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

José Lopes jose.lopes at itcenter.com.pt
Thu Jan 12 13:02:56 MSK 2017


I created the issue at JIRA: https://freeswitch.org/jira/browse/FS-9939 .
I tried to put all the information discussed here.

Thanks again for your help.


2017-01-10 18:59 GMT+00:00 Anthony Minessale <anthony.minessale at gmail.com>:

> 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/so
>> unds/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/${caller_id_number}/${des
>> tination_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.freesw
>>>>>>>>>>> itch.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/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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