Fongo Community Forums

mkaye wrote: 06/24/2024 created new pjsip trunk, incoming calls terminate after 30s

mkaye wrote: 07/07/2024 Contact User solved the pjsip register problem
but have the 30s call terminated problem with pjsip, not with chan_sip

gideons wrote:I decided to give it a try and add back the Contact User.
And so as I am making the call and following the logs, what a surprise: I couldn’t believe my eyes: an ACK !
Then the call went beyond the 32 seconds."

mkaye wrote: 07/12/2024 i don't see any option in the pjsip advanced parameters to disable timers (FreePBX17)

dzone wrote:We were able to solve the 15-minute hangup problem by using the pjsip.endpoint_custom_post.conf file. By adding “timers=no” to the trunk, the calls stopped timing out. I wish there had been a way to configure this in the pjsip trunk settings.

pjsip.endpoint_custom_post.conf

[mytrunk](+)
timers=no

Notice the (+) after the trunk name. It appears to need that to append to the other settings FreePBX manages.

mkaye wrote: 07/15/2024 didn't fix it
my trunk is called fpl
i used [fpl](+)

192.168.1.10

• External Address: Click "Detect Network Settings". Verify the detected IP is your public IP. If not, enter your Static IP manually if you have one.
• Local Networks: Click "+ Add Network" and manually enter your internal network(s). Ex.

  Code: Select all

  192.168.1.0 / 255.255.255.0

  .

0.0.0.0 (udp)

53060

• Go back to your router's configuration page, find "Port Forwarding".
• Create rules to forward incoming UDP traffic:
  • From Port

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    53060

    to your FreePBX server's internal static IP on port

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    53060

    . Use your chosen port.
  • From Port Range

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    10000

    -

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    20000

    to your FreePBX server's internal static IP for ports

    Code: Select all

    10000

    -

    Code: Select all

    20000

    . This is for RTP audio.
• Save the changes on your router.

53060

10000-20000

• Trunk Name:

  Code: Select all

  Freephoneline

• Outbound CallerID:

  Code: Select all

  <YOUR_FPL_NUMBER>

  (Use angle brackets around your 10-digit number. For Display Name use

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  "Your Name" <YOUR_FPL_NUMBER>

• Maximum Channels:

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  2

• Username:

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  YOUR_FPL_USERID

• Secret:

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  YOUR_FPL_PASSWORD

• Authentication: Select

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  Outbound

• Registration: Select

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  Send

• SIP Server:

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  voip.freephoneline.ca

  (Primary Server)
• SIP Server Port:

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  5060

  (Note: This is FPL's destination port, not your local listening port.)

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  53060

• Context:

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  from-trunk

• Transport: Select the UDP transport listening on

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  53060

  . Ensure this matches Step 10.

• Match (Permit):

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  voip.freephoneline.ca,voip2.freephoneline.ca,voip4.freephoneline.ca

  (Recommended)
• Contact User:

  Code: Select all

  YOUR_FPL_USERID

• From Domain:

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  voip.freephoneline.ca

  (Match the SIP Server field above.)
• From User:

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  YOUR_FPL_USERID

• Client URI:

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  sip:YOUR_FPL_USERID@voip.freephoneline.ca

  (Match the SIP Server field above.)
• Server URI:

  Code: Select all

  sip:voip.freephoneline.ca

  (Match the SIP Server field above.)
• AOR:

  Code: Select all

  sip:YOUR_FPL_USERID@voip.freephoneline.ca

  (Match the SIP Server field above.)
• Qualify Frequency:

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  20

  (Matches FPL Keep Alive Interval.)
• Registration Expiry:

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  3600

  (Matches FPL Recommendation.)
• Registration Retry Interval:

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  120

  (Matches FPL Recommendation.)
• Session Timers:

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  No

• Rewrite Contact:

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  Yes

  (This setting is important for NAT and ensuring your WAN IP is used correctly.)
• Force rport:

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  Yes

  (This setting is important for NAT and ensuring your WAN IP is used correctly.)
• Send RPID/PAI:

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  Send PAI

• Trust RPID:

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  Yes

• DTMF Mode:

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  RFC 4733

  (This is the most reliable method for touch-tones.)
• Direct Media:

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  No

  (This setting is important when behind NAT.)

