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

Anthony Minessale anthony.minessale at gmail.com
Thu Jan 5 21:13:47 MSK 2017


Try making the call with

fsctl debug_level 10

and observe the logs, answer followed by silence_stream should send audio
to the client.
Also try playing an audio file instead of silence stream to see if you hear
it.





On Thu, Jan 5, 2017 at 11:58 AM, José Lopes <jose.lopes at itcenter.com.pt>
wrote:

> 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
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services: consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>>>
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>>
>>
>>
>> --
>>
>> *Brian West*
>> brian at freeswitch.org
>>
>>
>> *Twitter: @FreeSWITCH , @briankwest*
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>>
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>>
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>>
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>>
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>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
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>



-- 
Anthony Minessale II       ♬ @anthmfs  ♬ @FreeSWITCH  ♬

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