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Today's Topics:

   1. Re: How to play a stream other than mp3 with      mod_shout.
      (Brian West)
   2. Re: Conference no sound (Brian West)
   3. Re: Issue with Freeswitch Behind nat (Brian West)


---------- Forwarded message ----------
From: Brian West <brian@freeswitch.org>
To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org>
Cc: 
Date: Wed, 17 Dec 2014 08:26:14 -0600
Subject: Re: [Freeswitch-users] How to play a stream other than mp3 with mod_shout.

Don't think mod_rtmp can actually do this, from loading it:

2014-12-17 08:25:39.983446 [NOTICE] switch_loadable_module.c:149 Adding Endpoint 'rtmp'

2014-12-17 08:25:39.983446 [NOTICE] switch_loadable_module.c:315 Adding API Function 'rtmp'

2014-12-17 08:25:39.983446 [NOTICE] switch_loadable_module.c:315 Adding API Function 'rtmp_contact'

This is all that gets registered.

On Sun, Dec 14, 2014 at 1:37 PM, 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, All

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.

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: Brian West <brian@freeswitch.org>
To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org>
Cc: 
Date: Wed, 17 Dec 2014 08:26:51 -0600
Subject: Re: [Freeswitch-users] Conference no sound

Please press 0 to verify that you're not hitting a bug that was recently fix, you also didn't outline the rev you're running.  Have you tried master?

On Wed, Dec 17, 2014 at 5:30 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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---------- Forwarded message ----------
From: Brian West <brian@freeswitch.org>
To: FreeSWITCH Users Help <freeswitch-users@lists.freeswitch.org>
Cc: 
Date: Wed, 17 Dec 2014 08:27:06 -0600
Subject: Re: [Freeswitch-users] Issue with Freeswitch Behind nat

sofia global siptrace on

from fs_cli

On Wed, Dec 17, 2014 at 7:44 AM, Frederick Pruneau <frederick@targointernet.com> wrote:

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.

Thanks for you help

2014-12-16 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.x

ext-rtp-ip=autonat:x.x.x.x

Set local-network-ac to rfc1918.auto

On 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 B

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.

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 v1.4.13

Server A = Freeswitch v.1.4.13

Server B = Freeswitch v.1.4.14 (Updated to latest release since we have issues with this server)

Thanks in advance.

PS: The failed call is from 514-448-0773.

 

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T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378)
iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest

_________________________________________________________________________
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_________________________________________________________________________
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--

Brian West
brian@freeswitch.org

Twitter: @FreeSWITCH , @briankwest
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378)
iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest

_________________________________________________________________________
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_________________________________________________________________________
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--

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Twitter: @FreeSWITCH , @briankwest
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378)
iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
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http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
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_________________________________________________________________________
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Official FreeSWITCH Sites
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--

Brian West
brian@freeswitch.org

Twitter: @FreeSWITCH , @briankwest
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378)
iNUM:+883 5100 1420 9001 | ISN:410*543 | Skype:briankwest

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