[Freeswitch-users] Audio cut off at the begin of the verto call to sip external voicemail
Anthony Minessale
anthony.minessale at gmail.com
Tue Jan 10 21:59:25 MSK 2017
Yes, please file a JIRA because that was my main concern. Its very easy
for me to forget about this thread or not notice it was updated.
Once it is in JIRA it serves a place holder for all the info.
On Tue, Jan 10, 2017 at 12:27 PM, José Lopes <jose.lopes at itcenter.com.pt>
wrote:
> 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/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/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
>>
>
>
> _________________________________________________________________________
> 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
https://www.youtube.com/watch?v=9XXgW34t40s
https://www.youtube.com/watch?v=NLaDpGQuZDA
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