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[Freeswitch-users] Recording and outbound rtp


 
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PostPosted: Tue Feb 24, 2009 1:04 pm    Post subject: [Freeswitch-users] Recording and outbound rtp Reply with quote

Hi,

I have a small javascript application that accepts a call, retrieves some dtmf digits and then records the call to an icecast server. This works great.

The problem I'm having is that when the call is being recorded freeswitch is no longer sending rtp packets back to the originating caller, in my case a Cisco 5300 with a bunch of T1 voice circuits in it. This makes sense, since no voice data back is being generated. Unfortunately my Cisco gear has rtp inactivity timers set up to hang up a call after 3 minutes of no incoming rtp packets, this is a global setting that cannot be configured for a single dial peer. Does anyone have a suggestion to generate rtp packets every once in a while? I tried setting comfort noise which did not seem to send anything. I could try playing a empty/short wav file every minute or so but the javascript call session.record is blocking, would a traditional javascript timer and callback to play a wav file be my best bet or is there a better approach? I'm using FreeSWITCH Version 1.0.trunk (12108M) on debian etch.

Thanks!
Dan-
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anthony.minessale at g...
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PostPosted: Tue Feb 24, 2009 2:17 pm    Post subject: [Freeswitch-users] Recording and outbound rtp Reply with quote

is it during a bridged call?


On Tue, Feb 24, 2009 at 11:49 AM, Dan <freeswitch-users@digitaldan.com (freeswitch-users@digitaldan.com)> wrote:
Quote:
Hi,

I have a small javascript application that accepts a call, retrieves some dtmf digits and then records the call to an icecast server. This works great.

The problem I'm having is that when the call is being recorded freeswitch is no longer sending rtp packets back to the originating caller, in my case a Cisco 5300 with a bunch of  T1 voice circuits in it.  This makes sense, since no voice data back is being generated.  Unfortunately my Cisco gear has rtp inactivity timers set up to hang up a call after 3 minutes of no incoming rtp packets, this is a global setting that cannot be configured for a single dial peer.  Does anyone have a suggestion to generate rtp packets every once in a while?  I tried setting comfort noise which did not seem to send anything.  I could try playing a empty/short wav file every minute or so but the javascript call session.record is blocking, would a traditional javascript timer and callback to play a wav file be my best bet or is there a better approach? I'm using FreeSWITCH Version 1.0.trunk (12108M) on debian etch.

Thanks!
Dan-




_______________________________________________
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




--
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale@hotmail.com ([email]MSN%3Aanthony_minessale@hotmail.com[/email])
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])
pstn:213-799-1400
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freeswitch-users at di...
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PostPosted: Tue Feb 24, 2009 3:13 pm    Post subject: [Freeswitch-users] Recording and outbound rtp Reply with quote

no, I'm matching the incoming sip call via the destination number in my public context and executing the javascript appliaction. This app directly answers the call and records it until the user hangs up.
D-

----- Original Message -----
From: "Anthony Minessale" <anthony.minessale@gmail.com>
To: freeswitch-users@lists.freeswitch.org
Sent: Tuesday, February 24, 2009 12:05:33 PM GMT -07:00 US/Canada Mountain
Subject: Re: [Freeswitch-users] Recording and outbound rtp

is it during a bridged call?


On Tue, Feb 24, 2009 at 11:49 AM, Dan <freeswitch-users@digitaldan.com (freeswitch-users@digitaldan.com)> wrote:
Quote:
Hi,

I have a small javascript application that accepts a call, retrieves some dtmf digits and then records the call to an icecast server. This works great.

The problem I'm having is that when the call is being recorded freeswitch is no longer sending rtp packets back to the originating caller, in my case a Cisco 5300 with a bunch of T1 voice circuits in it. This makes sense, since no voice data back is being generated. Unfortunately my Cisco gear has rtp inactivity timers set up to hang up a call after 3 minutes of no incoming rtp packets, this is a global setting that cannot be configured for a single dial peer. Does anyone have a suggestion to generate rtp packets every once in a while? I tried setting comfort noise which did not seem to send anything. I could try playing a empty/short wav file every minute or so but the javascript call session.record is blocking, would a traditional javascript timer and callback to play a wav file be my best bet or is there a better approach? I'm using FreeSWITCH Version 1.0.trunk (12108M) on debian etch.

