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I have a FreePBX 13 server set up with a SIP Trunk connection, however for some reason we are not getting the ring back tone for calls going out of the trunk connection.

I was able to implement a work around for this by placing the "Tr" options under "Asterisk Trunk Dial Options" to force Asterisk to produce the ring back tone for outbound calls.

However this only works while manually dialing from a soft-phone / VoIP Phone, when I try to launch a call via the Asterisk AMI "Originate" command we are not getting the ring back tone, even when the "r" option is set on the trunk. This is how the command I'm sending looks:

Action: Originate,
Channel: SIP/{extension},
Context: from-internal,
Exten: {phoneNumber},
Priority: 1,
Callerid: {callerId},
Timeout: 30000,
Async: yes

Any idea about what can I do to force the ring back tone?

Ox3
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3 Answers3

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This could be due to inband progress. Try to add the following to your sip.conf under [general] section,

prematuremedia=no    
progressinband=yes

if you are using freepbx, then navigate to your trunk settings in the user interface and add the above 2 configs under the peer settings.

restart asterisk and try after that.

Sadeesh Radhakrishnan
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Launch Originate via Local/ channel call, if still not work, launch via custom context which answer first.

arheops
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This is not an uncommon problem. Your ITSP is being lazy and either not providing or not forwarding RFC-complaince SIP 180/183 responses.

My usual solution is to put the local caller into an MOH state where the MOH is a ring tone, and then when the other end picks up, bridge the two calls.

This is kind of jarring, however, if the remote end is busy or is congested, since you go from "fake ring" to "real error tone". The other option is to reverse your dialing process ... call the remote end and then hook the local end, so that the local end doesn't need to hear ringing.

Of course, you can also reach out to your ITSP and tell them you need them to be RFC SIP 180/183 complaint.

MichelV69
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