Outboud Call drops voice after 333 seconds

FreePBX Endpoints
1

Hello,

a FreePBX 15.0.16.49 is used behind an OPNSense.
A PJSIP trunk to a LAN phone of a Fritzbox is configured.

Tab General:

  • Trunk Name “Fritzbox-Unitymedia”
  • Hide CallerID “No”
  • Outbound CallerID “<02***********>”
  • CID Options “Allow Anny CID”
  • Maximum Channels “2”
  • Asterisk Trunk Dial Options “Ttr”, "System
  • Continue if Busy “No”
  • "Disable Trunk “No”
  • Monitor Trunk Failires “No”

Tab “pjsip Settings”, “General”

  • Username “02*********”
  • Secret “*******************”
  • Authentication “Outbound”
  • Language Code “Default”
  • SIP server “Fritzbox IP”
  • SIP Server Port “5060”
  • Contect “from-ptsn”
  • Transport “0.0.0.0-udp”

Tab “pjsip Settings”, “Advanced”

  • DTMF Mode “Auto”
  • Send Connected Line “No”
  • Permanent Auth Rejection “No”
  • Forbidden Retry Interval “30”
  • Fatal Retry Interval “30”
  • General Retry interval “60”
  • Expiration “3600”
  • Max Retries “10000”
  • Qualify Frequency “60”
  • Outbound Proxy “”
  • User = Phone “No”
  • Contact User “Fritzbox LAN-Phone User”
  • From domain “Fritzbox IP”
  • From User “Fritzbox LAN-Phone User”
  • Client URI “”
  • Server URI “”
  • Media Address “”
  • AOR “”
  • AOR Contact “”
  • Match (Permit) “”
  • Support Path “No”
  • Support T.38 UDPTL "No
  • T.38 UDPTL Error Correction "None
  • T.38 UDPTL NAT "No
  • T.38 UDPTL MAXDATAGRAM “”
  • Fax Detect “No”
  • Trust RPID/PAI "No
  • Send RPID/PAI “No”
  • Send Private CallerID Information "Default
  • Inband Progress “No”
  • Direct Media “No”
  • Rewrite Contact “No”
  • RTP Symmetric “Yes”
  • Media Encryption “None”
  • Force rport “Yes”
  • Message Context “”

Tab “pjsip Settings”, “Codecs”

  • g722
  • alaw
  • ulaw
  • gsm
  • g726

Outbound Route:
Tab “Route Settings”

  • Route name “Unitymedia”
  • Route CID “”
  • Override Extension “No”
  • Route Password “”
  • Route Type “”
  • Music On Hold “default”
  • Time Match Time Zone "Use System Timezone
  • Time match Time Group “—Permanent Route --”
  • Trunk Sequence for Matched Routes “Fritzbox-Unitymedia”
  • Optional Destination on Congestion "Normal Congestion
    Note: Extension Routes is not registered

Tab “Dial Patterns”

  • +XXX.
  • XXX.

Tab “Additional Settings”

  • Call Recording “Don’t Care”
  • Notifications “None”
  • PIN set “None”

External calls work.
Internal to internal calls work.
For internal-to-external calls, no voice is transmitted in either direction after approx. 332, 333 seconds!

What could be the reason for this?
In sngrep and Wireshark I cannot see anything. OPNSense logs everything, but shows no abnormalities.

Translated with www.DeepL.com/Translator (free version)

2

/var/log/asterisk/full also shows no abnormalities despite asterisk -rx “sip set debug on”.

3

You’re using Chan_PJSIP, not Chan_SIP. Using chan_sip debug tools isn’t going to help you. You need to use pjsip set logger on to do get the SIP message debug.

4

300 second is 5 minutes, right?

33 seconds the amount of time it takes after no audio is received to tear down a call.

Sounds like the NAT firewall is dropping the NAT association after 5 minutes, and the PBX is shutting down the call 30 seconds later.

Of course, that’s just a guess.

5

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