Phantom SIP peers

FreePBX
1

This has been a snit with me for a long time and wondering what is causing this:

Chan_Sip Peers

Name/username Host Dyn Forcerport Comedia ACL Port Status Description
90701/ekhsquj8 (Unspecified) D No No A 0 UNKNOWN
90703 (Unspecified) D No No A 0 UNKNOWN
99701 (Unspecified) D No No A 0 UNKNOWN
99703 (Unspecified) D No No A 0 UNKNOWN

The first one just showed up the other day. 99701 and 99703 showed up originally. 701 and 703 are real extensions using PJSIP. I have no idea with the 9070x came from either especially with that userid that looks like it iwas randomly gneerated.

while we’re on it under PJSIP there’s this:

Endpoint: dpma_endpoint Unavailable 0 of inf

I’m assuming thats for provisioning of some sort?
Thanks leon

2

The 99 prefix peers are WebRTC devices and the 90 prefix is for Zulu devices. You can disable them in User Management.

3

THanks lorne this was perplexing me for years. COuld the one with the random userid come from the beta soft client?
also what ports does the webrtc use as well as the soft client? I tried it the other day and got oneway audio and it also seemed to lock up on me leaving an extension “off hook”

Leon

4

The random userid is from using Zulu, that is normal. Ports used are as shown on this page:
https://wiki.freepbx.org/display/PPS/Ports+used+on+your+PBX

You will not have success with WebRTC until you set up a proper TLS certificate (let’s encrypt works great) and use https for UCP. You may also need to define a webrtc stun server in Asterisk SIP Settings.

5

I have a cert already setup and TLS is working for me. does a stun server need to be defined can’t it (pbx) be used directly for webrtc or zulu?

thanks leon

6

Hi, i have some phantom Chan_Sip peers too:

Name/username Host Dyn Forcerport Comedia ACL Port Status Description
999916 (Unspecified) D No No A 0 UNKNOWN
999918 (Unspecified) D No No A 0 UNKNOWN
999920 (Unspecified) D No No A 0 UNKNOWN
3 sip peers [Monitored: 0 online, 3 offline Unmonitored: 0 online, 0 offline]

All users (7 total) and groups have WebRTC and Zulu disabled in user manager, but still these 3 stays in config :frowning: Users don’t use UCP at all, only use calling via phones (pjsip extensions and trunks) and chatting via XMPP.

What else can cause creating (or not deleting) of these chan_sip peers?
Versions: Asterisk 13.15.0, FreePBX 10.13.66-20, Sangoma 6 Distro