[Freeswitch-users] No audio when calling in via SIP phone

Iqbal Abdullah iqbal.b.abdullah at gmail.com
Tue Jul 29 14:04:55 MSD 2014


Hello everyone,

I have just gotten to know about FreeSWITCH and have tried installing
it on a EC2 instance and in the process of trying out some examples. I
am facing this issue of not being able to hear any sounds during a
call I make to my FreeSWITCH system.

My freeswitch system is the latest stable ver.1.4 and I followed the
instructions to install from
http://clintberry.com/2011/getting-started-with-plivo/

via the installation script show on the site. I changed the
installation script to use Slash and also 1.4 instead.

One of the exercises is to pass an incoming call to plivo and make
plivo return a <say> action, which for some reason is not reading back
the text to me (no audio) when I call 1005 from using sflphone on my
ubuntu.

This is the log on freeswitch:

2014-07-29 18:01:12.581530 [NOTICE] switch_channel.c:1054 New Channel
sofia/internal/1000 at xxxxxxxxxx.amazonaws.com
[5650b3dd-1519-4c6f-b720-d4d3a8a65c5c]
2014-07-29 18:01:12.681534 [INFO] mod_dialplan_xml.c:558 Processing
1000 <1000>->1005 in context default
2014-07-29 18:01:12.681534 [WARNING] switch_core_session.c:1554
sofia/internal/1000 at xxxxxxxxxx.amazonaws.com using scheduler due to
bypass media or media is not established.
2014-07-29 18:01:12.733800 [NOTICE] sofia_media.c:92 Pre-Answer
sofia/internal/1000 at xxxxxxxxxx.amazonaws.com!
2014-07-29 18:01:12.733800 [NOTICE] mod_dptools.c:1258 Channel
[sofia/internal/1000 at xxxxxxxxxx.amazonaws.com] has been answered
2014-07-29 18:01:14.121516 [NOTICE] mod_dptools.c:1232 Hangup
sofia/internal/1000 at xxxxxxxxxx.amazonaws.com [CS_RESET]
[NORMAL_CLEARING]
2014-07-29 18:01:14.141502 [NOTICE] switch_core_session.c:1632 Session
1 (sofia/internal/1000 at xxxxxxxxxx.amazonaws.com) Ended
2014-07-29 18:01:14.141502 [NOTICE] switch_core_session.c:1636 Close
Channel sofia/internal/1000 at xxxxxxxxxx.amazonaws.com [CS_DESTROY]

I can see that plivo is correctly calling the XML I have placed, as
this is the log I see on my web server:

[29/Jul/2014 01:59:13] "GET
/answer/?To=1005&Direction=inbound&From=1000&CallerName=1000&CallUUID=b74c06be-f4e9-4830-af68-e424b3b31e1c&CallStatus=ringing
HTTP/1.1" 200 93
[29/Jul/2014 01:59:14] "GET
/answer/?To=1005&Direction=inbound&From=1000&CallUUID=b74c06be-f4e9-4830-af68-e424b3b31e1c&HangupCause=NORMAL_CLEARING&CallStatus=completed
HTTP/1.1" 200 93

This is the XML I have prepared to test plivo:

<?xml version="1.0" encoding="UTF-8" ?>
<Response>
  <Speak>Hello World</Speak>
</Response>


And this is default.xml on freeswitch which forward any calls to 1005 to plivo:

<include>
  <context name="default">

    <!--
     This extension allows calling any digits of number
     freeswitch will call plivo outbound server on every incoming call
    -->
    <extension name="plivo">
        <condition field="destination_number" expression="^1005">
            <action application="enable_heartbeat" data="60"/>
            <action application="socket" data="127.0.0.1:8084 async full"/>
        </condition>
    </extension>
....
....
</include>

I have read that NAT issues might be causing a no-audio issue, but I
am not sure and do not know how to tell if this is the issue. The only
thing which look strange is the logs on freeswitch saying:

2014-07-29 18:01:12.681534 [WARNING] switch_core_session.c:1554
sofia/internal/1000 at xxxxxxxxxx.amazonaws.com using scheduler due to
bypass media or media is not established.

but what does that mean?



Join us at ClueCon 2016 Aug 8-12, 2016
More information about the FreeSWITCH-users mailing list