[Freeswitch-users] Calling internal extensions to/from outside is acting strange

Tae-Sung Shin prayersts at gmail.com
Mon Aug 9 19:59:28 PDT 2010


I found the resolution. According to FS log, problem was FS getting
"Transcoding_necessary" error and hangup whenever outbound call was
answered. My solution was simply to upgrade FS. The problematic version was
prebuild Windows version 1.0.4 but getting and compiling FS 1.0.6 source was
worth the effort. Everything is working fine now. 

 

Hope this helps anybody having a similar problem.

 

Tae-Sung Shin

 

From: Tae-Sung Shin [mailto:prayersts at gmail.com] 
Sent: Monday, August 09, 2010 8:22 AM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Calling internal extensions to/from outside is
acting strange

 

Hello Guys

 

First of all, I am a new user of Freeswitch. I spent a couple of days on
this issue and am desperate for some help.

 

Briefly, my problems are

 

1.       Calling out from SPA2102 (ext 1002) or a softphone (phonerlite)
(ext 1003) via gateway voip is disconnected as soon as it got answered after
ringing in the other side. 

2.       Calling from outside is disconnected 30 seconds after it got
answered. 

 

I don't have this problem with calls between the internal extensions.

Without extensions (direct communication between gateway and SPA2102), I
verified SPA2102 is working fine

 

My environment: <PSTN outside> --- <Freeswitch> --- <my extensions>

 

As Freeswitch wiki, suggested, I have following xml contents

 

.         Sip_profiles/external/voipms.xml

 

<include>

  <gateway name="sip.atlanta.voip.ms">

    <param name="username" value="XXXXXXX" />

    <param name="password" value="XXXXXXXXX" />

    <param name="register" value="true" />

    <param name="expire-seconds" value="60"/>

    <param name="register-transport" value="udp"/>

    <param name="retry-seconds" value="100"/>

  </gateway>

</include>

 

.         Dialplan/public/voipms.xml

 

 

<include>

    <extension name="AllOutbound">

      <condition field="network_addr" expression="^192\.168\.0\.11[2-7]$"/> 

      <condition field="destination_number" expression="^1?(\d{10})$">

        <action application="set"
data="effective_caller_id_number=XXXXXXX"/>

        <action application="set" data="effective_caller_id_name=XXXXXXX"/>

        <action application="bridge"
data="sofia/gateway/sip.atlanta.voip.ms/$1"/>

      </condition>

    </extension>

<extension name="Inbound_mss">

  <condition field="destination_number" expression="^5555555555$">

    <action application="bridge"
data="sofia/internal/1002 at 192.168.0.114:5060"/>

  </condition>

</extension>

<extension name="Inbound_pstn">

  <condition field="destination_number" expression="^1002$">

    <action application="bridge"
data="sofia/internal/1002 at 192.168.0.114:5060"/>

  </condition>

</extension>

<extension name="Inbound_sipphone">

  <condition field="destination_number" expression="^1003$">

    <action application="bridge"
data="sofia/internal/@1003 at 192.168.0.117:5060"/>

  </condition>

</extension>

 

</include>

 

I think I went through all wiki and other articles on the web but could find
a resolution for this issue. I would appreciate if you give me any hint.

 

Thanks

 

Tae-Sung Shin

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