Stefan Helander

FreePBX/Asterisk missing DID; how to force DID on trunk

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If you get a SIP account from a provider which isn’t really a trunk it can be configured as a trunk in FreePBX but DID on incoming calls is set to “s” so it is impossible to get an inbound route working.

The trunk in this case will only have one DID so a workaround is to set the DID “hardcoded” on the trunk in order to get the inbound route working.

Edit extensions_custom.conf and add something like this, but replace “123456789” with the DID you wish to set:

[custom-123456789]
exten => s,1,Noop(Setting DID to 123456789)
exten => s,n,Goto(from-trunk,123456789,1)

Now edit the trunk and go to the pjsip tab and enter custom-123456789 (replacing “123456789” with your DID) in the Context field.

Now all incoming calls in this trunk will get your desired DID and your inbound route will be able to catch them by the DID.

Stefan Helander

FreePBX extension receives call from wrong queue

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It turned out that the extension earlier had been a dynamic agent for the queue, and it was logged in as an active agent for the queue when the extension was removed from the queue. This caused calls in the queue being sent to the extension still.

Solution: Edit the Queue, add the extension as a dynamic agent, call *45 from the extension (“agent logged off”), dial *45 again (“agent logged in”) and finally dial *45 one more time (“agent logged off”). Now edit the Queue again and remove the extension from dynamic agents.

Stefan Helander

FreePBX multiple time conditions

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The easiest way to get multiple time conditions in FreePBX is to daisy chain them. A common scenario is to have a standard time condition for normal opening hours and in addition to that, time conditions for holidays occuring on normal weekdays, like christmas and new year for example.

  • Create one time group for normal opening hours, for example monday to friday 08.00-17.00.
  • Create one time condition for the normal opening hours, using the above time group. If time matches, connect the destination to the open IVR or queue. If time doesn’t match, connect it to a closed IVR or a voice message saying you’re closed. This is going to be the last time condition in the chain.
  • Create time groups for each of the holidays. If time matches, connect the destination to a voice message saying you’re closed. If time condition doesn’t match connect it to the next time condition. Remember, the last time condition in the chain should be the one for normal opening hours.
  • Connect the inbound route to the first time condition in your chain.

Inbound route -> Time condition for holiday 1 -> Time condition for holiday 2 -> Time condition for normal opening hours

Stefan Helander

Asterisk / FreePBX sip trunk registration problem, Serious Network Trouble

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The asterisk log file (/var/log/asterisk/full) shows entries like this:

[Sep  3 04:02:08] ERROR[3984] chan_sip.c: Serious Network Trouble; __sip_xmit returns error for pkt data

Solution: The server had been moved from one public IP-address to another. In Asterisk PBX settings, the fields for both External IP and Bind Address (under Advanced) needed adjustment to the new IP-address. After server reload everything worked normally.

Stefan Helander

Asterisk no sound from client

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I was running an Asterisk server behind NAT without any problems at one location. Due to reoargnization it was moved to a new location and placed behind a pfsense firewall. After a while users reported increasing problems with no sound from the clients.

It turned out running a SIP server behind a pfsense can be problematic. When moving to the SIP server to a white IP-adress (no NAT), protecting it using iptables firewall on the SIP-server, all problems disappeared.

It should be noted that the pfsense firewall was not running the latest version of pfsense so it could be caused by a problem solved in later versions of pfsense. However, for different reasons it was not possible to upgrade pfsense so the solution was to move the SIP-server from behind pfsense.

Stefan Helander

SIP-client behind NAT disconnects incoming external call after 8-10 seconds

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Scenario: a SIP server running Elastix (Asterisk and FreePBX) with SIP-clients behind NAT. Outgoing calls from the clients (which where CounterPath Xten and Siemens Gigaset C530ip, i.e. completely different brands) works well, but incoming calls are disconnected after 8-10 seconds.

Googleing the problem suggests the problem is caused by a SIP REINVITE on the client side, which can’t happen because of NAT. During testing, I discovered that calls between SIP-phones connected to the same SIP PBX worked fine, even between different locations. Only calls that originated externally would be interrupted.

This made me focus on the trunks instead and I found out that the problem indeed was a REINVITE but it occured on the trunk side.

The solution was to add the following lines to the PEER DETAILS and USER DETAILS on the trunk configuration in the SIP server:

directmedia=no
canreinvite=no