Hi,<div><br></div><div>I have to send a SIP Info message with flashing signal to an ATA with FXO port. It will hookflash the analogic port.</div><div>The call is been dialed from the same ATA FXO port.</div><div><br></div>
<div>The scenario I want to get is: PBX calls FXO -> ATA send INVITE to Asterisk (SIP) -> SIP Info (F DTMF) to ATA -> FXO hookflash</div><div><br></div><div>Here's the context of extensions.conf</div><div><br>
</div><div>[ivrfxo]</div><div><div>exten => 1000,1,read(NUM,en/vm-enter-num-to-call)</div><div>exten => 1000,2,SendDTMF(F)</div><div>exten => 1000,3,Hangup</div></div><div><br></div><div><br></div><div>And the peer in sip.conf</div>
<div><br></div><div><div>[atafxo]</div><div>type=peer</div><div>username=atafxo</div><div>fromuser=atafxo</div><div>secret=123456</div><div>dtmfmode=info</div><div>host=dynamic</div><div>context=ivrfxo</div></div><div><br>
</div><div><br></div><div>After the asterisk receives the invite, looking in the CLI, here's the print:</div><div><br></div><div><div>Connected to Asterisk 1.8.16.0 currently running on balalaika (pid = 4410)</div><div>
Verbosity was 0 and is now 39</div><div> == Using SIP RTP CoS mark 5</div><div> -- Executing [1000@urafxo:1] Read("SIP/atafxo-00000002", "NUM,en/vm-enter-num-to-call") in new stack</div><div> -- <SIP/atafxo-00000002> Playing 'en/vm-enter-num-to-call.gsm' (language 'en')</div>
<div> -- User entered '000'</div><div> -- Executing [1000@urafxo:2] SendDTMF("SIP/atafxo-00000002", "F") in new stack</div><div> -- Auto fallthrough, channel 'SIP/atafxo-00000002' status is 'UNKNOWN'</div>
</div><div><br></div><div>But capturing the packets, there is no SIP Info leaving the asterisk.</div><div><br></div><div><br></div><div>What can I do?</div><div><br></div><div><br></div><div>Thanks in advance,</div>