• Check only

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  ulaw

  and

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  g729

  . Uncheck all other codecs. Ensure these are the ONLY enabled codecs as recommended by FPL.

ptime=20

• Route Name:

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  FPL-Outgoing

• Trunk Sequence for Matched Routes: Select

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  Freephoneline

  (the trunk from Step 16).

    [2-689]11
    *98
    NXXNXXXXXX
    1NXXNXXXXXX
    011.
    

• Description:

  Code: Select all

  FPL-Incoming

• DID Number:

  Code: Select all

  YOUR_FPL_NUMBER

Extension

Ring Group

IVR

• GUI Method: Go to Reports -> Asterisk Info -> Registries tab. Look for your

  Code: Select all

  Freephoneline

  entry. Status should say

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  Registered

  .
• CLI Method: Log into your server via SSH, type

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  asterisk -rvvv

  . At the prompt, type

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  pjsip show registrations

  and press Enter. Check the status. Type

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  exit

  to leave.

*98

voip.freephoneline.ca

• Go back to Connectivity -> Trunks. Click the pencil icon to edit your

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  Freephoneline

  trunk.
• Go to the pjsip Settings -> General tab.
• Choose ONE Alternative:
  • Option 1: Change SIP Server to

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    voip2.freephoneline.ca

    . Keep SIP Server Port as

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    5060

    .
  • Option 2: Change SIP Server to

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    voip4.freephoneline.ca

    AND change SIP Server Port to

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    6060

    .
• Go to the pjsip Settings -> Advanced tab. IMPORTANT: Update the From Domain, Client URI, Server URI, and AOR fields to match the new SIP Server hostname you just entered. This is important!
• Click Submit.
• Click the red Apply Config button.
• Wait a minute or two and re-check registration status (Step 33). Test calls again.

• Registration Failed: Double-check Username/Password. Verify firewall rules (UDP

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  53060

  & RTP

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  10k-20k

  ). Check Logs (Reports -> Asterisk Logfiles).
• No Audio / One-Way Audio: Check NAT settings (Step 7). Check router Port Forwarding (UDP

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  53060

  &

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  10k-20k

  ). Check

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  Direct Media: No

  (Step 18).
• Calls Don't Connect: Check logs. Ensure SIP ALG is OFF on router. Check Outbound Route Dial Patterns (Step 25).

192.168.1.10

• External Address: Click "Detect Network Settings". Verify the detected IP is your public IP. If not, enter your Static IP manually if you have one.
• Local Networks: Click "+ Add Network" and manually enter your internal network(s). Ex.

  Code: Select all

  192.168.1.0 / 255.255.255.0

  .

53060

• Go back to your router's configuration page, find "Port Forwarding".
• Create rules to forward incoming UDP traffic:
  • From Port

    Code: Select all

    53060

    to your FreePBX server's internal static IP on port

    Code: Select all

    53060

    . Use your chosen port.
  • From Port Range

    Code: Select all

    10000

    -

    Code: Select all

    20000

    to your FreePBX server's internal static IP for ports

    Code: Select all

    10000

    -

    Code: Select all

    20000

    . This is for RTP audio.
  [\*] Save the changes on your router.

