Anthony Minessale | 1 Jul 2011 01:00
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Re: NAT Traversal on SFLphone - FS not Auto Changing port

you might want to include the sip-trace and the whole call setup .

Your log shows its getting audio from the phone and autodetects the correct rtp port.
Do you have both on at once? maybe one already owns your soundcard so it breaks the other, try turning them both off and only start the one.
while the call is up, get a pcap on both sides and see if audio is going to the right place.


On Thu, Jun 30, 2011 at 1:23 AM, Avi Marcus <avi-FU8ycyX5tNssV2N9l4h3zg@public.gmane.org> wrote:
http://pastebin.freeswitch.org/16627
I've got a Linksys ATA behind NAT and a softphone behind NAT. Both seem to register with the same UDP-NAT string and both have the same type of contact string.. but on one FS is rewriting the RTP IP to work properly - the 1102 works, but on 1000 I don't seem to be getting any audio.

Is there some hidden parameter I can't see affecting this? Both are dialing the same extension 9664 default MOH stuff...
This softphone actually seems to have a responsive gui in linux. If I can get the audio to actually work, that would great!
(Oh, it has tls/srtp too, it seems)

Thanks!
-Avi Marcus



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Anthony Minessale II

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<div>
<p>you might want to include the&nbsp;sip-trace&nbsp;and the whole call setup .</p>
<div>Your log shows its getting audio from the phone and autodetects the correct rtp port.</div>
<div>Do you have both on at once? maybe one already owns your soundcard so it breaks the other, try turning them both off and only start the one.</div>
<div>while the call is up, get a pcap on both sides and see if audio is going to the right place.</div>
<div>
<br><br><div class="gmail_quote">On Thu, Jun 30, 2011 at 1:23 AM, Avi Marcus <span dir="ltr">&lt;<a href="mailto:avi@...">avi@...</a>&gt;</span> wrote:<br><blockquote class="gmail_quote">
<div dir="ltr">
<a href="http://pastebin.freeswitch.org/16627" target="_blank">http://pastebin.freeswitch.org/16627</a><div>
I've got a Linksys ATA behind NAT and a softphone behind NAT. Both seem to register with the same UDP-NAT string and both have the same type of contact string.. but on one FS is rewriting the RTP IP to work properly - the 1102 works, but on 1000 I don't seem to be getting any audio.</div>

<div><br></div>
<div>Is there some hidden parameter I can't see affecting this? Both are dialing the same extension 9664 default MOH stuff...</div>
<div>This softphone actually seems to have a responsive gui in linux. If I can get the audio to actually work, that would great!</div>

<div>(Oh, it has tls/srtp too, it seems)</div>
<div><br></div>
<div>Thanks!</div>
<div>
<a href="http://pastebin.freeswitch.org/16627" target="_blank"></a><div dir="ltr">
<span><span>-Avi Marcus</span><br><br></span><div>
<div></div>

</div>
</div>
<br>
</div>
</div>
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ran zhang | 1 Jul 2011 01:28
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how to send messages to all members in a conference via freeswitch?

Hi All:

            I'm using Linphone as my VOIP client, i want 1 member to 
send instant messages to the conference and all other members in the 
conference should receive them.

           I can't figure out a way to do this,  Linphone only allows to 
send messages to other members on the contact list, even if Linphone can 
do it, can freeswitch handle this?

ran zhang | 1 Jul 2011 03:29
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Re: please help!!! how to set flag 'endconf' in bridging conference

       I attached the section in my dialplan that handles the bridging conference, when the first user (just say user 99)  dials '20', he will invite user 10 to join conference, the conference will only be established if user 10 accepts the invitation, after this, other users can join the conference by dialing '20'.

       I want that if the first user (user 99 in this case)  or the user been invited (user 10 in this case)  leaves the conference, no matter how many people are still in the conference, it will close down the conference.   so i'm trying to use the 'endconf' flag, but apparently it is not valid syntax for bridging conference as I get a config error while running.  If I take out the '+flags{endconf} ', i wont get a config error while running, but then conference will only close down when there is 1 person left.

      I have also tried creating 2 difference conference profiles,  one profile has the 'endconf' set in the 'member-flags', one profiles doesnt have 'endconf' set, so user99 and user20 joins the 20 <at> profile1 conference, and other users joins the 20 <at> profile2 conference, that doesnt seem to work neither.   I have pasted conference.conf.xml file as well for ur review.
      
<include> <context name="test"> <extension name="global" continue="true"> <condition> </condition> </extension> <extension name="conf"> <condition field="destination_number" expression="^(20)$" > </condition> <action application="answer"/> <action application="conference" data="bridge:20 <at> default+flags{endconf}:user/10"/> <action application="conference" data="20 <at> default"/> </condition> </extension> </context> </include>



On 6/30/2011 6:16 PM, Michael Collins wrote:
Please put this information on pastebin and reply to the list so that we can all discuss it.
-MC

On Thu, Jun 30, 2011 at 1:51 PM, ran zhang <rzhang-JA7S4awhwUdeQIzoc+Smag@public.gmane.org> wrote:
Mr Collins:

       I attached the section in my dialplan that handles the bridging conference, when the first user (just say user 99)  dials '20', he will invite user 10 to join conference, the conference will only be established if user 10 accepts the invitation, after this, other users can join the conference by dialing '20'.

       I want that if the first user (user 99 in this case)  or the user been invited (user 10 in this case)  leaves the conference, no matter how many people are still in the conference, it will close down the conference.   so i'm trying to use the 'endconf' flag, but apparently it is not valid syntax for bridging conference as I get a config error while running.  If I take out the '+flags{endconf} ', i wont get a config error while running, but then conference will only close down when there is 1 person left.

      I have also tried creating 2 difference conference profiles,  one profile has the 'endconf' set in the 'member-flags', one profiles doesnt have 'endconf' set, so user99 and user20 joins the 20 <at> profile1 conference, and other users joins the 20 <at> profile2 conference, that doesnt seem to work neither.   I have pasted conference.conf.xml file as well for ur review.
      
<include> <context name="test"> <extension name="global" continue="true"> <condition> </condition> </extension> <extension name="conf"> <condition field="destination_number" expression="^(20)$" > </condition> <action application="answer"/> <action application="conference" data="bridge:20 <at> default+flags{endconf}:user/10"/> <action application="conference" data="20 <at> default"/> </condition> </extension> </context> </include>
<configuration name="conference.conf" description="Audio Conference"> <!-- Advertise certain presence on startup . --> <advertise> <room name="3001 <at> $${domain}" status="FreeSWITCH"/> </advertise> <!-- These are the default keys that map when you do not specify a caller control group --> <!-- Note: none and default are reserved names for group names. Disabled if dist-dtmf member flag is set. --> <caller-controls> <group name="default"> <control action="mute" digits="0"/> <control action="deaf mute" digits="*"/> <control action="energy up" digits="9"/> <control action="energy equ" digits="8"/> <control action="energy dn" digits="7"/> <control action="vol talk up" digits="3"/> <control action="vol talk zero" digits="2"/> <control action="vol talk dn" digits="1"/> <control action="vol listen up" digits="6"/> <control action="vol listen zero" digits="5"/> <control action="vol listen dn" digits="4"/> <control action="hangup" digits="#"/> </group> </caller-controls> <!-- Profiles are collections of settings you can reference by name. --> <profiles> <!--If no profile is specified it will default to "default"--> <profile name="default"> <!-- Domain (for presence) --> <param name="domain" value="$${domain}"/> <!-- Sample Rate--> <param name="rate" value="8000"/> <!-- Number of milliseconds per frame --> <param name="interval" value="20"/> <!-- Energy level required for audio to be sent to the other users --> <param name="energy-level" value="300"/> <!--Can be | delim of waste|mute|deaf|dist-dtmf waste will always transmit data to each channel even during silence. dist-dtmf propagates dtmfs to all other members, but channel controls via dtmf will be disabled. --> <param name="member-flags" value="dist-dtmf"/> <!-- Name of the caller control group to use for this profile --> <!-- <param name="caller-controls" value="some name"/> --> <!-- TTS Engine to use --> <!--<param name="tts-engine" value="cepstral"/>--> <!-- TTS Voice to use --> <!--<param name="tts-voice" value="david"/>--> <!-- If TTS is enabled all audio-file params beginning with --> <!-- 'say:' will be considered text to say with TTS --> <!-- Override the default path here, after which you use relative paths in the other sound params --> <!-- Note: The default path is the conference's first caller's sound_prefix --> <!--<param name="sound-prefix" value="$${sounds_dir}/en/us/callie"/>--> <!-- File to play to acknowledge succees --> <!--<param name="ack-sound" value="beep.wav"/>--> <!-- File to play to acknowledge failure --> <!--<param name="nack-sound" value="beeperr.wav"/>--> <!-- File to play to acknowledge muted --> <!-- Conference pin --> <!--<param name="pin" value="12345"/>--> <!-- Default Caller ID Name for outbound calls --> <param name="caller-id-name" value="$${outbound_caller_name}"/> <!-- Default Caller ID Number for outbound calls --> <param name="caller-id-number" value="1234"/> <!-- Suppress start and stop talking events --> <!-- <param name="suppress-events" value="start-talking,stop-talking"/> --> <!-- enable comfort noise generation --> <param name="comfort-noise" value="true"/> <!-- Uncomment auto-record to toggle recording every conference call. --> <!-- Another valid value is shout://user:pass-WQCBuIXaXYjQT0dZR+AlfA@public.gmane.org/live.mp3 --> <!-- <param name="auto-record" value="$${recordings_dir}/${conference_name}_${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> --> </profile>



















On 6/30/2011 1:07 PM, Michael Collins wrote:
Can you pastebin exactly what you are doing to establish the call? Including any relevant dialplan entries. Also, if you have modified conference.conf.xml we would like to see that also.

-MC

On Thu, Jun 30, 2011 at 10:59 AM, ran zhang <rzhang-JA7S4awhwUdeQIzoc+Smag@public.gmane.org> wrote:
hi all:

I'm trying to creating a conference, so when first member enters the
conference, he has to invite another members
and have at least 1 other member to join to have the conference established, so i'm
using bridging conference.

I need this conference to be terminated when the original creator of the
conference leaves no matter how many members are still left in the conference.

i'm trying to set 'endconf' flag in a bridging conference using
'bridge:confname+flag{endconf}:user/10',
so it wil invite user extension 10, but its giving me config error while
running.

can someone tell me what to do to solve this problem or get around?  the key
is i only want the original member to be able to terminate the conference
when he leaves, not other members assuming there are at least 2 members.


