Passing digits on an answered incoming call

Asterisk Asterisk Support
1

I recently signed up for incoming VoIP provider.
Now when a call comes in Asterisk can answer it and send the call to another extension using the Dial Application. However when asking the user to input the extension of his choice i get dead air and following error.

Feb 17 13:44:57 NOTICE[13991958]: rtp.c:289 process_rfc3389: RFC3389 support incomplete. Turn off on client if possible
– Timeout on SIP/72.11.195.3-f6e2e588

However, if I turn on SIP-Debug I can then dial the ext of my choice however turning that on breaks other things.

Any idea what the above error means and how i can fix it.

2

Hi everyone,
Please correct me if I am wrong.
My observation of some of my phones behaviour tells me that ‘RFC3389 - support incomplete’ is to do with unsupported codecs either on a phone or in the Asterisk.

I have an old Cisco ATA and a X-Lite, both have only codecs listed in RFC3389 document http://www.faqs.org/rfcs/rfc3389.html

The rfc3389 abstract says in the abstract:

This document describes a Real-time Transport Protocol (RTP) payload
format for transporting comfort noise (CN). The CN payload type is
primarily for use with audio codecs that do not support comfort noise
as part of the codec itself such as ITU-T Recommendations G.711,
G.726, G.727, G.728, and G.722.

Because of this, the phone cannot send inband DTMF (even if I force the phone settings) or may not receive MusicOnHold.

I am still looking for an answer how to ‘turn it off on the client’ or how to switch Comfort Noice off on the Asterisk. Any help will be greatly appreciated.

Cheers
Richard