[Freeswitch-dev] No sound with Gizmo5 after transfer from a dial plan

[Michael Jerris]

That means they answered the call already.  Is that accurate?

On Jan 23, 2010, at 2:10 AM, Paul Li wrote:

> By comparing the log from a working service provider (Working) to that
> from Gizmo5, I have found out the problem might be Gizmo5 (service
> provider). While FreeSWITCH was performing TRANSFER,
> 
> Gizmo5: Channel sofia/softphone/001x...x entering state [completing][200]
> Working: Channel sofia/softphone/001x...x entering state [proceeding][180]
> 
> Gizmo5: Channel sofia/softphone/001x...x entering state [ready][200]
> Working: Channel sofia/softphone/001x...x entering state [proceeding][180]
> 
> Gizmo5: [Channel sofia/softphone/001x...x] has been answered
> Working: Pre-Answer sofia/softphone/001x...x
> 
> Both began "Running State Change CS_ROUTING". After routing was
> successful, the working service provider finally enter [200] state.
> 
> Working: Channel sofia/softphone/001x...x entering state [completing][200]
> Working: Channel sofia/softphone/001x...x entering state [ready][200]
> 
> At this point, the working service provider finally connects
> sofia/softphone/001x...x to the playback channel. I am wondering what
> response Gizmo5 sent to FreeSWITCH  and led FreeSWITCH  to think the
> call prematurally entered state [completing][200] and [ready][200].
> 
> 
> On Fri, Jan 22, 2010 at 12:30 AM, Paul Li <plite2012 at gmail.com> wrote:
>> It is a very interesting and challenging issue! What I am trying to
>> achieve is playback a message (notice) to PSTN telephones with Gizmo5
>> as service provider with the originate command as follows:
>> 
>> $ originate sofia/gateway/gizmo/0019876543210 5001     (Case 1)
>> 
>> $ originate sofia/gateway/gizmo/0019876543210 5002     (Case 2)
>> 
>> I can hear the playback from extension 5001 in the first case, while I
>> could not hear anything from extension 5002 in the second case. The
>> difference between them lies in that extension 5002 transfer the call
>> when the PSTN phone is connected. Please see the dial plan below. Both
>> dial plans are basically adapted from examples found in the online
>> documents. Both 5001 and 5002 work for my two other voip service
>> providers. On the surface, it appears to be Gizmo5's problem. But, I
>> am wondering if there is any issue on the FreeSwitch side.
>> 
>> My main goal is to delay playback until the end user has picked up the
>> phone. I have been looking for a way to do this but ended up with
>> nothing so far. I have tried setting {ignore_early_media=true}, but
>> that did not help either.
>> 
>> I am welcome any suggestion on how to achieve this. In the meanwhile,
>> I would like point out this issue.
>> 
>>        <!-- a simple IVR demo with direct playback -->
>>        <extension name="direct_ivr_demo1">
>>                <condition field="destination_number" expression="^5001$">
>>                        <action application="start_dtmf" />
>>                        <action application="playback" data="test.wav"/>
>>                </condition>
>>        </extension>
>> 
>>        <!-- a simple IVR demo with indirect playback -->
>>        <!-- use execute_on_answer to delay playing back until the bridge has
>> been established -->
>>        <extension name="indirect_ivr_demo2">
>>                <condition field="destination_number" expression="^5002$">
>>                        <action application="set" data="execute_on_answer=transfer
>> ANSWERCALL XML default"/>
>>                        <action application="sleep" data="60000"/>
>>                        <action application="transfer" data="NOANS XML default"/>
>>                </condition>
>>        </extension>
>> 
>>        <!-- execute_on_answer: Call answered at far end -->
>>        <extension name="Call_Answered">
>>                <condition field="destination_number" expression="^ANSWERCALL$">
>>                        <action application="start_dtmf" />
>>                        <action application="playback" data="test.wav"/>
>>                </condition>
>>        </extension>
>> 
>>        <!-- execute_on_answer: No answer at far end -->
>>          <extension name="Call_Not_Answered">
>>                <condition field="destination_number" expression="^NOANS$">
>>                        <action application="hangup" data="NO_USER_RESPONSE"/>
>>                </condition>
>>        </extension>
>> 
> 
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