Help: How to configure astpp as gateway to PBX

[Dayo]

I have a PBX which runs FS, precisely fusionpbx.

I would like directions on the best/preferred approach to connect the PBX to astpp.

1. I would like to send calls coming in on DIDs from ASTPP to the PBX.
   
2. I would also like to make calls from the PBX via ASTPP.

So far, what I have done are:

1. Create new SIP device with internal profile for PBX to call through. It works, but I'm not certain if this is the right way to do it
2. I have created DID. Pointed DID to SIP device account (as Local), but this does not work as expected.

Basically, I would like to have ASTPP work like a VoIP Gateway.

Thanks in advance

[Samir Doshi]

Hi Dayo, 

There are various ways to handle that. You may need to explore the best fit way for your case. 

For any commercial help on the same, you can always contact on sa...@inextrix.com. 

Best Regards
--
Samir Doshi
iNextrix Technologies Pvt. Ltd.
http://www.inextrix.com

Disclaimer:
The information contained in this communication is confidential and may be legally privileged. It is intended solely for the use of the individual or entity to whom it is addressed and others authorised to receive it. If you are not the intended recipient you are hereby notified that any disclosure, copying, distribution or taking action in reliance of the contents of this information is strictly prohibited and may be unlawful.

--
You received this message because you are subscribed to the Google Groups "ASTPP" group.
To unsubscribe from this group and stop receiving emails from it, send an email to astpp+un...@googlegroups.com.
To post to this group, send email to as...@googlegroups.com.
Visit this group at http://groups.google.com/group/astpp.
To view this discussion on the web visit https://groups.google.com/d/msgid/astpp/9e7a5714-097e-4212-89f0-7ad78aeca3a3%40googlegroups.com.
For more options, visit https://groups.google.com/d/optout.

[Dayo]

Hello Samir,

Been away for exams.

My case is simple (plus what I have done):

1. My ASTPP box is already connected to the VoIP provider OK

2. I have established connection between the PBX and ASTPP.

3. I have configured ASTPP as the GW on the PBX and calls are sent out OK

4. Inbound calls however have not worked. I have routed DID via SIP Devices and created a GW (with PBX settings) on ASTPP but inbound doesnt work.

I should mention I have connected the PBX directly to the provider and all works well. DIDs are recieved and dialplan picks it up for routing on the PBX.

This is not yet a commercial setup. Just tests envs running in VMs.

[Samir Doshi]

Please provide fs logs. 

Best Regards
--
Samir Doshi
iNextrix Technologies Pvt. Ltd.
http://www.inextrix.com

Disclaimer:
The information contained in this communication is confidential and may be legally privileged. It is intended solely for the use of the individual or entity to whom it is addressed and others authorised to receive it. If you are not the intended recipient you are hereby notified that any disclosure, copying, distribution or taking action in reliance of the contents of this information is strictly prohibited and may be unlawful.

To view this discussion on the web visit https://groups.google.com/d/msgid/astpp/9e506b3c-8cbe-44b4-95b6-01e5be8f256a%40googlegroups.com.

[Caesar]

Hello Dayo,

Maybe this helps you. I setup the DID under DIDs section. In the DID section, I selected "Other"  for the call type and for the destination I put: sofia/default/DIDGOESHERES@IPOFDESTINATIONPBX

DIDGOESHERE = DID or DNIS of DID you are trying to route to the destination PBX.

IPOFDESTINATIONPBX = IP address of destination PBX.

You will obviously need to setup a trunk on the destination PBX to allow incoming calls from the ASTPP server. I successfully did this using Asterisk as the destination pbx.

[Dayo]

Thanks Ceaser.

I was able to resolve the issue some time back. 

This was what worked. sofia/external/DIDNUMBER@DESTINATIONSERVERIP:5080^DIDNUMBER@DESTINATIONSERVERIP

I had to create an external profile, sitting on port 5080.

I have not tried to see if all works if I remove everything after the ^, but this works for me now.

Dayo

[David]

I do have similar issue, I can not get the inbound DID routed to the sip extension.

Any solution?