[asterisk-bugs] Asterisk not sending SIP Info with SendDTMF(F)

[Rafael Morita]

Hi,

I have to send a SIP Info message with flashing signal to an ATA with FXO
port. It will hookflash the analogic port.
The call is been dialed from the same ATA FXO port.

The scenario I want to get is: PBX calls FXO -> ATA send INVITE to Asterisk
(SIP) -> SIP Info (F DTMF) to ATA -> FXO hookflash

Here's the context of extensions.conf

[ivrfxo]
exten => 1000,1,read(NUM,en/vm-enter-num-to-call)
exten => 1000,2,SendDTMF(F)
exten => 1000,3,Hangup


And the peer in sip.conf

[atafxo]
type=peer
username=atafxo
fromuser=atafxo
secret=123456
dtmfmode=info
host=dynamic
context=ivrfxo


After the asterisk receives the invite, looking in the CLI, here's the
print:

Connected to Asterisk 1.8.16.0 currently running on balalaika (pid = 4410)
Verbosity was 0 and is now 39
  == Using SIP RTP CoS mark 5
    -- Executing [1000 at urafxo:1] Read("SIP/atafxo-00000002",
"NUM,en/vm-enter-num-to-call") in new stack
    -- <SIP/atafxo-00000002> Playing 'en/vm-enter-num-to-call.gsm'
(language 'en')
    -- User entered '000'
    -- Executing [1000 at urafxo:2] SendDTMF("SIP/atafxo-00000002", "F") in
new stack
    -- Auto fallthrough, channel 'SIP/atafxo-00000002' status is 'UNKNOWN'

But capturing the packets, there is no SIP Info leaving the asterisk.


What can I do?


Thanks in advance,
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