53060

10000-20000

• Trunk Name:

  Code: Select all

  Freephoneline-ChanSIP

• Outbound CallerID:

  Code: Select all

  \<YOUR\_FPL\_NUMBER\>

  (Use angle brackets around your 10-digit number. For Display Name use

  Code: Select all

  "Your Name" \<YOUR\_FPL\_NUMBER\>

  )[/i]
  [\*] Maximum Channels:

  Code: Select all

  2

• Trunk Name:

  Code: Select all

  FreephonelineSIP

  (Must match PEER name below, no spaces)
• PEER Details: (Paste the entire block below)

  Code: Select all

      type=peer
      host=voip.freephoneline.ca
      port=5060
      username=YOUR\_FPL\_USERID
      secret=YOUR\_FPL\_PASSWORD
      fromuser=YOUR\_FPL\_USERID
      fromdomain=voip.freephoneline.ca
      context=from-trunk
      transport=udp
      nat=force\_rport,comedia
      qualify=20000
      insecure=port,invite
      dtmfmode=rfc2833
      disallow=all
      allow=ulaw\&g729
      trustrpid=yes
      sendrpid=yes
      session-timers=refuse
      directmedia=no
      

  (Note:

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  qualify=20000

  sets check interval to 20 seconds (in ms) matching FPL keep-alive. Adjust host/port here if manually failing over later.)[/i]

• USER Context: (Leave blank or enter

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  YOUR\_FPL\_USERID

  )
• USER Details: (Leave blank)
  [\*] Register String: (Paste the line below)

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  YOUR\_FPL\_USERID:[email address removed]:5060\~3600/YOUR\_FPL\_USERID

  (This explicitly sets the destination server/port and the 3600 second expiry. Adjust host/port here if manually failing over later.)

• Route Name:

  Code: Select all

  FPL-ChanSIP-Out

• Trunk Sequence for Matched Routes: Select

  Code: Select all

  Freephoneline-ChanSIP

  (the trunk from Step 17).

    [2-689]11
    \*98
    NXXNXXXXXX
    1NXXNXXXXXX
    011\.
    

• Description:

  Code: Select all

  FPL-ChanSIP-In

• DID Number:

  Code: Select all

  YOUR\_FPL\_NUMBER

Extension

Ring Group

IVR

• GUI Method: Go to Reports -\> Asterisk Info -\> Registries tab. Look for your

  Code: Select all

  Freephoneline-ChanSIP

  entry. The State should say

  Code: Select all

  Registered

  .
• CLI Method: Log into your server via SSH, type

  Code: Select all

  asterisk -rvvv

  . At the prompt, type

  Code: Select all

  sip show registry

  and press Enter. Check the state. Type

  Code: Select all

  exit

  to leave.

\*98

voip.freephoneline.ca

• Go back to Connectivity -\> Trunks. Click the pencil icon to edit your

  Code: Select all

  Freephoneline-ChanSIP

  trunk.
• Go to the sip Settings -\> Outgoing tab.
• Edit PEER Details: Change the

  Code: Select all

  host=

  and

  Code: Select all

  port=

  lines to an alternative:
  • Option 1:

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    host=voip2.freephoneline.ca

    , keep

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    port=5060

    . Update `fromdomain=` too.
  • Option 2:

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    host=voip4.freephoneline.ca

    , change

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    port=6060

    . Update `fromdomain=` too.
• Go to the sip Settings -\> Incoming tab.
• Edit Register String: Change the hostname and port after the `@` symbol to match the PEER details you just set. Ex:

  Code: Select all

  YOUR\_FPL\_USERID:[email address removed]:6060\~3600/YOUR\_FPL\_USERID

  .
• Click Submit.
• Click the red Apply Config button.
• Wait a minute or two and re-check registration status (Step 32). Test calls again.

• Registration Failed: Double-check Username/Password. Verify firewall rules (UDP

  Code: Select all

  53060

  & RTP

  Code: Select all

  10k-20k

  ). Check Logs (Reports -\> Asterisk Logfiles).
• No Audio / One-Way Audio: Check NAT settings (Step 7). Check router Port Forwarding (UDP

  Code: Select all

  53060

  &

  Code: Select all

  10k-20k

  ). Check

  Code: Select all

  nat=force\_rport,comedia

  (Step 18). Check

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  Direct Media: No

  (Step 18).
  [\*] Calls Don't Connect: Check logs. Ensure SIP ALG is OFF on router. Check Outbound Route Dial Patterns (Step 24).