Comments (8)
Hi,
can you give your restxml and your FS dialplan ?
Seems FS is queueing dtmfs but never flushes the buffer .
I see you're using FreeTDM, is there any parameter for setting dtmfs like dtmf length, ... ?
Maybe we need to explicitly flush dtmfs in buffer ... or another bug fro FreeTDM ... :p
from plivoframework.
public.xml (copied from Plivo install files):
<include>
<context name="public">
<extension name="unloop">
<condition field="${unroll_loops}" expression="^true$"/>
<condition field="${sip_looped_call}" expression="^true$">
<action application="deflect" data="${destination_number}"/>
</condition>
</extension>
<!--
Tag anything pass thru here as an outside_call so you can make sure not
to create any routing loops based on the conditions that it came from
the outside of the switch.
-->
<extension name="outside_call" continue="true">
<condition>
<action application="set" data="outside_call=true"/>
<action application="set" data="RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}"/>
</condition>
</extension>
<extension name="call_debug" continue="true">
<condition field="${call_debug}" expression="^true$" break="never">
<action application="info"/>
</condition>
</extension>
<!--
This extension allows calling any digits of number
freeswitch will call plivo outbound server on every incoming call
-->
<extension name="plivo_public_did">
<condition field="destination_number" expression="^(\d+)$">
<action application="enable_heartbeat" data="60"/>
<action application="socket" data="69.168.53.39:8084 async full"/>
</condition>
</extension>
</context>
</include>
restxml:
/inbound/answered
<Response>
<GetDigits action="http://127.0.0.1:5002/inbound/getdigits">
<Play>/usr/local/freeswitch/scripts/jovan-welcome.wav</Play>
</GetDigits>
</Response>
/inbound/getdigits
<Response>
<Play>/usr/local/freeswitch/scripts/jovan-thankyou.wav</Play>
</Response>
from plivoframework.
ok
can you retry with GetDigits and timeout='20' and paste the FS and plivo logs ?
from plivoframework.
also try using a sip line instead of a freetdm to see if it works
and try freetdm in FS XML dialplan using play_and_get_digits to see if it works too .
from plivoframework.
Any news ?
I have another idea .
try using http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_flush_dtmf before play_and_get_digits in FS dialplan .
from plivoframework.
Sorry for the delay. We've been rearranging our stack due to issues between Sangoma (our T1 card vendor), freetdm and Freeswitch. I just finished migrating our T1 box to Asterisk, which is now handling the TDM layer and sending the voice via SIP to Freeswitch. With that out of the way, I need to retest to see if the issue still exists. I should be able to give an answer by tomorrow.
from plivoframework.
ok perfect :)
from plivoframework.
I'm closing this as I've given up on running FS on a separate server for now. Running FS on the same system as Plivo works great, running FS on a separate system doesn't work and simply isn't important enough to keep fighting with.
from plivoframework.
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from plivoframework.