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<div>
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; I attached the section in my dialplan that handles the
    bridging conference, when the first user (just say user 99)&nbsp; dials
    '20', he will invite user 10 to join conference, the conference will
    only be established if user 10 accepts the invitation, after this,
    other users can join the conference by dialing '20'. <br><br>
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; I want that if the first user (user 99 in this case)&nbsp; or the
    user been invited (user 10 in this case)&nbsp; leaves the conference, no
    matter how many people are still in the conference, it will close
    down the conference.&nbsp;&nbsp; so i'm trying to use the 'endconf' flag, but
    apparently it is not valid syntax for bridging conference as I get a
    config error while running.&nbsp; If I take out the '+flags{endconf} ', i
    wont get a config error while running, but then conference will only
    close down when there is 1 person left.<br><br>
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; I have also tried creating 2 difference conference profiles,&nbsp;
    one profile has the 'endconf' set in the 'member-flags', one
    profiles doesnt have 'endconf' set, so user99 and user20 joins the
    20 <at> profile1 conference, and other users joins the 20 <at> profile2
    conference, that doesnt seem to work neither.&nbsp;&nbsp; I have pasted
    conference.conf.xml file as well for ur review. <br>
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <br>&lt;include&gt;
  &lt;context name="test"&gt;
    &lt;extension name="global" continue="true"&gt;
      &lt;condition&gt;
      &lt;/condition&gt;
    &lt;/extension&gt;

    &lt;extension name="conf"&gt;
      &lt;condition field="destination_number" expression="^(20)$" &gt; &lt;/condition&gt;
        &lt;action application="answer"/&gt;        
        &lt;action application="conference" data="bridge:20 <at> default+flags{endconf}:user/10"/&gt;
        &lt;action application="conference" data="20 <at> default"/&gt; 
      &lt;/condition&gt;
    &lt;/extension&gt;   
  &lt;/context&gt;
&lt;/include&gt;
    <br><br><br><br>
    On 6/30/2011 6:16 PM, Michael Collins wrote:
    <blockquote cite="mid:BANLkTi=Cmuz6a3K8smK3Nr46E2ptg=pZdQ@..." type="cite">Please put this information on pastebin and reply to
      the list so that we can all discuss it.<br>
      -MC<br><br><div class="gmail_quote">On Thu, Jun 30, 2011 at 1:51 PM, ran
        zhang <span dir="ltr">&lt;<a moz-do-not-send="true" href="mailto:rzhang@...">rzhang@...</a>&gt;</span>
        wrote:<br><blockquote class="gmail_quote">
          <div text="#000000" bgcolor="#ffffff"> Mr Collins:<br><br>
            &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; I attached the section in my dialplan that handles
            the bridging conference, when the first user (just say user
            99)&nbsp; dials '20', he will invite user 10 to join conference,
            the conference will only be established if user 10 accepts
            the invitation, after this, other users can join the
            conference by dialing '20'. <br><br>
            &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; I want that if the first user (user 99 in this case)&nbsp;
            or the user been invited (user 10 in this case)&nbsp; leaves the
            conference, no matter how many people are still in the
            conference, it will close down the conference.&nbsp;&nbsp; so i'm
            trying to use the 'endconf' flag, but apparently it is not
            valid syntax for bridging conference as I get a config error
            while running.&nbsp; If I take out the '+flags{endconf} ', i wont
            get a config error while running, but then conference will
            only close down when there is 1 person left.<br><br>
            &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; I have also tried creating 2 difference conference
            profiles,&nbsp; one profile has the 'endconf' set in the
            'member-flags', one profiles doesnt have 'endconf' set, so
            user99 and user20 joins the 20 <at> profile1 conference, and
            other users joins the 20 <at> profile2 conference, that doesnt
            seem to work neither.&nbsp;&nbsp; I have pasted conference.conf.xml
            file as well for ur review. <br>
            &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <br>&lt;include&gt;
  &lt;context name="test"&gt;
    &lt;extension name="global" continue="true"&gt;
      &lt;condition&gt;
      &lt;/condition&gt;
    &lt;/extension&gt;

    &lt;extension name="conf"&gt;
      &lt;condition field="destination_number" expression="^(20)$" &gt; &lt;/condition&gt;
        &lt;action application="answer"/&gt;        
        &lt;action application="conference" data="bridge:20 <at> default+flags{endconf}:user/10"/&gt;
        &lt;action application="conference" data="20 <at> default"/&gt; 
      &lt;/condition&gt;
    &lt;/extension&gt;   
  &lt;/context&gt;
&lt;/include&gt;

            <br>&lt;configuration name="conference.conf" description="Audio Conference"&gt;
  &lt;!-- Advertise certain presence on startup . --&gt;
  &lt;advertise&gt;
    &lt;room name="3001 <at> $${domain}" status="FreeSWITCH"/&gt;
  &lt;/advertise&gt;

  &lt;!-- These are the default keys that map when you do not specify a caller control group --&gt;	
  &lt;!-- Note: none and default are reserved names for group names.  Disabled if dist-dtmf member flag is set. --&gt;	
  &lt;caller-controls&gt;
    &lt;group name="default"&gt;
      &lt;control action="mute" digits="0"/&gt;
      &lt;control action="deaf mute" digits="*"/&gt;
      &lt;control action="energy up" digits="9"/&gt;
      &lt;control action="energy equ" digits="8"/&gt;
      &lt;control action="energy dn" digits="7"/&gt;
      &lt;control action="vol talk up" digits="3"/&gt;
      &lt;control action="vol talk zero" digits="2"/&gt;
      &lt;control action="vol talk dn" digits="1"/&gt;
      &lt;control action="vol listen up" digits="6"/&gt;
      &lt;control action="vol listen zero" digits="5"/&gt;
      &lt;control action="vol listen dn" digits="4"/&gt;
      &lt;control action="hangup" digits="#"/&gt;
    &lt;/group&gt;
  &lt;/caller-controls&gt;

  &lt;!-- Profiles are collections of settings you can reference by name. --&gt;
  &lt;profiles&gt;
    &lt;!--If no profile is specified it will default to "default"--&gt;
    &lt;profile name="default"&gt;
      &lt;!-- Domain (for presence) --&gt;
      &lt;param name="domain" value="$${domain}"/&gt;
      &lt;!-- Sample Rate--&gt;
      &lt;param name="rate" value="8000"/&gt;
      &lt;!-- Number of milliseconds per frame --&gt;
      &lt;param name="interval" value="20"/&gt;
      &lt;!-- Energy level required for audio to be sent to the other users --&gt;
      &lt;param name="energy-level" value="300"/&gt;

      &lt;!--Can be | delim of waste|mute|deaf|dist-dtmf waste will always transmit data to each channel
          even during silence.  dist-dtmf propagates dtmfs to all other members, but channel controls
	  via dtmf will be disabled. --&gt;
      &lt;param name="member-flags" value="dist-dtmf"/&gt;

      &lt;!-- Name of the caller control group to use for this profile --&gt;
      &lt;!-- &lt;param name="caller-controls" value="some name"/&gt; --&gt;
      &lt;!-- TTS Engine to use --&gt;
      &lt;!--&lt;param name="tts-engine" value="cepstral"/&gt;--&gt;
      &lt;!-- TTS Voice to use --&gt;
      &lt;!--&lt;param name="tts-voice" value="david"/&gt;--&gt;

      &lt;!-- If TTS is enabled all audio-file params beginning with --&gt;
      &lt;!-- 'say:' will be considered text to say with TTS --&gt;
      &lt;!-- Override the default path here, after which you use relative paths in the other sound params --&gt;
      &lt;!-- Note: The default path is the conference's first caller's sound_prefix --&gt;
      &lt;!--&lt;param name="sound-prefix" value="$${sounds_dir}/en/us/callie"/&gt;--&gt;
      &lt;!-- File to play to acknowledge succees --&gt;
      &lt;!--&lt;param name="ack-sound" value="beep.wav"/&gt;--&gt;
      &lt;!-- File to play to acknowledge failure --&gt;
      &lt;!--&lt;param name="nack-sound" value="beeperr.wav"/&gt;--&gt;
      &lt;!-- File to play to acknowledge muted --&gt;
      &lt;!-- Conference pin --&gt;
      &lt;!--&lt;param name="pin" value="12345"/&gt;--&gt;
      &lt;!-- Default Caller ID Name for outbound calls --&gt;
      &lt;param name="caller-id-name" value="$${outbound_caller_name}"/&gt;
      &lt;!-- Default Caller ID Number for outbound calls --&gt;
      &lt;param name="caller-id-number" value="1234"/&gt;
      &lt;!-- Suppress start and stop talking events --&gt;
      &lt;!-- &lt;param name="suppress-events" value="start-talking,stop-talking"/&gt; --&gt;
      &lt;!-- enable comfort noise generation --&gt;
      &lt;param name="comfort-noise" value="true"/&gt;
      &lt;!-- Uncomment auto-record to toggle recording every conference call. --&gt;
      &lt;!-- Another valid value is   <a moz-do-not-send="true" href="mailto:shout://user:pass@.../live.mp3" target="_blank">shout://user:pass@.../live.mp3</a>   --&gt;
      &lt;!--
      &lt;param name="auto-record" value="$${recordings_dir}/${conference_name}_${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/&gt;
      --&gt;
    &lt;/profile&gt;

            <div>
              <div class="h5"> <br><br><br><br><br><br><br><br><br><br><br><br><br><br><br><br><br><br><br>
                On 6/30/2011 1:07 PM, Michael Collins wrote:
                <blockquote type="cite">Can you pastebin exactly what
                  you are doing to establish the call? Including any
                  relevant dialplan entries. Also, if you have modified
                  conference.conf.xml we would like to see that also.<br><br>
                  -MC<br><br><div class="gmail_quote"> On Thu, Jun 30, 2011 at
                    10:59 AM, ran zhang <span dir="ltr">&lt;<a moz-do-not-send="true" href="mailto:rzhang@..." target="_blank">rzhang@...</a>&gt;</span>
                    wrote:<br><blockquote class="gmail_quote"> hi all:<br><br>
                      I'm trying to creating a conference, so when first
                      member enters the<br>
                      conference, he has to invite another members<br>
                      and have at least 1 other member to join to have
                      the conference established, so i'm<br>
                      using bridging conference.<br><br>
                      I need this conference to be terminated when the
                      original creator of the<br>
                      conference leaves no matter how many members are
                      still left in the conference.<br><br>
                      i'm trying to set 'endconf' flag in a bridging
                      conference using<br>
                      'bridge:confname+flag{endconf}:user/10',<br>
                      so it wil invite user extension 10, but its giving
                      me config error while<br>
                      running.<br><br>
                      can someone tell me what to do to solve this
                      problem or get around? &nbsp;the key<br>
                      is i only want the original member to be able to
                      terminate the conference<br>
                      when he leaves, not other members assuming there
                      are at least 2 members.<br><br><br>
                      _______________________________________________<br>
                      Join us at ClueCon 2011, Aug 9-11, Chicago<br><a moz-do-not-send="true" href="http://www.cluecon.com" target="_blank">http://www.cluecon.com</a>
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                  <br>
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                </blockquote>
                <br>
</div>
            </div>
          </div>
        </blockquote>
      </div>
      <br>
</blockquote>
    <br>
</div>
Nandy Dagondon | 1 Jul 2011 04:04
Picon

Re: please help!!! how to set flag 'endconf' in bridging conference

hi michael,

i'd like to have that feature, too, because our telco has an unusual line signalling - it only tear down the connection when the A-party hangs up first (Clear Forward).  if i transfer the callee to the conference, the telco line remains off-hook forever. as an interim solution, i'm using the terminate_on_silence parameter.

-nandy

On Fri, Jul 1, 2011 at 9:29 AM, ran zhang <rzhang-JA7S4awhwUdeQIzoc+Smag@public.gmane.org> wrote:
       I attached the section in my dialplan that handles the bridging conference, when the first user (just say user 99)  dials '20', he will invite user 10 to join conference, the conference will only be established if user 10 accepts the invitation, after this, other users can join the conference by dialing '20'.

