Just installed FusionPBX on a VPS - migrating from FreePBX. Its completely cloud instadd all my SIP clients are coming from the internet - there is no local LAN traffic.
Two of the features that I am struggling with are video calls and SIP SIMPLE archiving
1. Unable to make video calls: I reviewed the post https://www.pbxforums.com/threads/solved-video-calls-arent-working.3268/#post-10484 and added H264 in my global codec. but video calls still fail. Extract of displayed from fs_cli below

2020-05-18 11:41:49.312115 [DEBUG] sofia.c:7325 Channel sofia/internal/3001@box10.mydomain.net:5060 entering state [received][100]
2020-05-18 11:41:49.312115 [DEBUG] sofia.c:7335 Remote SDP:
v=0
o=- 4543661351 52263 IN IP4 89.211.168.41
s=extuliz
c=IN IP4 89.211.168.41
t=0 0
m=audio 32996 RTP/AVP 0 8 18 101
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
m=video 27446 RTP/AVP 108 99
a=rtpmap:108 VP8/90000
a=fmtp:108 max-fr=30;max-fs=3600
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=42800d;packetization-mode=1;level-asymmetry-allowed=1
2020-05-18 11:41:49.312115 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[G722:9:8000:20:64000:1]
2020-05-18 11:41:49.312115 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
2020-05-18 11:41:49.312115 [DEBUG] switch_core_media.c:5647 Audio Codec Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match
2020-05-18 11:41:49.312115 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
2020-05-18 11:41:49.312115 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[G722:9:8000:20:64000:1]
2020-05-18 11:41:49.312115 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
2020-05-18 11:41:49.312115 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
2020-05-18 11:41:49.312115 [DEBUG] switch_core_media.c:5647 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match
2020-05-18 11:41:49.312115 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [G729:18:8000:20:8000:1]/[G722:9:8000:20:64000:1]
2020-05-18 11:41:49.312115 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [G729:18:8000:20:8000:1]/[PCMU:0:8000:20:64000:1]
2020-05-18 11:41:49.312115 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [G729:18:8000:20:8000:1]/[PCMA:8:8000:20:64000:1]
2020-05-18 11:41:49.312115 [DEBUG] switch_core_media.c:5508 Set telephone-event payload to 101@8000
2020-05-18 11:41:49.312115 [DEBUG] switch_core_media.c:5851 Set telephone-event payload to 101@8000
2020-05-18 11:41:49.312115 [DEBUG] switch_core_media.c:5909 sofia/internal/3001@box10.mydomain.net:5060 Set 2833 dtmf send payload to 101 recv payload to 101
2020-05-18 11:41:49.312115 [DEBUG] switch_core_media.c:6237 No matches with FTMP, fallback to ignoring FMTP
2020-05-18 11:41:49.312115 [DEBUG] switch_core_media.c:6245 No matches with inherit_codec, fallback to ignoring PT
2. SIP SIMPLE archiving: SIP SImple messaging is working fine between extensions, but the v_messages table in the freeswitch database is blank. The only messages that it is storing are messages send from the FusionPBX GUI (FUSION PBX -> Applications -> Messages)
Is there any way to store the messages between the extensions in the Postgress database?
Two of the features that I am struggling with are video calls and SIP SIMPLE archiving
1. Unable to make video calls: I reviewed the post https://www.pbxforums.com/threads/solved-video-calls-arent-working.3268/#post-10484 and added H264 in my global codec. but video calls still fail. Extract of displayed from fs_cli below

2020-05-18 11:41:49.312115 [DEBUG] sofia.c:7325 Channel sofia/internal/3001@box10.mydomain.net:5060 entering state [received][100]
2020-05-18 11:41:49.312115 [DEBUG] sofia.c:7335 Remote SDP:
v=0
o=- 4543661351 52263 IN IP4 89.211.168.41
s=extuliz
c=IN IP4 89.211.168.41
t=0 0
m=audio 32996 RTP/AVP 0 8 18 101
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
m=video 27446 RTP/AVP 108 99
a=rtpmap:108 VP8/90000
a=fmtp:108 max-fr=30;max-fs=3600
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=42800d;packetization-mode=1;level-asymmetry-allowed=1
2020-05-18 11:41:49.312115 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[G722:9:8000:20:64000:1]
2020-05-18 11:41:49.312115 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
2020-05-18 11:41:49.312115 [DEBUG] switch_core_media.c:5647 Audio Codec Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match
2020-05-18 11:41:49.312115 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
2020-05-18 11:41:49.312115 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[G722:9:8000:20:64000:1]
2020-05-18 11:41:49.312115 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
2020-05-18 11:41:49.312115 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
2020-05-18 11:41:49.312115 [DEBUG] switch_core_media.c:5647 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match
2020-05-18 11:41:49.312115 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [G729:18:8000:20:8000:1]/[G722:9:8000:20:64000:1]
2020-05-18 11:41:49.312115 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [G729:18:8000:20:8000:1]/[PCMU:0:8000:20:64000:1]
2020-05-18 11:41:49.312115 [DEBUG] switch_core_media.c:5592 Audio Codec Compare [G729:18:8000:20:8000:1]/[PCMA:8:8000:20:64000:1]
2020-05-18 11:41:49.312115 [DEBUG] switch_core_media.c:5508 Set telephone-event payload to 101@8000
2020-05-18 11:41:49.312115 [DEBUG] switch_core_media.c:5851 Set telephone-event payload to 101@8000
2020-05-18 11:41:49.312115 [DEBUG] switch_core_media.c:5909 sofia/internal/3001@box10.mydomain.net:5060 Set 2833 dtmf send payload to 101 recv payload to 101
2020-05-18 11:41:49.312115 [DEBUG] switch_core_media.c:6237 No matches with FTMP, fallback to ignoring FMTP
2020-05-18 11:41:49.312115 [DEBUG] switch_core_media.c:6245 No matches with inherit_codec, fallback to ignoring PT
2. SIP SIMPLE archiving: SIP SImple messaging is working fine between extensions, but the v_messages table in the freeswitch database is blank. The only messages that it is storing are messages send from the FusionPBX GUI (FUSION PBX -> Applications -> Messages)
Is there any way to store the messages between the extensions in the Postgress database?
Last edited: