[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 21:14:30 MSK 2017 

Also make sure you don't have answer_delay set in your vars.xml


On Thu, Jan 5, 2017 at 12:13 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> 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
>>
>> <http://www.itcenter.com.pt/> <http://www.itcenter.com.pt/>
>> <http://www.itcenter.com.pt/>
>> www.itcenter.com.pt [image: ITCENTER Store]
>> <http://store.itcenter.com.pt/> [image: ITCENTER Helpdesk]
>> <http://support.itcenter.com.pt/otrs/customer.pl>
>> [image: ITCENTER Facebook] <https://www.facebook.com/itcenterpt> [image:
>> ITCENTER Linkedin] <https://www.linkedin.com/company/itcenter> [image:
>> ITCENTER Twitter] <https://twitter.com/ITCenter_PT>
>>
>>
>> 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
>>>>
>>>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.com
>>>>
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>>>>
>>>>
>>>> ____________________________________________________________
>>>> _____________
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>>>>
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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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>>>
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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
>>
>> FreeSWITCH-users mailing list
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>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II       ♬ @anthmfs  ♬ @FreeSWITCH  ♬
>
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>
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> https://www.youtube.com/watch?v=NLaDpGQuZDA
>



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

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