SIP trunks offline ( Asterisk server A -> Satellite > Asterisk server B )

[GrandPoobah1 (Grand Poobah)]

Hi, I’m having problems bringing Asterisk SIP trunks online ( Asterisk A -> Satellite -> Asterisk B ) over a satellite link using password authentication. I’m fairly certain timers are expiring & preventing the trunks from registering online, ( i don’t have NAT enabled at the momemt & my FW rules are valid, leaving me with 3 questions

1. What would be the best and simplest configuration to use for the outbound /inbound trunk group settings ( I’m using the default parameters provided by Free pbx )

2. Is there a Jitter buffer setting / feature available with Free PBX

3. How can I troubleshoot / capture packets of the issue on a VMware host?

Any assistance would be greatly appreciated.

Regards,

GPB1

[amardeeprana]

If traffic is going beyond firewalls then you need to open SIP port on both side for your Servers Only. and allow both IP in Fail2ban if you are using it.

[cynjut (Dave Burgess)]

@GrandPoobah1

What would be the best and simplest configuration to use for the outbound /inbound trunk group settings ( I’m using the default parameters provided by Free pbx )

• Get static IP addresses at both ends and use IP Authentication. This simplifies your handshaking.
• Make sure your NAT rules are correct and wireshark and sngrep the crap out of everything. You may be running into NAT (because of the nature of Network Address Translation) not knowing how to get your traffic from point A through Satellite X to Point B.

@GrandPoobah1

Is there a Jitter buffer setting / feature available with Free PBX

Yes, but let’s save that until literally the last. It’s only useful for when calls are working, and it doesn’t sound like you’re there yet.

@GrandPoobah1

How can I troubleshoot / capture packets of the issue on a VMware host?

So, in addition to a technically challenging configuration when using in-house hardware, you are adding in the variables of virtualization and network routing. I mean. why make it challenging when you can make it Really HARD. I like your spirit and determination to add those extra variables.

There are lots of tools you can use, including the previously mentioned wireshark, sngrep, and others. You can also glean a lot of information from the system logs. Lorne’s guide to “helping us help you” is at
https://wiki.freepbx.org/display/SUP/Providing+Great+Debug
It really is a good place to start looking for more information.

[sorvani (Jared Busch)]

Also satellite server has horrible ping times to get to geosynchronous orbit. You will never have good phone service on current satellite based internet solutions.