[Freeswitch-users] Audio cut off at the begin of the verto call to sip external voicemail

[José Lopes]

Hello Brian,

Thanks for your reply.

I tried the dialplan bellow with silence_stream://2000, and i have that
issue.
I tried with silence_stream://3000 and the audio cut off is greater.
Without the playback, there is no audio cut off, but FreeSwitch doesn't
send any rtp packets to verto client before the bridge.

There is any thing more that i can do?


<include>
  <context name="default">
    <extension name="call_debug" continue="true">
      <condition field="${call_debug}" expression="^true$" break="never">
        <action application="info"/>
      </condition>
    </extension>
    <extension name="itsp_send_call">
      <condition field="destination_number" expression="^.*$">
        <action application="answer"/>
        <action application="playback" data="silence_stream://2000"/>
        <action application="bridge"
data="{absolute_codec_string='PCMU'}sofia/gateway/1002/${destination_number}"/>
      </condition>
    </extension>
  </context>
</include>


Best Regards,
Jose Lopes


Os melhores cumprimentos / Best regards,

José Lopes
Research and Development
Phone: +351 256 370 980
Email: jose.lopes at itcenter.com.pt

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2017-01-05 15:47 GMT+00:00 Brian West <brian at freeswitch.org>:

> Prefix them with silence_stream://2000 or 3000 and it should go away.
>
> /b
>
>
> On Thu, Jan 5, 2017 at 9:29 AM, Bipin Patel <bipin at xbipin.com> wrote:
>
>> hi,
>>
>> i have the same issue, i think its related to slow audio setup during the
>> call
>>
>>
>> Regards,
>> Bipin
>>
>>
>> ------------------------------
>> -------- Original Message --------
>> Subject: [Freeswitch-users] Audio cut off at the begin of the verto call
>> to sip external voicemail
>> From: José Lopes <jose.lopes at itcenter.com.pt>
>> <jose.lopes at itcenter.com.pt>
>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> <freeswitch-users at lists.freeswitch.org>
>> Date: 1/5/2017, 6:35:45 PM
>>
>> Hello Guys,
>>
>> I have audio cut off at the begin of the verto call to  FreeSwitch that
>> redirect to sip external voicemail (Access voicemail mailbox) .
>>
>> This happen when I use PCMU at verto codecs and sip codecs (if i use opus
>> at verto codecs, there is no issue, but this causes audio transcoding) .
>>
>> At dialplan i used the example "Bridging from WebRTC (mod_verto) to
>> PSTN/ITSPs" from https://freeswitch.org/conflue
>> nce/display/FREESWITCH/mod_verto.
>> I notice if i remove the playback action, there is no issue. But I need
>> the playback action to send rtp packets to verto client.
>>
>> I simulate this using another FreeSwitch as external voicemail server and
>> I only listen "id followed by pound" from the initial message of voicemail
>> ("Please enter your id followed by pound").
>> The log of this call is at https://pastebin.freeswitch.org/view/507fa115
>>
>> What I can do to use PCMU at verto codecs and sip codecs on type of call?
>> Should i open a issue on FreeSwitch JIRA ?
>>
>>
>> Best regards,
>> Jose Lopes
>>
>>
>> _________________________________________________________________________
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>>
>>
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>>
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>
>
>
> --
>
> *Brian West*
> brian at freeswitch.org
>
>
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