<p dir="ltr">Other calls working fine, in conference also both sides can hear conference instruction messages.</p>
<div class="gmail_quote">On 17 Dec 2014 19:15, <<a href="mailto:freeswitch-users-request@lists.freeswitch.org">freeswitch-users-request@lists.freeswitch.org</a>> wrote:<br type="attribution"><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Send FreeSWITCH-users mailing list submissions to<br>
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<br>Today's Topics:<br>
<br>
1. Re: Conference no sound (?talo Rossi)<br>
2. Re: How to play a stream other than mp3 with mod_shout.<br>
(Aqs Younas)<br>
3. Re: Issue with Freeswitch Behind nat (Frederick Pruneau)<br>
<br><br>---------- Forwarded message ----------<br>From: "Ítalo Rossi" <<a href="mailto:italorossib@gmail.com">italorossib@gmail.com</a>><br>To: FreeSWITCH Users Help <<a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a>><br>Cc: <br>Date: Wed, 17 Dec 2014 09:41:08 -0300<br>Subject: Re: [Freeswitch-users] Conference no sound<br><div dir="ltr">Is there audio when you call the other user directly, without the conference?</div><div class="gmail_extra"><br><div class="gmail_quote">On Wed, Dec 17, 2014 at 8:30 AM, sukitha jayasinghe <span dir="ltr"><<a href="mailto:sukithaj@gmail.com" target="_blank">sukithaj@gmail.com</a>></span> wrote:<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">Hi All,<div><br><div>I have configured freeswitch server with conference feature enabled, When I call the conference number, i can hear hold music. when second call dial the conference number hold music stops, but no voice between legs. what would be the reason for this.</div><div><br></div><div>Best Regards,</div></div><div>Sukitha</div></div>
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<br><br>---------- Forwarded message ----------<br>From: Aqs Younas <<a href="mailto:aqsyounas@gmail.com">aqsyounas@gmail.com</a>><br>To: FreeSWITCH Users Help <<a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a>><br>Cc: <br>Date: Wed, 17 Dec 2014 18:34:24 +0500<br>Subject: Re: [Freeswitch-users] How to play a stream other than mp3 with mod_shout.<br><div dir="ltr"><div><div><div><div><div>Hi, <span name="Moishe Grunstein">Moishe Grunstein<br><br></span></div><span name="Moishe Grunstein">I am able to successfully configure mod_vlc, but when i play i stream with vlc i hear nothing. Like dead call.<br><br></span></div><span name="Moishe Grunstein">Here is my default.xml<br><br><include><br> <context name="default"><br> <extension name="Testing"><br> <condition field="destination_number" expression="^(\+1|1)?\d+$"><br> <action application="answer"/><br> <action application="log" data="Call in progress"/><br> <action application="playback" data="vlc://<a href="http://s9.voscast.com:7584" target="_blank">http://s9.voscast.com:7584</a>"/><br> </condition><br> </extension><br> </context><br></include><br><br></span></div><span name="Moishe Grunstein">When call is answered and i hangup the call from my softphone, it does not actually ends on freeswitch side.<br></span></div><span name="Moishe Grunstein">Also when i type<br><br>freeswitch@internal> hupall<br>+OK hangup all channels with cause MANAGER_REQUEST<br><br></span></div><span name="Moishe Grunstein">But <br>freeswitch@internal> show calls count<br>2 total.<br></span><div><div><span name="Moishe Grunstein"><br></span><div><div><span name="Moishe Grunstein">Call doesn't terminate. When i play the stream with mod_shout everything works fine.<br></span></div><div><span name="Moishe Grunstein"><br></span></div><div><span name="Moishe Grunstein">My OS is Debian 7.6.<br><br></span></div><div><span name="Moishe Grunstein">Can you please provide my with example how to play stream with vlc. Or anything that i am doing wrong.<br><br></span></div><div><span name="Moishe Grunstein">Best Regards.<br></span></div><div><span name="Moishe Grunstein"><br></span></div></div></div></div></div><div class="gmail_extra"><br><div class="gmail_quote">On 15 December 2014 at 17:03, Aqs Younas <span dir="ltr"><<a href="mailto:aqsyounas@gmail.com" target="_blank">aqsyounas@gmail.com</a>></span> wrote:<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><div>Thanks for your reply. I will try that too.<br></div>Many thanks for your valuable replies.<br></div><div class="gmail_extra"><br><div class="gmail_quote"><div><div>On 15 December 2014 at 00:37, Danny Gershman <span dir="ltr"><<a href="mailto:danny.gershman@gmail.com" target="_blank">danny.gershman@gmail.com</a>></span> wrote:</div></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div>Also mod_rtmp lets you play from an FMS server.<div><div><br><br>On Friday, December 12, 2014, Aqs Younas <<a href="mailto:aqsyounas@gmail.com" target="_blank">aqsyounas@gmail.com</a>> 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>Hi, All<br><br></div>How can i play a live stream other than mp3 with mod_shout or any module.? Is there any way to buffer the stream before playing it with mod_shout.<br><br><br></div><div>Currently i have a list of streams and when i play them with mod_shout some work fine but others give (time out) error.<br><br></div><div>How can i play mostly stream in freeswitch?<br></div><div><br></div>Thanks<br></div>
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<br><br>---------- Forwarded message ----------<br>From: Frederick Pruneau <<a href="mailto:frederick@targointernet.com">frederick@targointernet.com</a>><br>To: FreeSWITCH Users Help <<a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a>><br>Cc: <br>Date: Wed, 17 Dec 2014 08:44:10 -0500<br>Subject: Re: [Freeswitch-users] Issue with Freeswitch Behind nat<br><div dir="ltr">Sorry for this noob question but how can I see sip traffic? Is there a specific command to show this? Is it what we find in freeswitch.log? If so, I attached my log file in my first post.<div><br></div><div>Thanks for you help</div><div class="gmail_extra"><br><div class="gmail_quote">2014-12-16 15:44 GMT-05:00 Brian West <span dir="ltr"><<a href="mailto:brian@freeswitch.org" target="_blank">brian@freeswitch.org</a>></span>:<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">have you looked at the signalling? What does the sip traffic show? Please pastebin that.</div><div><div><div class="gmail_extra"><br><div class="gmail_quote">On Tue, Dec 16, 2014 at 2:37 PM, Frederick Pruneau <span dir="ltr"><<a href="mailto:frederick@targointernet.com" target="_blank">frederick@targointernet.com</a>></span> wrote:<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">Same problem...<br><div class="gmail_extra"><br><div class="gmail_quote">2014-12-16 13:55 GMT-05:00 Brian West <span dir="ltr"><<a href="mailto:brian@freeswitch.org" target="_blank">brian@freeswitch.org</a>></span>:<div><div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">Guessing you don't have UPNP or NAT-PMP on your network, there for that won't work, <div><br></div><div>ext-sip-ip=autonat:x.x.x.x</div><div>ext-rtp-ip=autonat:x.x.x.x</div><div><br></div><div>Set local-network-ac to rfc1918.auto</div></div><div><div><div class="gmail_extra"><br><div class="gmail_quote">On Tue, Dec 16, 2014 at 12:15 PM, Support Technique <span dir="ltr"><<a href="mailto:support@targointernet.com" target="_blank">support@targointernet.com</a>></span> wrote:<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr"><div><param name="ext-rtp-ip" value="auto-nat"/></div><div> <param name="ext-sip-ip" value="auto-nat"/></div><div><param name="local-network-acl" value="localnet.auto"/><br></div></div><div class="gmail_extra"><br><div class="gmail_quote">2014-12-16 12:25 GMT-05:00 Brian West <span dir="ltr"><<a href="mailto:brian@freeswitch.org" target="_blank">brian@freeswitch.org</a>></span>:<div><div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">On the system behind nat what do you have ext-rtp-ip, ext-sip-ip and local-network-acl set to?</div><div class="gmail_extra"><br><div class="gmail_quote"><div><div>On Tue, Dec 16, 2014 at 10:41 AM, Frederick Pruneau <span dir="ltr"><<a href="mailto:frederick@targointernet.com" target="_blank">frederick@targointernet.com</a>></span> wrote:</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">Hi guys,<div><br></div><div>We have an issue with one freeswitch server behind nat. We have a setup like this:</div><div><br></div><div>-One master Freeswitch server</div><div><br></div><div>-One freeswitch server connected to the master (Public IP) - Server A</div><div><br></div><div>-One freeswitch server connected to the master (behind nat) - Server B</div><div><br></div><div>If server A call server B, nothing happens. There is no sound. After 30 sec, it times out. We have done a tcpdump. From server A to master packets are ok. From Master to server B, we have seen that there is no source and no destination ports for sip invite.</div><div><br></div><div>If we use our cellphone and we call server B, there is no problem.</div><div><br></div><div>I have attached the failed call pcap file and freeswitch's log file so you can take a look at them.</div><div><br></div><div>Master = Freeswitch v1.4.13</div><div>Server A = Freeswitch v.1.4.13</div><div>Server B = Freeswitch v.1.4.14 (Updated to latest release since we have issues with this server)</div><div><br></div><div>Thanks in advance.</div><div><br></div><div>PS: The failed call is from <a href="tel:514-448-0773" value="+15144480773" target="_blank">514-448-0773</a>.<br clear="all"><div> </div>
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<p><font face="courier new, monospace"><b><i><font size="4">Brian West</font></i></b><br><span style="font-size:x-small"><a href="mailto:brian@freeswitch.org" target="_blank">brian@freeswitch.org</a></span></font></p>
<p><font size="1" face="courier new, monospace"><img src="http://billing.freeswitch.org/templates/default/img/whmcslogo.png"><br></font></p><p><font face="courier new, monospace"><b><i>Twitter: @FreeSWITCH , @briankwest</i></b><br><a href="http://www.freeswitchbook.com" target="_blank">http://www.freeswitchbook.com</a><br><a href="http://www.freeswitchcookbook.com" target="_blank">http://www.freeswitchcookbook.com</a></font></p>
<p><font face="courier new, monospace"><b>T:</b><a href="tel:%2B19184209001" value="+19184209001" target="_blank">+19184209001</a> | <b>F:</b><a href="tel:%2B19184209002" value="+19184209002" target="_blank">+19184209002</a> | <b>M:</b>+1918424WEST (9378)<br><b>iNUM:</b>+883 5100 1420 9001 | <b>ISN:</b>410*543 | <b>Skype:</b>briankwest</font></p></div></div>
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