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

José Lopes jose.lopes at itcenter.com.pt
Tue Jan 10 21:27:48 MSK 2017 

Hello Anthony

Thanks for your reply. I am sorry, I didn't notice that it was a diagnose
test.
I am available to make the tests that you need to analyse this situation
and i put bellow information about the test you ask.
If you see it is better to create a issue on JIRA, i will do it.
Thanks for your and FreeSwitch Team effort.


If I add a sleep of 2000 at voicemail server, after the answer there is no
audio cut off (I put bellow the change on dialplan).
But there is silence of 3/4 seconds between the ivr-say_name and the
initial message from voicemail without audio cut off.

Let me know if you need more tests or information.


Dialplan on FreeSwitch Test:
<include>
  <context name="default">
    <extension name="itsp_send_call">
      <condition field="destination_number" expression="^.*$">
        <action application="answer"/>
        <action application="playback"
data="/usr/share/freeswitch/sounds/en/us/callie/ivr/8000/ivr-say_name.wav"/>
        <action application="set" data="ringback=$${us-ring}"/>
        <action application="bridge"
data="{absolute_codec_string='$${sip_codec_prefs}'}sofia/gateway/${caller_id_number}/${destination_number}"/>
      </condition>
    </extension>
  </context>
</include>

Extract of Dialplan on FreeSwitch External Voicemail Server

    <!-- voicemail main extension -->
    <extension name="vmain">
      <condition field="destination_number"
expression="^vmain$|^4000$|^\*98$">
        <action application="answer"/>
        <action application="sleep" data="1000"/>
        <action application="sleep" data="2000"/> <!-- line added -->
        <action application="voicemail" data="check default
${domain_name}"/>
      </condition>
    </extension>


Best Regards,
Jose Lopes




2017-01-10 17:22 GMT+00:00 Anthony Minessale <anthony.minessale at gmail.com>:

> The minute you call it an Issue you should be filing it on JIRA.
> We get countless emails a day to the list so I don't always read them all
> so you are lucky I have managed to follow this thread.
>
> https://freeswitch.org/jira
>
> We have a small team and dealing with the mailing list is a volunteer
> effort.
>
> Here is also a tip.  Just provide the info to questions asked.  I asked
> you to do a diagnostic test for me by adding sleep to the other FS.
> Regardless if you can change the production or not, its still relevant to
> me what happens when you change it.
>
>
>
>
> On Tue, Jan 10, 2017 at 3:58 AM, José Lopes <jose.lopes at itcenter.com.pt>
> wrote:
>
>> Hello Anthony,
>>
>> At this replicated scenario, the box I am calling on SIP is FS.
>> But on real scenario, the box I am calling on SIP is not Freeswitch, it
>> is an external voicemail server and the initial message have audio cut off.
>>
>> Thanks for the information about variable ringback, I am already using on
>> real scenario.
>>
>> One strange thing is if I use the codec OPUS at verto, this issue doesn't
>> happen.
>> But I need to use codec PCMU to avoid audio transcoding.
>>
>> Let me know if you need more information to debug this issue.
>>
>> Best Regards,
>> Jose Lopes
>>
>>
>>
>>
>> 2017-01-09 18:25 GMT+00:00 Anthony Minessale <anthony.minessale at gmail.com
>> >:
>>
>>> So that concludes that media is already established on the webrtc end
>>> and there is no problem with that.
>>> The box you are calling on SIP is also FS, you may want to add a sleep
>>> 2000 in that dialplan before the voicemail.
>>> Also since webrtc has no ringing indication you may want to set the
>>> variable ringback to get some audible feedback when making calls.
>>>
>>>
>>> On Fri, Jan 6, 2017 at 5:08 AM, José Lopes <jose.lopes at itcenter.com.pt>
>>> wrote:
>>>
>>>> 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
>>>>>>>>>
>>>>>>>>> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> ____________________________________________________________
>>>>>>>>> _____________
>>>>>>>>> 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
>>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free
>>>>>>>>> switch-users
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> --
>>>>>>>>
>>>>>>>> *Brian West*
>>>>>>>> brian at freeswitch.org
>>>>>>>>
>>>>>>>>
>>>>>>>> *Twitter: @FreeSWITCH , @briankwest*
>>>>>>>> http://www.freeswitchbook.com
>>>>>>>> http://www.freeswitchcookbook.com
>>>>>>>> https://www.gofundme.com/freeswitch_ubuntu
>>>>>>>>
>>>>>>>> Got Bugs? Report them here <https://freeswitch.org/jira>! |
>>>>>>>> Reddit: /r/freeswitch <https://www.reddit.com/r/freeswitch>
>>>>>>>>
>>>>>>>> *T:*+19184209001 <(918)%20420-9001> | *F:*+19184209002
>>>>>>>> <(918)%20420-9002> | *M:*+1918424WEST (9378)
>>>>>>>> *Skype:*briankwest
>>>>>>>>
>>>>>>>> ____________________________________________________________
>>>>>>>> _____________
>>>>>>>> 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
>>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free
>>>>>>>> switch-users
>>>>>>>> http://www.freeswitch.org
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> ____________________________________________________________
>>>>>>> _____________
>>>>>>> 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
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free
>>>>>>> switch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> 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
>>>>>> ☎ sip:888 at conference.freeswitch.org  ☎ +19193869900
>>>>>> <(919)%20386-9900>
>>>>>>
>>>>>> 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
>>>>> ☎ sip:888 at conference.freeswitch.org  ☎ +19193869900 <(919)%20386-9900>
>>>>>
>>>>>
>>>>> 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
>>>>> http://www.freeswitch.org
>>>>> http://confluence.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free
>>>>> switch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>> ____________________________________________________________
>>>> _____________
>>>> 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
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free
>>>> switch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>> 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
>>> ☎ sip:888 at conference.freeswitch.org  ☎ +19193869900 <(919)%20386-9900>
>>>
>>> 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
>>> http://www.freeswitch.org
>>> http://confluence.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _________________________________________________________________________
>> 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
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> 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
> ☎ sip:888 at conference.freeswitch.org  ☎ +19193869900 <(919)%20386-9900>
>
> 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
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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