[Freeswitch-users] FS Bridged call, no RTP until DTMF pressed

[Adnan Ahmed]

You can disregard this message. Further testing and isolation of the
software/hardware found the cause to be in the chan_dahdi.c file for
asterisk.  (Not actually an error but it was what was causing me issues.)

The conf mute was being turned on because it thought it was receiving a
DTMFdown event on the E&m wink with Feature Group B MF signalling trunk.
Not sure if it was an error or anything, so I just hard coded the file to
not use confmute, and now it works fine.

Adnan.

On Thu, Feb 11, 2016, 09:32 Adnan Ahmed <adnan.ahmed1 at gmail.com> wrote:

> Hi,
>
> I have a peculiar situation in which I'm hoping someone can help me out
> with.  I have a Dahdi trunk coming into Asterisk (*), which then sends the
> call directly to freeswitch (FS), FS will then bridge this incoming call to
> a SIP device.  The problem i'm having is that when FS bridges the call
> there is no media (or RTP packets) sent back to asterisk until I press a
> dtmf key from the caller side.
>
> The reason that * is there is due to the fact that mod_freeTDM for FS
> wasn't able to configure the trunk parameters required to control the T1
> (E&M with Feature Group B MF), with chan_dahdi in * i was able to set that
> up with signalling=featb.  The dialplan in asterisk is as follows,
>
> [from-pstn]
>> exten => _X.,1,NoOp(Incoming DID matches as ${EXTEN})
>> exten => _X.,n,Answer()
>> exten => _X.,n,Set(CALLERID(all)="0000000000"<0000000000>)
>> exten => _X.,n,Dial(SIP/freeswitch/1819${EXTEN:0:7},90,M(send-dtmf-1)r)
>> exten => _X.,n,Hangup()
>>
>
>
>> [macro-send-dtmf-1]
>> exten => s,1,SendDTMF(1)
>
>
>  I tried sending a DTMF from astersk, and FS recognizes the DTMF, but
> still no RTP until the key is physically pressed on the caller side.  The
> asterisk dialplan is very simple, answer the incoming dahdi call and send
> it to FS via SIP.  Once the DTMF is pressed, the audio is complete and no
> issues anymore, so its not a routing, or firewall issue.  Both asterisk and
> FS run on the same machine (* on port 5065, and FS on 5060).  Looking at
> the tcpdump traces, there really is no RTP from FS until after the DTMF is
> pressed, but the RTP from asterisk is always there.
>
> I have the output of "sofia global siptrace on" at the following pastebin:
> https://pastebin.freeswitch.org/24552
>
> In that SIP trace you will see the call as follows,
>
> Incoming call from *
> bridge to SIP device
> Failure to connect to SIP device
> Forward call to voicemail
> bridge to voicemail
> connects to voicemail system
> hangup
>
> I can press the DTMF at any point once the first bridge is dialed and will
> start hearing the audio from that point onwards ... in this case i pressed
> the DTMF key 1 (you see it being recognized in the FS sip trace log).  It
> makes no difference if I wait to press the DTMF till the second bridge or
> after the second bridge connects.
>
> I have even tried it with a sip device that answers on the first bridge
> session, and its the same scenario:  no audio until dtmf is pressed, again
> making no difference if its pressed right away or 10 seconds after the call
> is connected and the other party can hear me but i don't hear them until i
> press the dtmf.
>
> Thanks,
> Adnan.
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20160215/1fee8520/attachment.html