SIP to PA System gateway just stopped audio.

Obviously, this will be the result of a specific call, and RTP ports are randomly negotiated, but it will tell you what that call tried to do. So you can then look at your NAT and Firewall settings and see if something is weird.

JaredBusch

@JasGot said in SIP to PA System gateway just stopped audio.:

@JaredBusch said in SIP to PA System gateway just stopped audio.:

If it is registered, pop sngrep on the command line and make a phone call to it.

This?

Yes. You obviously, cropped the IP info, but using that, look and make sure both side have the right things allowed. Assuming the PBX is hosted, then the side with your firewall is the one to validate.

JasGot

I contacted the support staff at the host. I have to say, it never ceases to amaze me how good they are with this stuff.

They changed the NAT setting for the extension. I had no idea the setting was available because I have never setup a SIP device in VitalPBX before, only PJSIP. The NAT setting in question is not there for PJSIP.

This is the new setting on the Extension page for this extension:

I still don't know why it worked fine for a month and then suddenly stopped. I'll look in to that.

JaredBusch

@JasGot said in SIP to PA System gateway just stopped audio.:

I have never setup a SIP device in VitalPBX before

Don't do this. In Asterisk land (what VitalPBX uses) SIP (meaning anything using the chan_sip driver) is dead.

chan_pjsip is still SIP, just a different driver on the server. It has zero to do with anything not on the server.

If the device or provider accept the SIP protocol (aka everything does if it is not Cisco), then server side, you can use chan_pjsip.

JaredBusch

@JaredBusch said in SIP to PA System gateway just stopped audio.:

SIP (meaning anything using the chan_sip driver) is dead.

For reference: https://community.freepbx.org/t/asterisk-v20-rc1-is-released/85613