       I want that if the first user (user 99 in this case)  or the user been invited (user 10 in this case)  leaves the conference, no matter how many people are still in the conference, it will close down the conference.   so i'm trying to use the 'endconf' flag, but apparently it is not valid syntax for bridging conference as I get a config error while running.  If I take out the '+flags{endconf} ', i wont get a config error while running, but then conference will only close down when there is 1 person left.

      I have also tried creating 2 difference conference profiles,  one profile has the 'endconf' set in the 'member-flags', one profiles doesnt have 'endconf' set, so user99 and user20 joins the 20 <at> profile1 conference, and other users joins the 20 <at> profile2 conference, that doesnt seem to work neither.   I have pasted conference.conf.xml file as well for ur review.
      
<include> <context name="test"> <extension name="global" continue="true"> <condition> </condition> </extension> <extension name="conf"> <condition field="destination_number" expression="^(20)$" > </condition> <action application="answer"/> <action application="conference" data="bridge:20 <at> default+flags{endconf}:user/10"/> <action application="conference" data="20 <at> default"/> </condition> </extension> </context> </include>



On 6/30/2011 6:16 PM, Michael Collins wrote:
Please put this information on pastebin and reply to the list so that we can all discuss it.
-MC

On Thu, Jun 30, 2011 at 1:51 PM, ran zhang <rzhang-JA7S4awhwUdeQIzoc+Smag@public.gmane.org> wrote:
Mr Collins:

       I attached the section in my dialplan that handles the bridging conference, when the first user (just say user 99)  dials '20', he will invite user 10 to join conference, the conference will only be established if user 10 accepts the invitation, after this, other users can join the conference by dialing '20'.

       I want that if the first user (user 99 in this case)  or the user been invited (user 10 in this case)  leaves the conference, no matter how many people are still in the conference, it will close down the conference.   so i'm trying to use the 'endconf' flag, but apparently it is not valid syntax for bridging conference as I get a config error while running.  If I take out the '+flags{endconf} ', i wont get a config error while running, but then conference will only close down when there is 1 person left.

      I have also tried creating 2 difference conference profiles,  one profile has the 'endconf' set in the 'member-flags', one profiles doesnt have 'endconf' set, so user99 and user20 joins the 20 <at> profile1 conference, and other users joins the 20 <at> profile2 conference, that doesnt seem to work neither.   I have pasted conference.conf.xml file as well for ur review.
      
<include> <context name="test"> <extension name="global" continue="true"> <condition> </condition> </extension> <extension name="conf"> <condition field="destination_number" expression="^(20)$" > </condition> <action application="answer"/> <action application="conference" data="bridge:20 <at> default+flags{endconf}:user/10"/> <action application="conference" data="20 <at> default"/> </condition> </extension> </context> </include>
<configuration name="conference.conf" description="Audio Conference"> <!-- Advertise certain presence on startup . --> <advertise> <room name="3001 <at> $${domain}" status="FreeSWITCH"/> </advertise> <!-- These are the default keys that map when you do not specify a caller control group --> <!-- Note: none and default are reserved names for group names. Disabled if dist-dtmf member flag is set. --> <caller-controls> <group name="default"> <control action="mute" digits="0"/> <control action="deaf mute" digits="*"/> <control action="energy up" digits="9"/> <control action="energy equ" digits="8"/> <control action="energy dn" digits="7"/> <control action="vol talk up" digits="3"/> <control action="vol talk zero" digits="2"/> <control action="vol talk dn" digits="1"/> <control action="vol listen up" digits="6"/> <control action="vol listen zero" digits="5"/> <control action="vol listen dn" digits="4"/> <control action="hangup" digits="#"/> </group> </caller-controls> <!-- Profiles are collections of settings you can reference by name. --> <profiles> <!--If no profile is specified it will default to "default"--> <profile name="default"> <!-- Domain (for presence) --> <param name="domain" value="$${domain}"/> <!-- Sample Rate--> <param name="rate" value="8000"/> <!-- Number of milliseconds per frame --> <param name="interval" value="20"/> <!-- Energy level required for audio to be sent to the other users --> <param name="energy-level" value="300"/> <!--Can be | delim of waste|mute|deaf|dist-dtmf waste will always transmit data to each channel even during silence. dist-dtmf propagates dtmfs to all other members, but channel controls via dtmf will be disabled. --> <param name="member-flags" value="dist-dtmf"/> <!-- Name of the caller control group to use for this profile --> <!-- <param name="caller-controls" value="some name"/> --> <!-- TTS Engine to use --> <!--<param name="tts-engine" value="cepstral"/>--> <!-- TTS Voice to use --> <!--<param name="tts-voice" value="david"/>--> <!-- If TTS is enabled all audio-file params beginning with --> <!-- 'say:' will be considered text to say with TTS --> <!-- Override the default path here, after which you use relative paths in the other sound params --> <!-- Note: The default path is the conference's first caller's sound_prefix --> <!--<param name="sound-prefix" value="$${sounds_dir}/en/us/callie"/>--> <!-- File to play to acknowledge succees --> <!--<param name="ack-sound" value="beep.wav"/>--> <!-- File to play to acknowledge failure --> <!--<param name="nack-sound" value="beeperr.wav"/>--> <!-- File to play to acknowledge muted --> <!-- Conference pin --> <!--<param name="pin" value="12345"/>--> <!-- Default Caller ID Name for outbound calls --> <param name="caller-id-name" value="$${outbound_caller_name}"/> <!-- Default Caller ID Number for outbound calls --> <param name="caller-id-number" value="1234"/> <!-- Suppress start and stop talking events --> <!-- <param name="suppress-events" value="start-talking,stop-talking"/> --> <!-- enable comfort noise generation --> <param name="comfort-noise" value="true"/> <!-- Uncomment auto-record to toggle recording every conference call. --> <!-- Another valid value is shout://user:pass-WQCBuIXaXYjQT0dZR+AlfA@public.gmane.org/live.mp3 --> <!-- <param name="auto-record" value="$${recordings_dir}/${conference_name}_${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> --> </profile>



















On 6/30/2011 1:07 PM, Michael Collins wrote:
Can you pastebin exactly what you are doing to establish the call? Including any relevant dialplan entries. Also, if you have modified conference.conf.xml we would like to see that also.

-MC

On Thu, Jun 30, 2011 at 10:59 AM, ran zhang <rzhang-JA7S4awhwUdeQIzoc+Smag@public.gmane.org> wrote:
hi all:

I'm trying to creating a conference, so when first member enters the
conference, he has to invite another members
and have at least 1 other member to join to have the conference established, so i'm
using bridging conference.

I need this conference to be terminated when the original creator of the
conference leaves no matter how many members are still left in the conference.

i'm trying to set 'endconf' flag in a bridging conference using
'bridge:confname+flag{endconf}:user/10',
so it wil invite user extension 10, but its giving me config error while
running.

can someone tell me what to do to solve this problem or get around?  the key
is i only want the original member to be able to terminate the conference
when he leaves, not other members assuming there are at least 2 members.


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<div>
<div>hi michael,<br>
</div>
<div><br></div>
<div>i'd like to have that feature, too, because our telco has an unusual line signalling - it only tear down the connection when the A-party hangs up first (Clear Forward). &nbsp;if i transfer the callee to the conference, the telco line remains off-hook forever. as an interim solution, i'm using the terminate_on_silence parameter.</div>

<br><div>-nandy<br>
</div>
<div><br></div>
<div class="gmail_quote">On Fri, Jul 1, 2011 at 9:29 AM, ran zhang <span dir="ltr">&lt;<a href="mailto:rzhang@...">rzhang@...</a>&gt;</span> wrote:<br><blockquote class="gmail_quote">

  

    

  <div text="#000000" bgcolor="#ffffff">
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; I attached the section in my dialplan that handles the
    bridging conference, when the first user (just say user 99)&nbsp; dials
    '20', he will invite user 10 to join conference, the conference will
    only be established if user 10 accepts the invitation, after this,
    other users can join the conference by dialing '20'. <br><br>
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; I want that if the first user (user 99 in this case)&nbsp; or the
    user been invited (user 10 in this case)&nbsp; leaves the conference, no
    matter how many people are still in the conference, it will close
    down the conference.&nbsp;&nbsp; so i'm trying to use the 'endconf' flag, but
    apparently it is not valid syntax for bridging conference as I get a
    config error while running.&nbsp; If I take out the '+flags{endconf} ', i
    wont get a config error while running, but then conference will only
    close down when there is 1 person left.<br><br>
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; I have also tried creating 2 difference conference profiles,&nbsp;
    one profile has the 'endconf' set in the 'member-flags', one
    profiles doesnt have 'endconf' set, so user99 and user20 joins the
    20 <at> profile1 conference, and other users joins the 20 <at> profile2
    conference, that doesnt seem to work neither.&nbsp;&nbsp; I have pasted
    conference.conf.xml file as well for ur review. <br>
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <br>&lt;include&gt;
  &lt;context name="test"&gt;
    &lt;extension name="global" continue="true"&gt;
      &lt;condition&gt;
      &lt;/condition&gt;
    &lt;/extension&gt;

    &lt;extension name="conf"&gt;
      &lt;condition field="destination_number" expression="^(20)$" &gt; &lt;/condition&gt;
        &lt;action application="answer"/&gt;        
        &lt;action application="conference" data="bridge:20 <at> default+flags{endconf}:user/10"/&gt;
        &lt;action application="conference" data="20 <at> default"/&gt; 
      &lt;/condition&gt;
    &lt;/extension&gt;   
  &lt;/context&gt;
&lt;/include&gt;    <br><br><br><br>
    On 6/30/2011 6:16 PM, Michael Collins wrote:
    <blockquote type="cite">Please put this information on pastebin and reply to
      the list so that we can all discuss it.<br>
      -MC<br><br><div class="gmail_quote">On Thu, Jun 30, 2011 at 1:51 PM, ran
        zhang <span dir="ltr">&lt;<a href="mailto:rzhang <at> gosilverplus.com" target="_blank">rzhang@...</a>&gt;</span>
        wrote:<br><blockquote class="gmail_quote">
          <div text="#000000" bgcolor="#ffffff"> Mr Collins:<br><br>
            &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; I attached the section in my dialplan that handles
            the bridging conference, when the first user (just say user
            99)&nbsp; dials '20', he will invite user 10 to join conference,
            the conference will only be established if user 10 accepts
            the invitation, after this, other users can join the
            conference by dialing '20'. <br><br>
            &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; I want that if the first user (user 99 in this case)&nbsp;
            or the user been invited (user 10 in this case)&nbsp; leaves the
            conference, no matter how many people are still in the
            conference, it will close down the conference.&nbsp;&nbsp; so i'm
            trying to use the 'endconf' flag, but apparently it is not
            valid syntax for bridging conference as I get a config error
            while running.&nbsp; If I take out the '+flags{endconf} ', i wont
            get a config error while running, but then conference will
            only close down when there is 1 person left.<br><br>
            &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; I have also tried creating 2 difference conference
            profiles,&nbsp; one profile has the 'endconf' set in the
            'member-flags', one profiles doesnt have 'endconf' set, so
            user99 and user20 joins the 20 <at> profile1 conference, and
            other users joins the 20 <at> profile2 conference, that doesnt
            seem to work neither.&nbsp;&nbsp; I have pasted conference.conf.xml
            file as well for ur review. <br>
            &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <br>&lt;include&gt;
  &lt;context name="test"&gt;
    &lt;extension name="global" continue="true"&gt;
      &lt;condition&gt;
      &lt;/condition&gt;
    &lt;/extension&gt;

    &lt;extension name="conf"&gt;
      &lt;condition field="destination_number" expression="^(20)$" &gt; &lt;/condition&gt;
        &lt;action application="answer"/&gt;        
        &lt;action application="conference" data="bridge:20 <at> default+flags{endconf}:user/10"/&gt;
        &lt;action application="conference" data="20 <at> default"/&gt; 
      &lt;/condition&gt;
    &lt;/extension&gt;   
  &lt;/context&gt;
&lt;/include&gt;
            <br>&lt;configuration name="conference.conf" description="Audio Conference"&gt;
  &lt;!-- Advertise certain presence on startup . --&gt;
  &lt;advertise&gt;
    &lt;room name="3001 <at> $${domain}" status="FreeSWITCH"/&gt;
  &lt;/advertise&gt;

  &lt;!-- These are the default keys that map when you do not specify a caller control group --&gt;	
  &lt;!-- Note: none and default are reserved names for group names.  Disabled if dist-dtmf member flag is set. --&gt;	
  &lt;caller-controls&gt;
    &lt;group name="default"&gt;
      &lt;control action="mute" digits="0"/&gt;
      &lt;control action="deaf mute" digits="*"/&gt;
      &lt;control action="energy up" digits="9"/&gt;
      &lt;control action="energy equ" digits="8"/&gt;
      &lt;control action="energy dn" digits="7"/&gt;
      &lt;control action="vol talk up" digits="3"/&gt;
      &lt;control action="vol talk zero" digits="2"/&gt;
      &lt;control action="vol talk dn" digits="1"/&gt;
      &lt;control action="vol listen up" digits="6"/&gt;
      &lt;control action="vol listen zero" digits="5"/&gt;
      &lt;control action="vol listen dn" digits="4"/&gt;
      &lt;control action="hangup" digits="#"/&gt;
    &lt;/group&gt;
  &lt;/caller-controls&gt;

  &lt;!-- Profiles are collections of settings you can reference by name. --&gt;
  &lt;profiles&gt;
    &lt;!--If no profile is specified it will default to "default"--&gt;
    &lt;profile name="default"&gt;
      &lt;!-- Domain (for presence) --&gt;
      &lt;param name="domain" value="$${domain}"/&gt;
      &lt;!-- Sample Rate--&gt;
      &lt;param name="rate" value="8000"/&gt;
      &lt;!-- Number of milliseconds per frame --&gt;
      &lt;param name="interval" value="20"/&gt;
      &lt;!-- Energy level required for audio to be sent to the other users --&gt;
      &lt;param name="energy-level" value="300"/&gt;

      &lt;!--Can be | delim of waste|mute|deaf|dist-dtmf waste will always transmit data to each channel
          even during silence.  dist-dtmf propagates dtmfs to all other members, but channel controls
	  via dtmf will be disabled. --&gt;
      &lt;param name="member-flags" value="dist-dtmf"/&gt;

      &lt;!-- Name of the caller control group to use for this profile --&gt;
      &lt;!-- &lt;param name="caller-controls" value="some name"/&gt; --&gt;
      &lt;!-- TTS Engine to use --&gt;
      &lt;!--&lt;param name="tts-engine" value="cepstral"/&gt;--&gt;
      &lt;!-- TTS Voice to use --&gt;
      &lt;!--&lt;param name="tts-voice" value="david"/&gt;--&gt;

      &lt;!-- If TTS is enabled all audio-file params beginning with --&gt;
      &lt;!-- 'say:' will be considered text to say with TTS --&gt;
      &lt;!-- Override the default path here, after which you use relative paths in the other sound params --&gt;
      &lt;!-- Note: The default path is the conference's first caller's sound_prefix --&gt;
      &lt;!--&lt;param name="sound-prefix" value="$${sounds_dir}/en/us/callie"/&gt;--&gt;
      &lt;!-- File to play to acknowledge succees --&gt;
      &lt;!--&lt;param name="ack-sound" value="beep.wav"/&gt;--&gt;
      &lt;!-- File to play to acknowledge failure --&gt;
      &lt;!--&lt;param name="nack-sound" value="beeperr.wav"/&gt;--&gt;
      &lt;!-- File to play to acknowledge muted --&gt;
      &lt;!-- Conference pin --&gt;
      &lt;!--&lt;param name="pin" value="12345"/&gt;--&gt;
      &lt;!-- Default Caller ID Name for outbound calls --&gt;
      &lt;param name="caller-id-name" value="$${outbound_caller_name}"/&gt;
      &lt;!-- Default Caller ID Number for outbound calls --&gt;
      &lt;param name="caller-id-number" value="1234"/&gt;
      &lt;!-- Suppress start and stop talking events --&gt;
      &lt;!-- &lt;param name="suppress-events" value="start-talking,stop-talking"/&gt; --&gt;
      &lt;!-- enable comfort noise generation --&gt;
      &lt;param name="comfort-noise" value="true"/&gt;
      &lt;!-- Uncomment auto-record to toggle recording every conference call. --&gt;
      &lt;!-- Another valid value is   <a href="mailto:shout://user:pass <at> server.com/live.mp3" target="_blank">shout://user:pass@.../live.mp3</a>   --&gt;
      &lt;!--
      &lt;param name="auto-record" value="$${recordings_dir}/${conference_name}_${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/&gt;
      --&gt;
    &lt;/profile&gt;
<div><div class="h5">
            <div>
              <div> <br><br><br><br><br><br><br><br><br><br><br><br><br><br><br><br><br><br><br>
                On 6/30/2011 1:07 PM, Michael Collins wrote:
                <blockquote type="cite">Can you pastebin exactly what
                  you are doing to establish the call? Including any
                  relevant dialplan entries. Also, if you have modified
                  conference.conf.xml we would like to see that also.<br><br>
                  -MC<br><br><div class="gmail_quote"> On Thu, Jun 30, 2011 at
                    10:59 AM, ran zhang <span dir="ltr">&lt;<a href="mailto:rzhang@..." target="_blank">rzhang@...</a>&gt;</span>
                    wrote:<br><blockquote class="gmail_quote"> hi all:<br><br>
                      I'm trying to creating a conference, so when first
                      member enters the<br>
                      conference, he has to invite another members<br>
                      and have at least 1 other member to join to have
                      the conference established, so i'm<br>
                      using bridging conference.<br><br>
                      I need this conference to be terminated when the
                      original creator of the<br>
                      conference leaves no matter how many members are
                      still left in the conference.<br><br>
                      i'm trying to set 'endconf' flag in a bridging
                      conference using<br>
                      'bridge:confname+flag{endconf}:user/10',<br>
                      so it wil invite user extension 10, but its giving
                      me config error while<br>
                      running.<br><br>
                      can someone tell me what to do to solve this
                      problem or get around? &nbsp;the key<br>
                      is i only want the original member to be able to
                      terminate the conference<br>
                      when he leaves, not other members assuming there
                      are at least 2 members.<br><br><br>
                      _______________________________________________<br>
                      Join us at ClueCon 2011, Aug 9-11, Chicago<br><a href="http://www.cluecon.com" target="_blank">http://www.cluecon.com</a>
                      877-7-4ACLUE<br><br>
                      FreeSWITCH-users mailing list<br><a href="mailto:FreeSWITCH-users@...rg" target="_blank">FreeSWITCH-users@...</a><br><a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br>
                      UNSUBSCRIBE:<a href="http://lists.freeswitch.org/mailman/options/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/options/freeswitch-users</a><br><a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br>
</blockquote>
                  </div>
                  <br>_______________________________________________
Join us at ClueCon 2011, Aug 9-11, Chicago
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                </blockquote>
                <br>
</div>
            </div>
          </div></div>
</div>
        </blockquote>
      </div>
      <br>
</blockquote>
    <br>
</div>

<br>_______________________________________________<br>
Join us at ClueCon 2011, Aug 9-11, Chicago<br><a href="http://www.cluecon.com" target="_blank">http://www.cluecon.com</a> 877-7-4ACLUE<br><br>
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</blockquote>
</div>
<br>
</div>
Nandy Dagondon | 1 Jul 2011 04:12
Picon

Re: please help!!! how to set flag 'endconf' in bridging conference

hi michael,

pls disregard me last post. i just digged the "endconf" parameter. i'll try it. tks.


On Fri, Jul 1, 2011 at 10:04 AM, Nandy Dagondon <gcd <at> i.ph> wrote:
hi michael,

i'd like to have that feature, too, because our telco has an unusual line signalling - it only tear down the connection when the A-party hangs up first (Clear Forward).  if i transfer the callee to the conference, the telco line remains off-hook forever. as an interim solution, i'm using the terminate_on_silence parameter.

-nandy

On Fri, Jul 1, 2011 at 9:29 AM, ran zhang <rzhang-JA7S4awhwUdeQIzoc+Smag@public.gmane.org> wrote:
       I attached the section in my dialplan that handles the bridging conference, when the first user (just say user 99)  dials '20', he will invite user 10 to join conference, the conference will only be established if user 10 accepts the invitation, after this, other users can join the conference by dialing '20'.

       I want that if the first user (user 99 in this case)  or the user been invited (user 10 in this case)  leaves the conference, no matter how many people are still in the conference, it will close down the conference.   so i'm trying to use the 'endconf' flag, but apparently it is not valid syntax for bridging conference as I get a config error while running.  If I take out the '+flags{endconf} ', i wont get a config error while running, but then conference will only close down when there is 1 person left.

      I have also tried creating 2 difference conference profiles,  one profile has the 'endconf' set in the 'member-flags', one profiles doesnt have 'endconf' set, so user99 and user20 joins the 20 <at> profile1 conference, and other users joins the 20 <at> profile2 conference, that doesnt seem to work neither.   I have pasted conference.conf.xml file as well for ur review.
      
<include> <context name="test"> <extension name="global" continue="true"> <condition> </condition> </extension> <extension name="conf"> <condition field="destination_number" expression="^(20)$" > </condition> <action application="answer"/> <action application="conference" data="bridge:20 <at> default+flags{endconf}:user/10"/> <action application="conference" data="20 <at> default"/> </condition> </extension> </context> </include>



On 6/30/2011 6:16 PM, Michael Collins wrote:
Please put this information on pastebin and reply to the list so that we can all discuss it.
-MC

On Thu, Jun 30, 2011 at 1:51 PM, ran zhang <rzhang-JA7S4awhwUdeQIzoc+Smag@public.gmane.org> wrote:
Mr Collins:

       I attached the section in my dialplan that handles the bridging conference, when the first user (just say user 99)  dials '20', he will invite user 10 to join conference, the conference will only be established if user 10 accepts the invitation, after this, other users can join the conference by dialing '20'.

       I want that if the first user (user 99 in this case)  or the user been invited (user 10 in this case)  leaves the conference, no matter how many people are still in the conference, it will close down the conference.   so i'm trying to use the 'endconf' flag, but apparently it is not valid syntax for bridging conference as I get a config error while running.  If I take out the '+flags{endconf} ', i wont get a config error while running, but then conference will only close down when there is 1 person left.

      I have also tried creating 2 difference conference profiles,  one profile has the 'endconf' set in the 'member-flags', one profiles doesnt have 'endconf' set, so user99 and user20 joins the 20 <at> profile1 conference, and other users joins the 20 <at> profile2 conference, that doesnt seem to work neither.   I have pasted conference.conf.xml file as well for ur review.
      
<include> <context name="test"> <extension name="global" continue="true"> <condition> </condition> </extension> <extension name="conf"> <condition field="destination_number" expression="^(20)$" > </condition> <action application="answer"/> <action application="conference" data="bridge:20 <at> default+flags{endconf}:user/10"/> <action application="conference" data="20 <at> default"/> </condition> </extension> </context> </include>
<configuration name="conference.conf" description="Audio Conference"> <!-- Advertise certain presence on startup . --> <advertise> <room name="3001 <at> $${domain}" status="FreeSWITCH"/> </advertise> <!-- These are the default keys that map when you do not specify a caller control group --> <!-- Note: none and default are reserved names for group names. Disabled if dist-dtmf member flag is set. --> <caller-controls> <group name="default"> <control action="mute" digits="0"/> <control action="deaf mute" digits="*"/> <control action="energy up" digits="9"/> <control action="energy equ" digits="8"/> <control action="energy dn" digits="7"/> <control action="vol talk up" digits="3"/> <control action="vol talk zero" digits="2"/> <control action="vol talk dn" digits="1"/> <control action="vol listen up" digits="6"/> <control action="vol listen zero" digits="5"/> <control action="vol listen dn" digits="4"/> <control action="hangup" digits="#"/> </group> </caller-controls> <!-- Profiles are collections of settings you can reference by name. --> <profiles> <!--If no profile is specified it will default to "default"--> <profile name="default"> <!-- Domain (for presence) --> <param name="domain" value="$${domain}"/> <!-- Sample Rate--> <param name="rate" value="8000"/> <!-- Number of milliseconds per frame --> <param name="interval" value="20"/> <!-- Energy level required for audio to be sent to the other users --> <param name="energy-level" value="300"/> <!--Can be | delim of waste|mute|deaf|dist-dtmf waste will always transmit data to each channel even during silence. dist-dtmf propagates dtmfs to all other members, but channel controls via dtmf will be disabled. --> <param name="member-flags" value="dist-dtmf"/> <!-- Name of the caller control group to use for this profile --> <!-- <param name="caller-controls" value="some name"/> --> <!-- TTS Engine to use --> <!--<param name="tts-engine" value="cepstral"/>--> <!-- TTS Voice to use --> <!--<param name="tts-voice" value="david"/>--> <!-- If TTS is enabled all audio-file params beginning with --> <!-- 'say:' will be considered text to say with TTS --> <!-- Override the default path here, after which you use relative paths in the other sound params --> <!-- Note: The default path is the conference's first caller's sound_prefix --> <!--<param name="sound-prefix" value="$${sounds_dir}/en/us/callie"/>--> <!-- File to play to acknowledge succees --> <!--<param name="ack-sound" value="beep.wav"/>--> <!-- File to play to acknowledge failure --> <!--<param name="nack-sound" value="beeperr.wav"/>--> <!-- File to play to acknowledge muted --> <!-- Conference pin --> <!--<param name="pin" value="12345"/>--> <!-- Default Caller ID Name for outbound calls --> <param name="caller-id-name" value="$${outbound_caller_name}"/> <!-- Default Caller ID Number for outbound calls --> <param name="caller-id-number" value="1234"/> <!-- Suppress start and stop talking events --> <!-- <param name="suppress-events" value="start-talking,stop-talking"/> --> <!-- enable comfort noise generation --> <param name="comfort-noise" value="true"/> <!-- Uncomment auto-record to toggle recording every conference call. --> <!-- Another valid value is shout://user:pass-WQCBuIXaXYjQT0dZR+AlfA@public.gmane.org/live.mp3 --> <!-- <param name="auto-record" value="$${recordings_dir}/${conference_name}_${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> --> </profile>



















On 6/30/2011 1:07 PM, Michael Collins wrote:
Can you pastebin exactly what you are doing to establish the call? Including any relevant dialplan entries. Also, if you have modified conference.conf.xml we would like to see that also.

-MC

On Thu, Jun 30, 2011 at 10:59 AM, ran zhang <rzhang-JA7S4awhwUdeQIzoc+Smag@public.gmane.org> wrote:
hi all:

I'm trying to creating a conference, so when first member enters the
conference, he has to invite another members
and have at least 1 other member to join to have the conference established, so i'm
using bridging conference.

I need this conference to be terminated when the original creator of the
conference leaves no matter how many members are still left in the conference.

i'm trying to set 'endconf' flag in a bridging conference using
'bridge:confname+flag{endconf}:user/10',
so it wil invite user extension 10, but its giving me config error while
running.

can someone tell me what to do to solve this problem or get around?  the key
is i only want the original member to be able to terminate the conference
when he leaves, not other members assuming there are at least 2 members.


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<div>
<div>hi michael,<br>
</div>
<div><br></div>
<div>pls disregard me last post. i just digged the "endconf" parameter. i'll try it. tks.</div>
<br><br><div class="gmail_quote">On Fri, Jul 1, 2011 at 10:04 AM, Nandy Dagondon <span dir="ltr">&lt;<a href="mailto:gcd <at> i.ph">gcd <at> i.ph</a>&gt;</span> wrote:<br><blockquote class="gmail_quote">
<div>hi michael,<br>
</div>
<div><br></div>
<div>i'd like to have that feature, too, because our telco has an unusual line signalling - it only tear down the connection when the A-party hangs up first (Clear Forward). &nbsp;if i transfer the callee to the conference, the telco line remains off-hook forever. as an interim solution, i'm using the terminate_on_silence parameter.</div>

<br><div>-nandy<br>
</div>
<div><div class="h5">
<div><br></div>
<div class="gmail_quote">On Fri, Jul 1, 2011 at 9:29 AM, ran zhang <span dir="ltr">&lt;<a href="mailto:rzhang@...m" target="_blank">rzhang@...</a>&gt;</span> wrote:<br><blockquote class="gmail_quote">

  

    

  <div text="#000000" bgcolor="#ffffff">
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; I attached the section in my dialplan that handles the
    bridging conference, when the first user (just say user 99)&nbsp; dials
    '20', he will invite user 10 to join conference, the conference will
    only be established if user 10 accepts the invitation, after this,
    other users can join the conference by dialing '20'. <br><br>
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; I want that if the first user (user 99 in this case)&nbsp; or the
    user been invited (user 10 in this case)&nbsp; leaves the conference, no
    matter how many people are still in the conference, it will close
    down the conference.&nbsp;&nbsp; so i'm trying to use the 'endconf' flag, but
    apparently it is not valid syntax for bridging conference as I get a
    config error while running.&nbsp; If I take out the '+flags{endconf} ', i
    wont get a config error while running, but then conference will only
    close down when there is 1 person left.<br><br>
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; I have also tried creating 2 difference conference profiles,&nbsp;
    one profile has the 'endconf' set in the 'member-flags', one
    profiles doesnt have 'endconf' set, so user99 and user20 joins the
    20 <at> profile1 conference, and other users joins the 20 <at> profile2
    conference, that doesnt seem to work neither.&nbsp;&nbsp; I have pasted
    conference.conf.xml file as well for ur review. <br>
    &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <br>&lt;include&gt;
  &lt;context name="test"&gt;
    &lt;extension name="global" continue="true"&gt;
      &lt;condition&gt;
      &lt;/condition&gt;
    &lt;/extension&gt;

    &lt;extension name="conf"&gt;
      &lt;condition field="destination_number" expression="^(20)$" &gt; &lt;/condition&gt;
        &lt;action application="answer"/&gt;        
        &lt;action application="conference" data="bridge:20 <at> default+flags{endconf}:user/10"/&gt;
        &lt;action application="conference" data="20 <at> default"/&gt; 
      &lt;/condition&gt;
    &lt;/extension&gt;   
  &lt;/context&gt;
&lt;/include&gt;    <br><br><br><br>
    On 6/30/2011 6:16 PM, Michael Collins wrote:
    <blockquote type="cite">Please put this information on pastebin and reply to
      the list so that we can all discuss it.<br>
      -MC<br><br><div class="gmail_quote">On Thu, Jun 30, 2011 at 1:51 PM, ran
        zhang <span dir="ltr">&lt;<a href="mailto:rzhang <at> gosilverplus.com" target="_blank">rzhang@...</a>&gt;</span>
        wrote:<br><blockquote class="gmail_quote">
          <div text="#000000" bgcolor="#ffffff"> Mr Collins:<br><br>
            &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; I attached the section in my dialplan that handles
            the bridging conference, when the first user (just say user
            99)&nbsp; dials '20', he will invite user 10 to join conference,
            the conference will only be established if user 10 accepts
            the invitation, after this, other users can join the
            conference by dialing '20'. <br><br>
            &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; I want that if the first user (user 99 in this case)&nbsp;
            or the user been invited (user 10 in this case)&nbsp; leaves the
            conference, no matter how many people are still in the
            conference, it will close down the conference.&nbsp;&nbsp; so i'm
            trying to use the 'endconf' flag, but apparently it is not
            valid syntax for bridging conference as I get a config error
            while running.&nbsp; If I take out the '+flags{endconf} ', i wont
            get a config error while running, but then conference will
            only close down when there is 1 person left.<br><br>
            &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; I have also tried creating 2 difference conference
            profiles,&nbsp; one profile has the 'endconf' set in the
            'member-flags', one profiles doesnt have 'endconf' set, so
            user99 and user20 joins the 20 <at> profile1 conference, and
            other users joins the 20 <at> profile2 conference, that doesnt
            seem to work neither.&nbsp;&nbsp; I have pasted conference.conf.xml
            file as well for ur review. <br>
            &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <br>&lt;include&gt;
  &lt;context name="test"&gt;
    &lt;extension name="global" continue="true"&gt;
      &lt;condition&gt;
      &lt;/condition&gt;
    &lt;/extension&gt;

    &lt;extension name="conf"&gt;
      &lt;condition field="destination_number" expression="^(20)$" &gt; &lt;/condition&gt;
        &lt;action application="answer"/&gt;        
        &lt;action application="conference" data="bridge:20 <at> default+flags{endconf}:user/10"/&gt;
        &lt;action application="conference" data="20 <at> default"/&gt; 
      &lt;/condition&gt;
    &lt;/extension&gt;   
  &lt;/context&gt;
&lt;/include&gt;
            <br>&lt;configuration name="conference.conf" description="Audio Conference"&gt;
  &lt;!-- Advertise certain presence on startup . --&gt;
  &lt;advertise&gt;
    &lt;room name="3001 <at> $${domain}" status="FreeSWITCH"/&gt;
  &lt;/advertise&gt;

  &lt;!-- These are the default keys that map when you do not specify a caller control group --&gt;	
  &lt;!-- Note: none and default are reserved names for group names.  Disabled if dist-dtmf member flag is set. --&gt;	
  &lt;caller-controls&gt;
    &lt;group name="default"&gt;
      &lt;control action="mute" digits="0"/&gt;
      &lt;control action="deaf mute" digits="*"/&gt;
      &lt;control action="energy up" digits="9"/&gt;
      &lt;control action="energy equ" digits="8"/&gt;
      &lt;control action="energy dn" digits="7"/&gt;
      &lt;control action="vol talk up" digits="3"/&gt;
      &lt;control action="vol talk zero" digits="2"/&gt;
      &lt;control action="vol talk dn" digits="1"/&gt;
      &lt;control action="vol listen up" digits="6"/&gt;
      &lt;control action="vol listen zero" digits="5"/&gt;
      &lt;control action="vol listen dn" digits="4"/&gt;
      &lt;control action="hangup" digits="#"/&gt;
    &lt;/group&gt;
  &lt;/caller-controls&gt;

  &lt;!-- Profiles are collections of settings you can reference by name. --&gt;
  &lt;profiles&gt;
    &lt;!--If no profile is specified it will default to "default"--&gt;
    &lt;profile name="default"&gt;
      &lt;!-- Domain (for presence) --&gt;
      &lt;param name="domain" value="$${domain}"/&gt;
      &lt;!-- Sample Rate--&gt;
      &lt;param name="rate" value="8000"/&gt;
      &lt;!-- Number of milliseconds per frame --&gt;
      &lt;param name="interval" value="20"/&gt;
      &lt;!-- Energy level required for audio to be sent to the other users --&gt;
      &lt;param name="energy-level" value="300"/&gt;

      &lt;!--Can be | delim of waste|mute|deaf|dist-dtmf waste will always transmit data to each channel
          even during silence.  dist-dtmf propagates dtmfs to all other members, but channel controls
	  via dtmf will be disabled. --&gt;
      &lt;param name="member-flags" value="dist-dtmf"/&gt;

      &lt;!-- Name of the caller control group to use for this profile --&gt;
      &lt;!-- &lt;param name="caller-controls" value="some name"/&gt; --&gt;
      &lt;!-- TTS Engine to use --&gt;
      &lt;!--&lt;param name="tts-engine" value="cepstral"/&gt;--&gt;
      &lt;!-- TTS Voice to use --&gt;
      &lt;!--&lt;param name="tts-voice" value="david"/&gt;--&gt;

      &lt;!-- If TTS is enabled all audio-file params beginning with --&gt;
      &lt;!-- 'say:' will be considered text to say with TTS --&gt;
      &lt;!-- Override the default path here, after which you use relative paths in the other sound params --&gt;
      &lt;!-- Note: The default path is the conference's first caller's sound_prefix --&gt;
      &lt;!--&lt;param name="sound-prefix" value="$${sounds_dir}/en/us/callie"/&gt;--&gt;
      &lt;!-- File to play to acknowledge succees --&gt;
      &lt;!--&lt;param name="ack-sound" value="beep.wav"/&gt;--&gt;
      &lt;!-- File to play to acknowledge failure --&gt;
      &lt;!--&lt;param name="nack-sound" value="beeperr.wav"/&gt;--&gt;
      &lt;!-- File to play to acknowledge muted --&gt;
      &lt;!-- Conference pin --&gt;
      &lt;!--&lt;param name="pin" value="12345"/&gt;--&gt;
      &lt;!-- Default Caller ID Name for outbound calls --&gt;
      &lt;param name="caller-id-name" value="$${outbound_caller_name}"/&gt;
      &lt;!-- Default Caller ID Number for outbound calls --&gt;
      &lt;param name="caller-id-number" value="1234"/&gt;
      &lt;!-- Suppress start and stop talking events --&gt;
      &lt;!-- &lt;param name="suppress-events" value="start-talking,stop-talking"/&gt; --&gt;
      &lt;!-- enable comfort noise generation --&gt;
      &lt;param name="comfort-noise" value="true"/&gt;
      &lt;!-- Uncomment auto-record to toggle recording every conference call. --&gt;
      &lt;!-- Another valid value is   <a href="mailto:shout://user:pass <at> server.com/live.mp3" target="_blank">shout://user:pass@.../live.mp3</a>   --&gt;
      &lt;!--
      &lt;param name="auto-record" value="$${recordings_dir}/${conference_name}_${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/&gt;
      --&gt;
    &lt;/profile&gt;
<div><div>
            <div>
              <div> <br><br><br><br><br><br><br><br><br><br><br><br><br><br><br><br><br><br><br>
                On 6/30/2011 1:07 PM, Michael Collins wrote:
                <blockquote type="cite">Can you pastebin exactly what
                  you are doing to establish the call? Including any
                  relevant dialplan entries. Also, if you have modified
                  conference.conf.xml we would like to see that also.<br><br>
                  -MC<br><br><div class="gmail_quote"> On Thu, Jun 30, 2011 at
                    10:59 AM, ran zhang <span dir="ltr">&lt;<a href="mailto:rzhang@..." target="_blank">rzhang@...</a>&gt;</span>
                    wrote:<br><blockquote class="gmail_quote"> hi all:<br><br>
                      I'm trying to creating a conference, so when first
                      member enters the<br>
                      conference, he has to invite another members<br>
                      and have at least 1 other member to join to have
                      the conference established, so i'm<br>
                      using bridging conference.<br><br>
                      I need this conference to be terminated when the
                      original creator of the<br>
                      conference leaves no matter how many members are
                      still left in the conference.<br><br>
                      i'm trying to set 'endconf' flag in a bridging
                      conference using<br>
                      'bridge:confname+flag{endconf}:user/10',<br>
                      so it wil invite user extension 10, but its giving
                      me config error while<br>
                      running.<br><br>
                      can someone tell me what to do to solve this
                      problem or get around? &nbsp;the key<br>
                      is i only want the original member to be able to
                      terminate the conference<br>
                      when he leaves, not other members assuming there
                      are at least 2 members.<br><br><br>
                      _______________________________________________<br>
                      Join us at ClueCon 2011, Aug 9-11, Chicago<br><a href="http://www.cluecon.com" target="_blank">http://www.cluecon.com</a>
                      877-7-4ACLUE<br><br>
                      FreeSWITCH-users mailing list<br><a href="mailto:FreeSWITCH-users@...rg" target="_blank">FreeSWITCH-users@...</a><br><a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br>
                      UNSUBSCRIBE:<a href="http://lists.freeswitch.org/mailman/options/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/options/freeswitch-users</a><br><a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br>
</blockquote>
                  </div>
                  <br>_______________________________________________
Join us at ClueCon 2011, Aug 9-11, Chicago
<a href="http://www.cluecon.com" target="_blank">http://www.cluecon.com</a> 877-7-4ACLUE

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                </blockquote>
                <br>
</div>
            </div>
          </div></div>
</div>
        </blockquote>
      </div>
      <br>
</blockquote>
    <br>
</div>

<br>_______________________________________________<br>
Join us at ClueCon 2011, Aug 9-11, Chicago<br><a href="http://www.cluecon.com" target="_blank">http://www.cluecon.com</a> 877-7-4ACLUE<br><br>
FreeSWITCH-users mailing list<br><a href="mailto:FreeSWITCH-users@..." target="_blank">FreeSWITCH-users@...</a><br><a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br>
UNSUBSCRIBE:<a href="http://lists.freeswitch.org/mailman/options/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/options/freeswitch-users</a><br><a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br><br>
</blockquote>
</div>
<br>
</div></div>
</blockquote>
</div>
<br>
</div>
Bret Watson | 1 Jul 2011 05:34

inbound trunk from Linksys 3102?

Hi All,
one of the few things that really worked on my * setup was the 
inbound trunk from the 3102...

* config was
trunk name 1-pstn
disallow=all
allow=ulaw
canreinvite=no
context=from-trunk
dtmfmode=rfc2833
host=dynamic
incominglimit=1
nat=never
port=5061
qualify=yes
secret=xxxxxxxx
type=friend
username=1-pstn

How do I translate this into freeswitch?

Thanks!

Bret

Max Alex | 1 Jul 2011 06:12
Picon

Re: Answer issue on inbound call

Hi,
Can any one provide their suggestions and help for this issue,
I really need to resolve and get this working.

Thanks,
Max Alex
Voip Developer



On Tue, Jun 28, 2011 at 6:01 PM, Max Alex <max.asterisk-Re5JQEeQqe8AvxtiuMwx3w@public.gmane.org> wrote:
Hi,
Thanks for your reply.
I have enabled logger as per your help.
I have given completed log on following link of pastebin.
http://pastebin.freeswitch.org/16616

You can see this line as it is showing answered and this call is answered on my cell phone but on softphone it is ringing and i have rejected from there.
2011
-06-28 17:47:52.909657 [NOTICE] mod_freetdm.c:1953 Channel [FreeTDM/2:1/01234567890] has been answered    

It is pre answering the cell phone when it is ringing on phone
Waiting for your help.

Thanks,
Max Alex
Voip Developer



On Mon, Jun 27, 2011 at 9:15 PM, Michael Collins <msc-YF8E+gPBBv73h3GqohbjpQ@public.gmane.org> wrote:
Get a complete, unedited, unabridged console debug log w/ siptrace and put it on pastebin w/ "FreeSWITCH Log" for the syntax highlighting. Use "sofia global siptrace on" to make sure you can all SIP traffic.

-MC


On Mon, Jun 27, 2011 at 6:05 AM, Max Alex <max.asterisk-Re5JQEeQqe8AvxtiuMwx3w@public.gmane.org> wrote:
Hi,
Thanks for reply,
I have tried the same way and reloaded freeswitch, but still it is answered on first ring of the call.
When it is ringing the call on 1001 and the same time it is answered on cell phone, so something is done when it is ringing on 1001.

Here is the dialplan for the same

    <extension name="Local_Extension">
      <condition field="destination_number" expression="^(10[01][0-9])$">
        <action application="set" data="dialed_extension=$1"/>
        <action application="export" data="dialed_extension=$1"/>
        <action application="bind_meta_app" data="2 b s record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
        <action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/>-->
        <action application="set" data="ringback=${us-ring}"/>
        <action application="set" data="transfer_ringback=$${hold_music}"/>
        <action application="set" data="call_timeout=30"/>
          <action application="set" data="hangup_after_bridge=true"/>
         <action application="set" data="continue_on_fail=true"/>
        <action application="hash" data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/>
        <action application="hash" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/>
        <action application="set" data="called_party_callgroup=${user_data(${dialed_extension} <at> ${domain_name} var callgroup)}"/>
        <action application="hash" data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/>

       <action application="bridge" data="{ignore_early_media=true}user/${dialed_extension} <at> ${domain_name}"/>
       <action application="sleep" data="1000"/>
        <action application="voicemail" data="default ${domain_name} ${dialed_extension}"/>
      </condition>
    </extension>


Please help me for this issue.


Thanks,
Max Alex
Voip Developer



On Fri, Jun 24, 2011 at 11:38 AM, Michael Collins <msc-YF8E+gPBBv73h3GqohbjpQ@public.gmane.org> wrote:
Are you using the default dialplan? I think you might just need to ignore early media on your bridge app. If you are using the default.xml file then locate "Local_Extension" and the bridge line:

    <action application="bridge" data="user/${dialed_extension} <at> ${domain_name}"/>

Change it to this, then try again:

    <action application="bridge" data="{ignore_early_media=true}user/${dialed_extension} <at> ${domain_name}"/>

If I understand correctly, the "symptom" you are experiencing is the normal operation of the bridge app (and it's cousin, the originate API command). When the b-leg sends back media, including ringing, the bridge (or the originate) will be considered "successful," and in the case of bridge, the b-leg's audio (the early media) will be connected to the a-leg. If you set ignore_early_media=true then you are explicitly telling the bridge app that you only want to connect the b-leg to the a-leg if the b-leg actually answers.

I hope that made sense...

-MC



On Thu, Jun 23, 2011 at 9:32 PM, Max Alex <max.asterisk-Re5JQEeQqe8AvxtiuMwx3w@public.gmane.org> wrote:
Hi,
Thanks for reply.
Current scenario is below.

PSTN call -> sangoma device -> freeswitch incoming context -> routed to 1001 -> ringing (Answered on cellphone)
Here when it is routed to 1001 the call it is started ringing, but on phone that call is answered and hearding sound of ringing.

Required flow:
PSTN call -> sangoma device -> freeswitch incoming context -> routed to 1001 -> ringing (Ringing on cellphone)

I hope it is understable, the call should not be answered until 1001 answer it, right not when it is started ring it is answered on cell phone.
that should not be happen as it is not answered yet.

Waiting for your reply.


Thanks,
Max Alex
Voip Developer



On Fri, Jun 24, 2011 at 5:48 AM, Michael Collins <msc-YF8E+gPBBv73h3GqohbjpQ@public.gmane.org> wrote:
I'm not sure I understand the problem. What is happening vs. what you believe should be happening?
-MC


On Thu, Jun 23, 2011 at 3:31 AM, Max Alex <max.asterisk-Re5JQEeQqe8AvxtiuMwx3w@public.gmane.org> wrote:
Hi,
Thanks for your reply.
Here is my configuration and log
http://pastebin.freeswitch.org/16571

I am using A200 analog sangoma device with freeswitch, it is working fine but when it is routing call to 1001 then it is answered.
Please provider your suggestions.

Thanks,
Max Alex
Voip Developer




On Wed, Jun 22, 2011 at 8:51 PM, Michael Collins <msc-YF8E+gPBBv73h3GqohbjpQ@public.gmane.org> wrote:
I thought the A200 was an analog card? Maybe I have my numbers mixed up...

Go ahead and collect a debug log of this call. It might help to have your configs posted as well. Use pastebin.freeswitch.org. See this wiki article for tips on how to collect information:

-MC

On Wed, Jun 22, 2011 at 3:23 AM, Max Alex <max.asterisk-Re5JQEeQqe8AvxtiuMwx3w@public.gmane.org> wrote:
Hi,
I have installed freeswitch latest version with sangoma card A200 as well,
I have installed and configured freetdm module with wanpipe drivers for freeswitch,
We are properly receiving the inbound calls in public context and then we are routing that call to 1001 extension,
it is properly routing to 1001 as well, but we have one issue while routing on 1001.

Here is the issue description.
I am calling from my cell phone to that DID number of pri line, and then it will start ringing on 1001 extension,
When 1001 extension start ringing the call is answered on my cell phone,
it is something like freeswitch preanswer or autoanswer the call, how can i stop this answer call when it is ringing on 1001 extension,
Waiting for good reply.

Thanks,
Max Alex



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<div>
<p>Hi,<br>Can any one provide their suggestions and help for this issue,<br>I really need to resolve and get this working.<br><br clear="all">Thanks,<br>Max Alex<br>Voip Developer<br><br><br><br></p>
<div class="gmail_quote">On Tue, Jun 28, 2011 at 6:01 PM, Max Alex <span dir="ltr">&lt;<a href="mailto:max.asterisk@...">max.asterisk@...</a>&gt;</span> wrote:<br><blockquote class="gmail_quote">
<span><span><div class="im">Hi,<br>Thanks for your reply.<br>
</div>I have enabled logger as per your help.<br>I have given completed log on following link of pastebin.<br><a href="http://pastebin.freeswitch.org/16616" target="_blank">http://pastebin.freeswitch.org/16616</a><br><br>You can see this line as it is showing answered and this call is answered on my cell phone but on softphone it is ringing and i have rejected from there.<br>2011</span><span>-06</span><span>-28</span> <span>17</span>:<span>47</span>:<span>52.909657</span> <span>[</span>NOTICE<span>]</span> mod_freetdm.c:<span>1953</span> Channel <span>[</span>FreeTDM/<span>2</span>:<span>1</span>/<span>01234567890</span><span>]</span> has been answered&nbsp; &nbsp;&nbsp; <br><br>It is pre answering the cell phone when it is ringing on phone<br>Waiting for your help.<br><br clear="all"></span><div class="im">Thanks,<br>Max Alex<br>Voip Developer<br><br><br><br>
</div>
<div>
<div></div>
<div class="h5">
<div class="gmail_quote">On Mon, Jun 27, 2011 at 9:15 PM, Michael Collins <span dir="ltr">&lt;<a href="mailto:msc@..." target="_blank">msc@...</a>&gt;</span> wrote:<br><blockquote class="gmail_quote">
Get a complete, unedited, unabridged console debug log w/ siptrace and put it on pastebin w/ "FreeSWITCH Log" for the syntax highlighting. Use "sofia global siptrace on" to make sure you can all SIP traffic.<br>
<br>-MC<div>
<div></div>
<div>
<br><br><div class="gmail_quote">On Mon, Jun 27, 2011 at 6:05 AM, Max Alex <span dir="ltr">&lt;<a href="mailto:max.asterisk@..." target="_blank">max.asterisk@...</a>&gt;</span> wrote:<br><blockquote class="gmail_quote">
Hi,<br>Thanks for reply,<br>I have tried the same way and reloaded freeswitch, but still it is answered on first ring of the call.<br>When it is ringing the call on 1001 and the same time it is answered on cell phone, so something is done when it is ringing on 1001.<br><br>Here is the dialplan for the same<br><br>&nbsp;&nbsp;&nbsp; &lt;extension name="Local_Extension"&gt;<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &lt;condition field="destination_number" expression="^(10[01][0-9])$"&gt;<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &lt;action application="set" data="dialed_extension=$1"/&gt;<br>

&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &lt;action application="export" data="dialed_extension=$1"/&gt;<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &lt;action application="bind_meta_app" data="2 b s record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/&gt;<br>

&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &lt;action application="bind_meta_app" data="3 b s execute_extension::cf XML features"/&gt;--&gt;<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &lt;action application="set" data="ringback=${us-ring}"/&gt;<br>

&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &lt;action application="set" data="transfer_ringback=$${hold_music}"/&gt;<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &lt;action application="set" data="call_timeout=30"/&gt;<br>&nbsp; &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &lt;action application="set" data="hangup_after_bridge=true"/&gt;<br>

&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &lt;action application="set" data="continue_on_fail=true"/&gt;<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &lt;action application="hash" data="insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}"/&gt;<br>

&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &lt;action application="hash" data="insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}"/&gt;<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &lt;action application="set" data="called_party_callgroup=${user_data(${dialed_extension} <at> ${domain_name} var callgroup)}"/&gt;<br>

&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &lt;action application="hash" data="insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}"/&gt;<div>
<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &lt;action application="bridge" data="{ignore_early_media=true}user/${dialed_extension} <at> ${domain_name}"/&gt;<br>
</div>
&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &lt;action application="sleep" data="1000"/&gt;<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &lt;action application="voicemail" data="default ${domain_name} ${dialed_extension}"/&gt;<br>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; &lt;/condition&gt;<br>

&nbsp;&nbsp;&nbsp; &lt;/extension&gt;<br><br><br>Please help me for this issue.<div>
<br><br clear="all">Thanks,<br>Max Alex<br>Voip Developer<br><br><br><br>
</div>
<div>
<div></div>
<div>
<div class="gmail_quote">On Fri, Jun 24, 2011 at 11:38 AM, Michael Collins <span dir="ltr">&lt;<a href="mailto:msc@..." target="_blank">msc@...</a>&gt;</span> wrote:<br><blockquote class="gmail_quote">
Are you using the default dialplan? I think you might just need to ignore early media on your bridge app. If you are using the default.xml file then locate "Local_Extension" and the bridge line:<br><br>&nbsp;&nbsp;&nbsp; &lt;action application="bridge" data="user/${dialed_extension} <at> ${domain_name}"/&gt;<br><br>Change it to this, then try again:<br><br>&nbsp;&nbsp;&nbsp; &lt;action application="bridge" data="{ignore_early_media=true}user/${dialed_extension} <at> ${domain_name}"/&gt;<br><br>If I understand correctly, the "symptom" you are experiencing is the normal operation of the bridge app (and it's cousin, the originate API command). When the b-leg sends back media, including ringing, the bridge (or the originate) will be considered "successful," and in the case of bridge, the b-leg's audio (the early media) will be connected to the a-leg. If you set ignore_early_media=true then you are explicitly telling the bridge app that you only want to connect the b-leg to the a-leg if the b-leg actually answers.<br><br>I hope that made sense...<br><br>-MC<div>
<div></div>
<div>
<br><br><br><div class="gmail_quote">On Thu, Jun 23, 2011 at 9:32 PM, Max Alex <span dir="ltr">&lt;<a href="mailto:max.asterisk <at> gmail.com" target="_blank">max.asterisk@...</a>&gt;</span> wrote:<br><blockquote class="gmail_quote">Hi,<br>Thanks for reply.<br>Current scenario is below.<br><br>PSTN call -&gt; sangoma device -&gt; freeswitch incoming context -&gt; routed to 1001 -&gt; ringing (Answered on cellphone)<br>

Here when it is routed to 1001 the call it is started ringing, but on phone that call is answered and hearding sound of ringing.<br><br>Required flow:<br>PSTN call -&gt; sangoma device -&gt; freeswitch incoming context -&gt; routed to 1001 -&gt; ringing (Ringing on cellphone)<br><br>I hope it is understable, the call should not be answered until 1001 answer it, right not when it is started ring it is answered on cell phone.<br>

that should not be happen as it is not answered yet.<br><br>Waiting for your reply.<div>
<br><br clear="all">Thanks,<br>Max Alex<br>Voip Developer<br><br><br><br>
</div>
<div>
<div></div>
<div>
<div class="gmail_quote">On Fri, Jun 24, 2011 at 5:48 AM, Michael Collins <span dir="ltr">&lt;<a href="mailto:msc@..." target="_blank">msc@...</a>&gt;</span> wrote:<br><blockquote class="gmail_quote">
I'm not sure I understand the problem. What is happening vs. what you believe should be happening?<br>-MC<div>
<div></div>
<div>
<br><br><div class="gmail_quote">On Thu, Jun 23, 2011 at 3:31 AM, Max Alex <span dir="ltr">&lt;<a href="mailto:max.asterisk@..." target="_blank">max.asterisk@...</a>&gt;</span> wrote:<br><blockquote class="gmail_quote">Hi,<br>Thanks for your reply.<br>Here is my configuration and log <br><a href="http://pastebin.freeswitch.org/16571" target="_blank">http://pastebin.freeswitch.org/16571</a><br><br>I am using A200 analog sangoma device with freeswitch, it is working fine but when it is routing call to 1001 then it is answered.<br>
Please provider your suggestions.<br><br clear="all">Thanks,<br>Max Alex<br>Voip Developer<div>
<div></div>
<div>
<br><br><br><br><div class="gmail_quote">On Wed, Jun 22, 2011 at 8:51 PM, Michael Collins <span dir="ltr">&lt;<a href="mailto:msc@..." target="_blank">msc@...</a>&gt;</span> wrote:<br><blockquote class="gmail_quote">

I thought the A200 was an analog card? Maybe I have my numbers mixed up...<div><br></div>
<div>Go ahead and collect a debug log of this call. It might help to have your configs posted as well. Use <a href="http://pastebin.freeswitch.org" target="_blank">pastebin.freeswitch.org</a>. See this wiki article for tips on how to collect information:</div>

<div><a href="http://wiki.freeswitch.org/wiki/Reporting_Bugs" target="_blank">http://wiki.freeswitch.org/wiki/Reporting_Bugs</a></div>
<div><br></div>
<div>-MC<br><br><div class="gmail_quote">
<div>
<div></div>
<div>
On Wed, Jun 22, 2011 at 3:23 AM, Max Alex <span dir="ltr">&lt;<a href="mailto:max.asterisk@..." target="_blank">max.asterisk@...</a>&gt;</span> wrote:<br>
</div>
</div>
<blockquote class="gmail_quote">
<div>
<div></div>
<div>Hi,<br>I have installed freeswitch latest version with sangoma card A200 as well,<br>

I have installed and configured freetdm module with wanpipe drivers for freeswitch,<br>
We are properly receiving the inbound calls in public context and then we are routing that call to 1001 extension,<br>
it is properly routing to 1001 as well, but we have one issue while routing on 1001.<br><br>Here is the issue description.<br>I am calling from my cell phone to that DID number of pri line, and then it will start ringing on 1001 extension,<br>

When 1001 extension start ringing the call is answered on my cell phone,<br>it is something like freeswitch preanswer or autoanswer the call, how can i stop this answer call when it is ringing on 1001 extension,<br>Waiting for good reply.<br><br clear="all">Thanks,<br>Max Alex<br><br><br><br>
</div>
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</blockquote>
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<br>
</div>
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</blockquote>
</div>
<br>
</div>
</div>
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</blockquote>
</div>
<br>
</div>
</div>
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</blockquote>
</div>
<br>
</div>
</div>
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</blockquote>
</div>
<br><div>
</div>
</div>
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ankIT WALiA | 1 Jul 2011 06:26
Picon

Re: (no subject)

Hi Sharad,

Appreciating your efforts, I have few questions.

  1. Can I schedule the reminders at specific time?
  2. The reminder calls will be IVR and these IVRs Menu will have to be configured in FreeSwitch XML or any GUI Fusionpbx, Freepbx or scripts.
  3. As per the configuration you have used, if we need to send several reminder calls at let us say on 100 mobile numbers, how is the performance?
  4. Last but not the least, under what license, your application is available?
Thanks
Ankit

On Thu, Jun 30, 2011 at 10:05 AM, sharad garg <sharad2710-PkbjNfxxIARBDgjK7y7TUQ@public.gmane.org> wrote:
Hi to all,
 
This is just let to know this forum that we have completed the work on Alarm Service. All this is done using our great Freeswitch only.

The GUI also is designed for the same.

You can have a view at the attached PPT for the details.

Thanks to Freeswitch developers & community off course who helped us in this project directly or indirectly.

Regards
Sharad



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--
Life is like a rose its upto u feel it as its fragrance or thorns
<div>
<p>Hi Sharad,<br><br>Appreciating your efforts, I have few questions.<br><br></p>
<ol>
<li>Can I schedule the reminders at specific time?</li>
<li>The reminder calls will be IVR and these IVRs Menu will have to be configured in FreeSwitch XML or any GUI Fusionpbx, Freepbx or scripts.</li>
<li>As per the configuration you have used, if we need to send several reminder calls at let us say on 100 mobile numbers, how is the performance?</li>
<li>Last but not the least, under what license, your application is available? <br>
</li>
</ol>Thanks<br>Ankit<br><br><div class="gmail_quote">On Thu, Jun 30, 2011 at 10:05 AM, sharad garg <span dir="ltr">&lt;<a href="mailto:sharad2710@...">sharad2710@...</a>&gt;</span> wrote:<br><blockquote class="gmail_quote">

<div><div dir="ltr">
<div>Hi to all,</div>
<div>&nbsp;</div>
<div>This is just let to know this forum that we have completed the work on 
Alarm Service. All this is done using our great Freeswitch only.<br><br>The GUI also is designed for the same.<br><br>You can have a view at the attached PPT for the details.<br><br>Thanks to Freeswitch developers &amp; community off course who helped us in this project directly or indirectly.<br><br>Regards<br>Sharad<br><br><br>
</div> 		 	   		  </div></div>
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<br><br clear="all"><br>-- <br>Life is like a rose its upto u feel it as its fragrance or thorns<br>
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Bryan Smart | 1 Jul 2011 07:13

Playing multiple files simultaneously

Is it possible for Freeswitch to play more than one file to a channel at a time? What I've seen and tried from
the dialplan and scripts either queues files to play, or will stop a currently playing file so that the
newly requested file will play. This also seems to be the case in conferences. When I send multiple play
commands to conferences, the files are queued.

As for how this might be used, think of an IVR that plays queued prompts, yet continuously plays looping
music or a Shoutcast stream in the background. I also want to be able to play short cue tones that start at the
same time as a prompt (don't want to pre-mix them in to a single file, though).

Is this currently possible through any means? Perhaps with the event socket?

Bryan

David Ponzone | 1 Jul 2011 07:39
Picon

Re: t38-passthru sofia param

You need to enable t38-passthru when you want FS to forward the T38 reinvite from a leg to anothern.

David Ponzone  Direction Technique
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26

Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur.




Le 30/06/2011 à 21:25, Spencer Thomason a écrit :

Just a quick follow up.  The fact that the reINVITE wasn't being passed correctly across the bridge was pilot error.. I had enable-soa=false on the profile.  I am however a little confused as when to use the t38-passthru parameter.    Is it needed even when doing bypass media or proxy media?

On Jun 30, 2011, at 11:40 AM, Spencer Thomason wrote:

Hello all,
I have a few questions regarding the t38-passthru sofia param.  One of the ways we are using FreeSWITCH is as a simple B2BUA in our SBCs.  The topology looks like this:

                                           Freeswitch
                                              /\          \/
FS and Asterisk boxen -> OpenSIPS -> PSTN

The Freeswitch instance is setup with inbound late negotiation and bypass media = true and the dialplan consists of a very simple bridge statement that sends a call back to the proxy.  

The problem is that when someone sends a fax and the GW send a t.38 reINVITE, Freeswitch is sending the original SDP (PCMU in this case) back to Leg A and the fax then fails using PCMU.  Is this parameter needed when bypass media is already enabled?

Thanks,
Spencer



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<div>You need to enable t38-passthru when you want FS to forward the T38 reinvite from a leg to anothern.<div>
<br><div>
<span class="Apple-style-span"><span class="Apple-style-span"><div>
<div>David Ponzone &nbsp;<span class="Apple-style-span">Direction Technique</span>
</div>
<div><span class="Apple-style-span">email: <a href="mailto:david.ponzone@...">david.ponzone@...</a></span></div>
<div><span class="Apple-style-span">tel: &nbsp; &nbsp; &nbsp;01 74 03 18 97</span></div>
<div><span class="Apple-style-span">gsm: &nbsp; 06 66 98 76 34</span></div>
<div><br></div>
<div>Service Client<span class="Apple-converted-space">&nbsp;</span>IPeva</div>
<div><span class="Apple-style-span"><div><span class="Apple-style-span">tel: &nbsp; &nbsp; &nbsp;0811 46 26 26</span></div>
<div><span class="Apple-style-span"><div>
<span><a href="BLOCKED::http://www.ipeva.fr/">www.ipeva.fr</a></span><span>&nbsp; -&nbsp; &nbsp;<span><a href="BLOCKED::http://www.ipeva-studio.com/">www.ipeva-studio.com</a></span></span>
</div>
<div><span class="Apple-style-span"><br></span></div>
<div><span class="Apple-style-span"><div>Ce message et toutes les pi&egrave;ces jointes sont confidentiels et &eacute;tablis &agrave; l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris&eacute;e est interdite. Tout message &eacute;lectronique est susceptible d'alt&eacute;ration.&nbsp;IPeva&nbsp;d&eacute;cline toute responsabilit&eacute; au titre de ce message s'il a &eacute;t&eacute; alt&eacute;r&eacute;, d&eacute;form&eacute; ou falsifi&eacute;. Si vous n'&ecirc;tes pas destinataire de ce message, merci de le d&eacute;truire imm&eacute;diatement et d'avertir l'exp&eacute;diteur.</div>
<div><br></div></span></div></span></div></span></div>
</div></span><br class="Apple-interchange-newline"></span><br class="Apple-interchange-newline">
</div>
<br><div>
<div>Le 30/06/2011 &agrave; 21:25, Spencer Thomason a &eacute;crit :</div>
<br class="Apple-interchange-newline"><blockquote type="cite"><div>Just a quick follow up. &nbsp;The fact that the reINVITE wasn't being passed correctly across the bridge was pilot error.. I had enable-soa=false on the profile. &nbsp;I am however a little confused as when to use the t38-passthru parameter. &nbsp;&nbsp;&nbsp;Is it needed even when doing bypass media or proxy media?<br><br>On Jun 30, 2011, at 11:40 AM, Spencer Thomason wrote:<br><br><blockquote type="cite">Hello all,<br>
</blockquote>
<blockquote type="cite">I have a few questions regarding the t38-passthru sofia param. &nbsp;One of the ways we are using FreeSWITCH is as a simple B2BUA in our SBCs. &nbsp;The topology looks like this:<br>
</blockquote>
<blockquote type="cite"><br></blockquote>
<blockquote type="cite"> &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;Freeswitch<br>
</blockquote>
<blockquote type="cite"> &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;/\ &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;\/<br>
</blockquote>
<blockquote type="cite">FS and Asterisk boxen -&gt; OpenSIPS -&gt; PSTN<br>
</blockquote>
<blockquote type="cite"><br></blockquote>
<blockquote type="cite">The Freeswitch instance is setup with inbound late negotiation and bypass media = true and the dialplan consists of a very simple bridge statement that sends a call back to the proxy. &nbsp;<br>
</blockquote>
<blockquote type="cite"><br></blockquote>
<blockquote type="cite">The problem is that when someone sends a fax and the GW send a t.38 reINVITE, Freeswitch is sending the original SDP (PCMU in this case) back to Leg A and the fax then fails using PCMU. &nbsp;Is this parameter needed when bypass media is already enabled?<br>
</blockquote>
<blockquote type="cite"><br></blockquote>
<blockquote type="cite">Thanks,<br>
</blockquote>
<blockquote type="cite">Spencer<br>
</blockquote>
<blockquote type="cite"><br></blockquote>
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