<div dir="ltr">An update on why this was happening so others will know:<div><br></div><div>I disabled the Sip ALG on my router and everything now seems to be working fine.</div><div><br></div><div>So, hold music will not work on some Polycom phones behind NAT with ALG connecting to remote Freeswitch servers.<br>
<div><br></div><div>Thanks,</div><div>Lyle Pratt<br><div class="gmail_extra"><br><br><div class="gmail_quote">On Thu, Feb 6, 2014 at 7:04 AM, <span dir="ltr"><<a href="mailto:freeswitch-users-request@lists.freeswitch.org" target="_blank">freeswitch-users-request@lists.freeswitch.org</a>></span> wrote:<br>
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<br>Today's Topics:<br>
<br>
1. Re: routing incoming SMS? (Privus P)<br>
2. Re: Filler Music in Call Recordings (Michael Jerris)<br>
3. Re: Hold Music not Playing when pressing Hold on Polycom<br>
Phone (Michael Jerris)<br>
<br><br>---------- Forwarded message ----------<br>From: Privus P <<a href="mailto:privus007@gmail.com" target="_blank">privus007@gmail.com</a>><br>To: FreeSWITCH Users Help <<a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a>><br>
Cc: <br>Date: Thu, 6 Feb 2014 12:58:35 +0000<br>Subject: Re: [Freeswitch-users] routing incoming SMS?<br><div dir="ltr"><div><div>Hi Donny,<br><br></div>Perhaps you could share your javascript and lua script so that we could try to figure out together what isn't working.<br>
</div>Judging by the apparent lack of response to this thread, it seems to be the best way forward to try and resolve this.<br>
</div><div class="gmail_extra"><br><br><div class="gmail_quote">On Thu, Feb 6, 2014 at 6:50 AM, Donny Hardyanto <span dir="ltr"><<a href="mailto:hardyanto.donny@gmail.com" target="_blank">hardyanto.donny@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><p dir="ltr">Hi,</p>
<p dir="ltr">I have similar problem. I use gateway to GSM provider that support SIP simple . So far i can send sms from fs to gateway and receive sms in fs from gateway. But I was lost how to send the incoming sms from gateway to client. </p>
<p dir="ltr">Also my client support sip simple also. When the sms from client arrive at fs, i dont know how to forward them to gateway. </p>
<p dir="ltr">I use Javascript heavily and tried to produce the same thing as lua script in javascript but it always fail. The event fire command does not produce any thing. </p>
<p dir="ltr">Can any one show us or point us how to forward sms?</p>
<p dir="ltr">Dinny</p><div><div>
<div class="gmail_quote">On Feb 4, 2014 8:06 PM, "Peter Villeneuve" <<a href="mailto:petervnv1@gmail.com" target="_blank">petervnv1@gmail.com</a>> wrote:<br type="attribution"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div dir="ltr">Hi,<div><br></div><div>Thanks for helping out. Indeed I would love to try out a lua script to handle SMS.</div><div>Can anyone point to a sample script that I can study and play with?</div><div><br></div><div>
Any help is much appreciated. I'm still stuck with SMS hitting FS but not being forwarded to the peer....</div><div><br></div><div>Thanks,</div><div>Peter</div></div><div class="gmail_extra"><br><br><div class="gmail_quote">
On Sun, Feb 2, 2014 at 5:46 PM, Privus P <span dir="ltr"><<a href="mailto:privus007@gmail.com" target="_blank">privus007@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div dir="ltr"><div><div><div>Hi Peter,<br><br></div>I'm not really sure why your SMS isn't being routed correctly.<br></div>I'm sure others in this list have a lot more experience and can help you out, but have you thought of using a lua script to handle SMS instead of relying on the send action?<br>
<br></div><br></div><div class="gmail_extra"><br><br><div class="gmail_quote"><div><div>On Sat, Feb 1, 2014 at 2:00 PM, Peter Villeneuve <span dir="ltr"><<a href="mailto:petervnv1@gmail.com" target="_blank">petervnv1@gmail.com</a>></span> wrote:<br>
</div></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div><div dir="ltr"><div>Hi,</div><div><br></div><div>I'm experimenting with GSMopen and I'm having trouble doing something which is likely pretty simple.</div>
<div>I have enabled mod_sms and GSMopen in FS. I can send SIP simple messages between 2 registered peers OK.</div>
<div>I can see that incoming GSM SMS messages arrive in FS as expected, but I can't seem to get them converted into SIP SIMPLE format and routed to the peer (1000).</div><div>Once they arrive in FS, I can see in the log:</div>
<div><br></div><div>2014-01-30 20:00:54.840999 [INFO] mod_sms.c:336 Processing text message +4412398746->gsm01 in context default</div><div>Chatplan: gsm01 parsing [default->basic p2p] continue=true</div><div>Chatplan: gsm01 Regex (PASS) [basic p2p] to(gsm01) =~ /^(.*)$/ break=on-false</div>
<div>Chatplan: gsm01 Action send() </div><div><br></div><div><br></div><div>My chatplan has:</div><div><br></div><div><?xml version="1.0" encoding="utf-8"?></div><div><include></div><div> <context name="default"></div>
<div><br></div><div> <extension name="basic p2p" continue="true"></div><div> <condition field="to" expression="^(.*)$"></div><div><span style="white-space:pre-wrap"> </span><!--<span style="white-space:pre-wrap"> </span><action application="lua" data="test.lua"/> --></div>
<div><span style="white-space:pre-wrap"> </span> <action application="send"/></div><div> </condition></div><div> </extension></div><div><br></div><div> </context></div><div>
</include></div><div><br></div><div>And in my gsmopen.conf.xml:</div><div><br></div><div><configuration name="gsmopen.conf" description="GSMopen Configuration"></div><div> <global_settings></div>
<div> <param name="debug" value="8"/></div><div> <param name="dialplan" value="XML"/></div><div> <param name="context" value="default"/></div>
<div> <param name="hold-music" value="$${moh_uri}"/></div><div> <param name="destination" value="1000"/></div><div> </global_settings></div><div><br></div>
<div>
<br></div><div>I believe I'm doing something wrong in the chatplan. Is there any transfer action like in the regular XML dialplan?</div><div>I tried creating in the default dialplan the following, hoping that it would be similar to sofia calls and transfer the incoming SMS to peer 1000, but no dice.</div>
<div><br></div><div><include></div><div> <extension name="sms_inbound"></div><div> <condition field="destination_number" expression="^(gsm01)$"></div><div> <span style="white-space:pre-wrap"> </span><action application="transfer" data="1000 XML default"/></div>
<div> </condition></div><div> </extension></div><div></include></div><div><br></div><div>So, basically, how can I route the incoming SMS to peer 1000?</div><div><br></div><div>Thanks</div></div>
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<br><br>---------- Forwarded message ----------<br>From: Michael Jerris <<a href="mailto:mike@jerris.com" target="_blank">mike@jerris.com</a>><br>To: FreeSWITCH Users Help <<a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a>><br>
Cc: <br>Date: Thu, 6 Feb 2014 08:02:03 -0500<br>Subject: Re: [Freeswitch-users] Filler Music in Call Recordings<br><div style="word-wrap:break-word">You would have to make a media bug similar to the one for record session, that has additional api commands to target the session by uuid that changes a flag in a private struct of weather the recording is paused or not, and then inside the callback for the recording, trigger different behavior when its paused. As there is no audio being muxed, there is no reason at all to use mod_conference.<div>
<br><div><div>On Feb 5, 2014, at 7:42 PM, JP <<a href="mailto:jaykris@gmail.com" target="_blank">jaykris@gmail.com</a>> wrote:</div><br><blockquote type="cite"><div dir="ltr"><p class="MsoNormal">I'm trying to implement a feature that allows a user to
selectively pause and resume a call recording via FreeSWITCH's event socket
interface. The audio file should contain filler music for the duration of
each pause operation. Is there any simple straight forward way to
accomplish this? I would prefer avoiding the overhead incurred by using
the conferencing module for this if possible. </p></div></blockquote></div><br></div></div><br><br>---------- Forwarded message ----------<br>From: Michael Jerris <<a href="mailto:mike@jerris.com" target="_blank">mike@jerris.com</a>><br>
To: FreeSWITCH Users Help <<a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a>><br>Cc: <br>Date: Thu, 6 Feb 2014 08:04:04 -0500<br>Subject: Re: [Freeswitch-users] Hold Music not Playing when pressing Hold on Polycom Phone<br>
<div style="word-wrap:break-word">I don't see the phone actually sending the hold signal there, try turning on sip trace and confirming what the phone is actually sending. Compare that to your dev environment where it works.<div>
<br></div><div>Mike</div><div><br><div><div>On Feb 6, 2014, at 12:11 AM, Lyle Pratt <<a href="mailto:lylepratt@gmail.com" target="_blank">lylepratt@gmail.com</a>> wrote:</div><br><blockquote type="cite"><div dir="ltr">
Hi Guys,<div><br></div><div>When I press the HOLD button (or start to transfer someone, join a conference, etc) the configured hold music will not play.<br><div><br></div><div>I previously had this working in production, but all of the sudden it is no longer working I can't figure out why. It works fine in my development environment on my local network, but not when I'm registering to Public IPs. I first thought it was a Polycom specific NAT issue, but then I tried it with a different phone (Cisco) from a different network with no success either.</div>
<div><br></div><div>Here is a Pastebin of the Freeswitch console logs when I press the hold button on my device and no hold music occurs.</div></div><div><a href="http://pastebin.com/p6xtwyz7" target="_blank">http://pastebin.com/p6xtwyz7</a><br>
</div><div><br></div><div>Does anyone have any ideas on how on earth I messed this up? Its the last piece of the puzzle I have in completely getting this working! Any help appreciated</div></div></blockquote></div></div>
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