"overnight" asterisk sets SDP RTP port to 0.0.0.0

Asterisk Asterisk SIP
1

From one day to the other, I have the issue that asterisk is setting the RTP port to 0.0.0.0, so the extensions are silent or say the call is on hold (which is right according to RFC rules). It happens on internal calls as well as on external. if I set “pjsip set logger on” i see

v=0
o=- 1318079683 1318079683 IN IP4 0.0.0.0
s=Asterisk
c=IN IP4 0.0.0.0

Here is the full log of a call between between extension 74@192.168.1.8 to 78@192.168.1.125 via asterisk@192.168.1.7: https://markdownshare.com/view/9cb1b780-b054-4b42-a7b9-ab96c5338cdf

I could only find that asterisk should set it’s own IP, so i have no clue why it sets 0.0.0.0. While being connected to the machine via ssh, i tried to restart asterisk (as network seems as I use ssh) but it stays the same. I am clueless.

2

What version of Asterisk are you using? As well - what are the contents of /etc/hosts? Asterisk can use that to determine your local IP address to place in things.

3

I digged through all related conf files and found it’s a bug in freepbx which gets updated on my system by a cronjob. The update added

bind_rtp_to_media_address=yes
media_address=0.0.0.0

for each extension. I opened a bug report for FreePBX.

https://issues.freepbx.org/browse/FREEPBX-15009

Thanks jcolp for your fast reply, I first opened issues for FreePBX as i suspected it to be the reason for this, but didn’t get feedback for 2 days. So i highly value your reply within minutes!

4

Ah yes, we take the value there as-is and put it in.