Hello everyone,
I have a Debian 12 VPS (OVH) running Asterisk 20.11.1. My goal is to receive inbound calls on a DID from my SIP trunk providers. I’ve tried also to place test calls from a Windows softphone (MicroSIP). However, Asterisk never proceeds beyond 100 Trying for inbound calls, and I see no channel created when I run core show channels.
Environment
- Host : OVH VPS with 1 vCore, 2GB RAM, 20GB SSD
- Public IP : SERVERIP (no NAT on the server side)
- OS: Debian 12
- Asterisk Version : 20.11.1 (installed from source)
- Firewall : None active (for now, to eliminate blocking issues)
Symptoms
- Inbound Calls from my SIP trunk:
- Asterisk sees the INVITE, replies 100 Trying, but never sends a 200 OK.
- core show channels always shows 0 active channels.
- Internal Calls from MicroSIP to extension 700:
- Same pattern: INVITE arrives, Asterisk replies 100 Trying, then the phone cancels after waiting.
- No channel is created (core show channels → 0).
PJSIP Configuration
Relevant portion of /etc/asterisk/pjsip.conf:
[transport-udp]
type=transport
protocol=udp
bind=0.0.0.0:5060
external_signaling_address=SERVERIP
external_media_address=SERVERIP
; local_net=192.168.1.0/24 ; (Uncomment if needed)
[trunk]
type=endpoint
transport=transport-udp
context=from-trunk
disallow=all
allow=ulaw
allow=alaw
aors=trunk_aor
[trunk_aor]
type=aor
contact=sip:eu.st.ssl7.net ;Using VoIPstudio.com with GoTrunk.com
[trunk_identify]
type=identify
endpoint=trunk
; Some known IPs from my SIP provider
match=109.233.115.107
match=69.164.221.134
; I also tried adding match=134.19.166.167 or other IPs from logs
[testar]
type=endpoint
transport=transport-udp
context=from-internal
disallow=all
allow=ulaw
allow=alaw
auth=testar_auth
aors=testar_aor
direct_media=no
rtp_symmetric=yes
force_rport=yes
rewrite_contact=yes
[testar_auth]
type=auth
username=testar
password=testar
[testar_aor]
type=aor
max_contacts=1
[testar_identify]
type=identify
endpoint=testar
match=188.82.89.2
Extensions Configuration
Relevant portion of /etc/asterisk/extensions.conf:
[from-internal]
exten => 700,1,Answer()
same => n,Echo()
same => n,Hangup()
[from-trunk]
exten => _X.,1,NoOp(#### Incoming call [from-trunk] ####)
same => n,Answer()
same => n,Playback(demo-congrats)
same => n,Hangup()
What I Observe in Logs
Example inbound call from the trunk (public IP):
<--- Received SIP request (INVITE) from UDP:109.233.115.107:5060 --->
...
<--- Transmitting SIP response to UDP:109.233.115.107:5060 --->
SIP/2.0 100 Trying
...
<--- (No further 200 OK) --->
Shortly afterward:
CANCEL sip:PHONENUMBER@SERVERIP:5060
SIP/2.0 200 OK
SIP/2.0 487 Request Terminated
core show channels → 0 active channels.
The same pattern occurs for internal calls from MicroSIP (188.82.89.2) to extension 700. Asterisk logs show INVITE → 401 Unauthorized → re-INVITE → 100 Trying, then CANCEL after ~10s. No channel is ever created.
What I’ve Already Tried
- AstDB Permissions : I fixed a prior “read-only database” error by setting ownership:
sudo chown asterisk:asterisk /var/lib/asterisk/astdb.sqlite3
sudo chmod 664 /var/lib/asterisk/astdb.sqlite3
sudo chown -R asterisk:asterisk /var/lib/asterisk /var/spool/asterisk /var/log/asterisk /etc/asterisk
That removed the DB write warnings.
- Bridging Modules : Confirmed they’re all loaded:
module show like bridge
shows bridge_builtin_features.so, bridge_simple.so, bridge_native_rtp.so, etc. are Running.
- Match Additional IPs : I added match=134.19.166.167 (and other addresses) in my [trunk_identify] after seeing them in the logs, then did pjsip reload. Same result.
- Trying a Simpler Dialplan : Changed [from-internal] to Wait(5) or Playback(hello-world) instead of Echo(). No change—still only 100 Trying.
- No Local Firewall : Confirmed no iptables or other filters on this VPS. The traffic definitely arrives at Asterisk (verified via tcpdump).
- Check for Logging / Permission Errors : No new warnings or errors in /var/log/asterisk/full after 100 Trying. It simply doesn’t proceed to create a channel.
- core show channels always yields 0 active channels while the call is “ringing.”
Request for Help
I’m stuck because I see no obvious reason why Asterisk won’t create a channel and execute the dialplan. The inbound calls definitely arrive, Asterisk sends 100 Trying, but then it just stalls.
Questions:
- Are there any known issues with Asterisk 20.11.1 on Debian 12 that would cause no channel creation after receiving an INVITE?
- Could there be a module or config snippet I’m missing that preempts the normal channel flow?
- Is there a deeper debug or trace I can enable to see why Asterisk is halting after 100 Trying?
Thank you in advance for any suggestions! I’m open to any ideas or further debug steps.