Asterisk 20.11.1 on Debian 12: Only “100 Trying,” No Channel Created, Never Answers Calls

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

  1. 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.
  1. 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

  1. 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.

  1. 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.

  1. 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.
  2. Trying a Simpler Dialplan : Changed [from-internal] to Wait(5) or Playback(hello-world) instead of Echo(). No change—still only 100 Trying.
  3. No Local Firewall : Confirmed no iptables or other filters on this VPS. The traffic definitely arrives at Asterisk (verified via tcpdump).
  4. 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.
  5. 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.

There are no known issues as such. It is highly likely your setup.

Are you selectively loading modules in modules.conf? There are a ton of PJSIP ones, including a few for calls. If those aren’t loaded then stuff like this will happen.

1 Like

Yes I was and that was simply the problem.

In the beginning of my configurations I messed a little with modules, so I was using the old chan_sip.so module but then configuring the pjsip.conf and also, even worse, doing noload => chan_pjsip.so

So yeah. By commenting those two, all works like a charm now! :slight_smile:

Thank you so much.