I would add that when SIP ALG is on with some routers it breaks things too … point being to try it both ways ☺
From: Af <[email protected]> on behalf of George Skorup <[email protected]> Reply-To: <[email protected]> Date: Sunday, April 8, 2018 at 5:29 PM To: <[email protected]> Subject: Re: [AFMUG] OT SIP issue What kind of router/firewall are you working with? No audio is usually a SIP ALG thing. You need the ALG on to rewrite the SIP headers when behind NAT. On 4/8/2018 2:30 PM, Chuck McCown wrote: Tried both ways, no joy. Sent from my iPhone On Apr 8, 2018, at 1:23 PM, Forrest Christian (List Account) <[email protected]> wrote: Are they behind nat? Sounds like it might be a reinvite issue, asterisk will try to get out of the audio path by telling the endpoints to talk directly to each other. If nat is involved asterisk will often tell the endpoints to talk directly even if they have no direct connection between them. Disabling reinvite may help if this is the case. On Sun, Apr 8, 2018, 11:27 AM Chuck McCown <[email protected]> wrote: Pulling our hair out. The Aastra phones will call each other, we can send dtmf but no audio. Linksys sip ata does the same thing. These were working, very frustrating.
