Vega50 FXO and FreePBX

Ok so I am working on a few things here.

First I have a site that does not have a SIP provider but has local calling via 3 analog lines.
I have a FreePBX testing server setup.

FreePBX 13.0.190.19
Asterisk 11.25.1

I am trying to utilize a Vega50 GW with 4 FXO lines to present those lines into the PBX.

Product version Vega 50
Firmware File VEGAEURO_R088S059

I have the Vega50 setup so phones utilize Outbound Proxy to register to the PBX server via ENP. I can register an extension using XLite and another extension on a Yealink T46G.

PRODTestingPBX*CLI> sip show peers
Name/username Host Dyn Forcerport Comedia ACL Port Status Description
101/;vsc=52652A49E15582B2 x.x.x.121 D No No A 5060 OK (32 ms)
102/;vsc=52652A49E15582B2 x.x.x.121 D No No A 5060 OK (44 ms)

x.x.x.121 is the Vega50

Vega50-out/vega x.x.x121 Yes Yes 5062 OK (5 ms)
vega/vega x.x.x.121 D Yes Yes 5060 OK (39 ms)
22 sip peers [Monitored: 14 online, 8 offline Unmonitored: 0 online, 0 offline]

The above is sip show peers showing the trunk is registered

Here are my sip Settings Outgoing fro the Vega50-out
host=x.x.x.121
username=vega*********
secret=*********
type=peer
qualify=yes
allow=ulaw
port=5062

Here are my sip Settings Incoming from the vega
host=dynamic
username=vega*********
secret=*********
type=peer
qualify=yes
fromdomain=x.x.x.121
dtmfmode=rfc2833
allow=ulaw

Ok so it shows me that the Vega50 is up and since I see my extensions are registered. All of that is good.

Here is my SIP ACCOUNT settings in XLite
Account Name:102
User ID:102
Domain: x.x.x.187
Password:***********
Display name:102
Authorization name:102
|x| Registered with domain and receive calls

|.| Proxy Address: x.x.x.121

Now if i dial a local number on the XLite extension 102.

https://pastebin.com/qWEirqtt

See the above pastebin. This is me dialing on XLite to a local number.

The call goes out from the PBX to the Vega50 and calls me just fine. Now I am having a hard time routing inbound calls from the FXO ports send that to an inbound route to the PBX. I have my inbound route created with a DID of ########03 which is the DID that is connected the the FXO port.

When i call that DID now the phone rings and rings. I do a core show channels and i do not see the call hitting the PBX.


Above is a wireshark while the call is going i see traffic from the Vega50 which i’m going to say is the phones registering.

I am looking for any corrections in what i am doing wrong. I looked at this wiki and nothing there shows me what to do.
http://wiki.freepbx.org/pages/viewpage.action?pageId=60522572

I also looked here and yes this is how i setup the trunk. But does not show me inbound route setup.
http://wiki.freepbx.org/display/VG/Registering+Vega+with+FreePBX

Thank you.

Not at all familiar with the Vega but maybe a codec mismatch issue?

I thought that was my issue since it’s coming from the softphone in this case XLite.

So i am starting down that road. Something about my XLite being on a NAT.

But i am worried that only one response all weekend.

Did you submit a ticket? We’re almost all users here.

Hi!

ie It’s unrelated…

There are apparently SIP keep-alives (apparently proprietary to Counterpath) from XLite/Bria… Disable them there…

Good luck and have a nice day!

Nick

Ok so i found my issue with the softphone. I’ll reply in the morning with the settings i changed.

But i’m still stuck at the calls coming from the FXO port dialing inbound and up to the FreePBX server.

I am not seeing calls coming in over the ISTP connection.

Sangoma is not answering my ticket i submitted last week.