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.