I setup my VoiP.ms trunk according to Chris Sherwood’s guide on VoiP.ms’ wiki. As a new user I can’t post a link.
I am running FreePBX as a VM, my router is pfSense. If I disable the trunk and enable it again I can call out and make inbound calls just find. After 15-30 minutes calls will not pass. Occasionally when I log into VoiP.ms my sub account will say “no registration found” but it almost always says “registered.”
I don’t have any ports forwarded in the router. Are there any settings I can change that will keep the connection open on the router, if that is the issue?
Create a NAT rules and create an alias in Pfsense for the POP server found in your account. Then point to that alias in source of the rule.
In pjsip Settings → Advanced for the trunk, try setting Expiration to something short such as 120.
If you already had that or it doesn’t help, when the problem occurs, does Asterisk show the trunk as registered and reachable? If not, with pjsip logger on, paste the Asterisk log for a failed registration / qualify at pastebin.freepbx.org and post the link here. If yes, paste the log for an attempted outgoing call.
If you are too new to post links, just post the last eight hex characters of the URL.
Thanks Stewart, I just set the expiration in the trunk to 120. Let me give the system a little while to see if it will timeout.
In the FreePBX GUI when I go to Reports > Asterisk Info, the trunk shows ONLINE.
Further down in Peers > PJSIP
Endpoint: voip-ms_trunk Not in use 0 of inf
Aor: voip-ms_trunk 0
Contact: voip-ms_trunk/sip:######******@atlanta2. 2b207c5a69 Avail 23.183
Transport: 0.0.0.0-udp udp 3 96 0.0.0.0:5060
I’m not sure I know how to turn the pjsip logger on?
Really appreciate your help so far.
Thanks Stewart, moving the expiration time to 120 has taken care of my connection issues. Really appreciate your help!!!
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