did you manually program the 841, or use a template?
Did it ever work for outgoing?

 

[Alhakimi]

It might help if you provided a bit more information...

Is this on a system locally hosted?
Have you looked at the 3CX Activity Log to see if the call is being sent to the correct IP and port, and if there are any error messages?
Did you manually program the 841, or use a template?
Did it ever work for outgoing?

Thank you for your reply, here are more details on the scenario:
I am hosting it locally and i am attaching the configuration screenshot I did in my system, the screenshots here shows that i am using Gateway and not SIP Trunk , but the same issue with both scenarios.
this is new installation, incomming calls are working but outgoing not.
i don't see an option to configure 841 as template.

 

[leejor]

You've blacked out the local IPs, so I can't tell if calls are actually routing to the proper IP. (revealing local IPs are not a security concern).
If you use the Activity Log, in Verbose, you may get some additional information. Unlike an ATA, registration doesn't tell 3CX where to send calls, that is determined by your settings in the trunk parameters, you tell 3CX what IP and what port..

If the call is going to the correct IP and port, and you are getting the error message, then it may be something in the gateway that is telling 3CX that it cannot process the call. Many times that will happen because the phone line to the port being used, is open. Because you are receiving calls, I have to assume that is not the case here.

So, check the Activity Log, if 3CX seems to be sending the call to the correct place, then you may have to look at the settings on the gateway.

 

[fortech]

Hi Rashed,

just a quick footnote to your query thread, in case it might be helpful.

To clarify,
- is this a new setup, which you are building out from scratch / or an old setup where things worked and you are trying to make config changes ?
- did you refer to any 3cx setup hints about how to configure Grandstream FXS devices? For example > https://www.3cx.com/voip-gateways/grandstream-ht-fxs/
- is your 3cx local instance, the handytone is in same lan? or do you have cloud-offsite 3cx? if so are you using an SBC to facilitate the FXS<>3cx communication?

and

You mention in your initial question, "We are still limited to use PSTN Lines" but - It might be this is a limitation you have conceptually more than 'strictly speaking 3cx operationally possible'

for example, if you wanted to move forward debug of the outbound calls,
- get a SIP Trunk DID in your callerID-DID-region from a 3cx-supported SIP trunk vendor. You can purchase such a SIP trunk for a very modest monthly cost.
- link the SIP trunk to your 3cx instance, then configure a basic call flow for this sip trunk (ie, inbound call to DID on this trunk routes to IVR / or a specific handset-phone-EXT in the 3cx system).
- validate that test calls inbound work as expected.
- then do a test outbound call flow test; configure temporary outbound flow to route your outbound calls via this SIP Trunk
- make sure they go OK.
- note that there is the scenario possible, depending on your SIP trunk provider and how they enforce or validate CID - you can optionally use a setup where the inbound and outbound calls in 3cx are *not* routed via same actual service provider.

ie, if you have actual DID bound to a copper landline service, and this routes inbound calls via Carrier -> Copper Line -> Grandstream > 3cx.
You can config your outbound calls to flow via 3cx > SIP trunk > Connect to outside world
and if your SIP trunk carrier is configured to accept (outbound call DID not match actual DID I have provisioned on this carrier, but rather "my landline DID CID") - then the calls you place outbound to world will still have proper CID / callers won't know you are not really calling from the 'correct' analog landline phone service carrier.
Note typically to make this config work, the SIP trunk provider will have normally some method to validate that you are legit / entitled to do this kind of CID_Masquerade, ie, when you make setup on their service you will need to be able to answer an inbound test call to REAL_DID and 'punch in the challenge MFA Code provided by your SIP carrier" to validate "yes, that is me, I control both the legacy DID phone# as well as the new sip trunk service, and I want to use one to impersonate the other, and this is legit, not a scammer trying to impersonate phone number of someone else in a non-valid manner".

different sip trunk carriers will do this in different methods (more or less, if they support it at all)
as a simple illustrative reference because I know this works - with "Voip MS" Sip trunk carrier, they have docs here > https://wiki.voip.ms/article/Caller_ID -> which give hints on "how to setup a verified-CID-number / if you wish to use that CID on outbound calls and this DID# is not a number hosted with the VoipMS-Service you are using". (This is just one illustrative example to provide context.)

hope this is possibly helpful

good luck with the debug work

Tim