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

[Paul Li]

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>
>