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Today's Topics:
1. Re: Conference no sound (?talo Rossi)
2. Re: How to play a stream other than mp3 with mod_shout.
(Aqs Younas)
3. Re: Issue with Freeswitch Behind nat (Frederick Pruneau)
---------- Forwarded message ----------
From: "Ítalo Rossi" <italorossib@gmail.com>
To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org>
Cc:
Date: Wed, 17 Dec 2014 09:41:08 -0300
Subject: Re: [Freeswitch-users] Conference no soundIs there audio when you call the other user directly, without the conference?On Wed, Dec 17, 2014 at 8:30 AM, sukitha jayasinghe <sukithaj@gmail.com> wrote:Hi All,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.Best Regards,Sukitha
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http://www.freeswitch.org--Ítalo Rossi
---------- Forwarded message ----------
From: Aqs Younas <aqsyounas@gmail.com>
To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org>
Cc:
Date: Wed, 17 Dec 2014 18:34:24 +0500
Subject: Re: [Freeswitch-users] How to play a stream other than mp3 with mod_shout.ButAlso when i typeWhen call is answered and i hangup the call from my softphone, it does not actually ends on freeswitch side.Here is my default.xmlHi, Moishe GrunsteinI am able to successfully configure mod_vlc, but when i play i stream with vlc i hear nothing. Like dead call.
<include>
<context name="default">
<extension name="Testing">
<condition field="destination_number" expression="^(\+1|1)?\d+$">
<action application="answer"/>
<action application="log" data="Call in progress"/>
<action application="playback" data="vlc://http://s9.voscast.com:7584"/>
</condition>
</extension>
</context>
</include>
freeswitch@internal> hupall
+OK hangup all channels with cause MANAGER_REQUEST
freeswitch@internal> show calls count
2 total.Call doesn't terminate. When i play the stream with mod_shout everything works fine.My OS is Debian 7.6.Can you please provide my with example how to play stream with vlc. Or anything that i am doing wrong.Best Regards.On 15 December 2014 at 17:03, Aqs Younas <aqsyounas@gmail.com> wrote:Thanks for your reply. I will try that too.Many thanks for your valuable replies.On 15 December 2014 at 00:37, Danny Gershman <danny.gershman@gmail.com> wrote:_________________________________________________________________________Also mod_rtmp lets you play from an FMS server.
On Friday, December 12, 2014, Aqs Younas <aqsyounas@gmail.com> wrote:Hi, AllHow 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.Currently i have a list of streams and when i play them with mod_shout some work fine but others give (time out) error.How can i play mostly stream in freeswitch?Thanks
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---------- Forwarded message ----------
From: Frederick Pruneau <frederick@targointernet.com>
To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org>
Cc:
Date: Wed, 17 Dec 2014 08:44:10 -0500
Subject: Re: [Freeswitch-users] Issue with Freeswitch Behind natSorry 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.Thanks for you help2014-12-16 15:44 GMT-05:00 Brian West <brian@freeswitch.org>:have you looked at the signalling? What does the sip traffic show? Please pastebin that.On Tue, Dec 16, 2014 at 2:37 PM, Frederick Pruneau <frederick@targointernet.com> wrote:Same problem...2014-12-16 13:55 GMT-05:00 Brian West <brian@freeswitch.org>:Guessing you don't have UPNP or NAT-PMP on your network, there for that won't work,ext-sip-ip=autonat:x.x.x.xext-rtp-ip=autonat:x.x.x.xSet local-network-ac to rfc1918.autoOn Tue, Dec 16, 2014 at 12:15 PM, Support Technique <support@targointernet.com> wrote:<param name="ext-rtp-ip" value="auto-nat"/><param name="ext-sip-ip" value="auto-nat"/><param name="local-network-acl" value="localnet.auto"/>2014-12-16 12:25 GMT-05:00 Brian West <brian@freeswitch.org>:On the system behind nat what do you have ext-rtp-ip, ext-sip-ip and local-network-acl set to?On Tue, Dec 16, 2014 at 10:41 AM, Frederick Pruneau <frederick@targointernet.com> wrote:_________________________________________________________________________Hi guys,We have an issue with one freeswitch server behind nat. We have a setup like this:-One master Freeswitch server-One freeswitch server connected to the master (Public IP) - Server A-One freeswitch server connected to the master (behind nat) - Server BIf 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.If we use our cellphone and we call server B, there is no problem.I have attached the failed call pcap file and freeswitch's log file so you can take a look at them.Master = Freeswitch v1.4.13Server A = Freeswitch v.1.4.13Server B = Freeswitch v.1.4.14 (Updated to latest release since we have issues with this server)Thanks in advance.
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