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

José Lopes jose.lopes at itcenter.com.pt
Fri Jan 6 14:08:01 MSK 2017


Hello Anthony,

Thanks for your reply.

I tried to use an audio file (sounds/en/us/callie/ivr/8000/ivr-say_name.wav
with ~2 seconds) instead of silence_stream.
When i make the call from verto client, i ear the audio file, then no audio
for ~2/3 seconds and then i ear "id followed by pound" (audio cut off from
voicemail initial message "Please enter your id followed by pound").

I checked if i have the variable answer_delay and i don't have it.

The log of this call is at https://pastebin.freeswitch.org/view/e130e172 .

There is any thing more that i can do?


Best Regards,
Jose Lopes

2017-01-05 18:14 GMT+00:00 Anthony Minessale <anthony.minessale at gmail.com>:

> 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
>>>
>>> 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*
>>>> http://www.freeswitchbook.com
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>>>>
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>>>>
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>>>> <(918)%20420-9002> | *M:*+1918424WEST (9378)
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>>>>
>>>> ____________________________________________________________
>>>> _____________
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>>>> http://www.freeswitchsolutions.com
>>>>
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>>>>
>>>
>>>
>>> ____________________________________________________________
>>> _____________
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>>>
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>>
>>
>>
>> --
>> Anthony Minessale II       ♬ @anthmfs  ♬ @FreeSWITCH  ♬
>>
>>http://freeswitch.org/http://cluecon.com/>> http://twitter.com/FreeSWITCH
>> ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+
>> <http://freeswitch.org/g+>*
>>
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>>
>> https://www.youtube.com/watch?v=9XXgW34t40s
>> https://www.youtube.com/watch?v=NLaDpGQuZDA
>>
>
>
>
> --
> Anthony Minessale II       ♬ @anthmfs  ♬ @FreeSWITCH  ♬
>
>http://freeswitch.org/http://cluecon.com/> http://twitter.com/FreeSWITCH
> ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+
> <http://freeswitch.org/g+>*
>
> ClueCon Weekly Development Call
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>
> https://www.youtube.com/watch?v=9XXgW34t40s
> https://www.youtube.com/watch?v=NLaDpGQuZDA
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
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> http://confluence.freeswitch.org
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>
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