Thanks!
Dan-




_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org (Freeswitch-users@lists.freeswitch.org)
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




--
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale@hotmail.com ([email]MSN%3Aanthony_minessale@hotmail.com[/email])
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])
pstn:213-799-1400

_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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anthony.minessale at g...
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PostPosted: Wed Feb 25, 2009 9:30 am    Post subject: [Freeswitch-users] Recording and outbound rtp Reply with quote

We would have to code in a feature to purposely write silence back during a recording that does not currently exist.
You could perhaps post it on the bounty section in jira.


On Tue, Feb 24, 2009 at 2:02 PM, <freeswitch-users@digitaldan.com (freeswitch-users@digitaldan.com)> wrote:
Quote:
no, I'm matching the incoming sip call via the destination number in my public context and executing the javascript appliaction.  This app directly answers the call and records it until the user hangs up.
D-


----- Original Message -----
From: "Anthony Minessale" <anthony.minessale@gmail.com (anthony.minessale@gmail.com)>
To: freeswitch-users@lists.freeswitch.org (freeswitch-users@lists.freeswitch.org)
Sent: Tuesday, February 24, 2009 12:05:33 PM GMT -07:00 US/Canada Mountain
Subject: Re: [Freeswitch-users] Recording and outbound rtp

is it during a bridged call?


On Tue, Feb 24, 2009 at 11:49 AM, Dan <freeswitch-users@digitaldan.com (freeswitch-users@digitaldan.com)> wrote:
Quote:
Hi,

I have a small javascript application that accepts a call, retrieves some dtmf digits and then records the call to an icecast server. This works great.

The problem I'm having is that when the call is being recorded freeswitch is no longer sending rtp packets back to the originating caller, in my case a Cisco 5300 with a bunch of  T1 voice circuits in it.  This makes sense, since no voice data back is being generated.  Unfortunately my Cisco gear has rtp inactivity timers set up to hang up a call after 3 minutes of no incoming rtp packets, this is a global setting that cannot be configured for a single dial peer.  Does anyone have a suggestion to generate rtp packets every once in a while?  I tried setting comfort noise which did not seem to send anything.  I could try playing a empty/short wav file every minute or so but the javascript call session.record is blocking, would a traditional javascript timer and callback to play a wav file be my best bet or is there a better approach? I'm using FreeSWITCH Version 1.0.trunk (12108M) on debian etch.

Thanks!
Dan-




_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org (Freeswitch-users@lists.freeswitch.org)
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




--
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale@hotmail.com ([email]MSN%3Aanthony_minessale@hotmail.com[/email])
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])
pstn:213-799-1400



_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org (Freeswitch-users@lists.freeswitch.org) http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org


_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org (Freeswitch-users@lists.freeswitch.org)
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




--
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale@hotmail.com ([email]MSN%3Aanthony_minessale@hotmail.com[/email])
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])
pstn:213-799-1400
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freeswitch-users at di...
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PostPosted: Wed Feb 25, 2009 11:10 am    Post subject: [Freeswitch-users] Recording and outbound rtp Reply with quote

Thanks, I will look around and see if I can come up with a solution. I'll post back here and on the wiki if I find one.

D-

----- Original Message -----
From: "Anthony Minessale" <anthony.minessale@gmail.com>
To: freeswitch-users@lists.freeswitch.org
Sent: Wednesday, February 25, 2009 7:19:50 AM GMT -07:00 US/Canada Mountain
Subject: Re: [Freeswitch-users] Recording and outbound rtp

We would have to code in a feature to purposely write silence back during a recording that does not currently exist.
You could perhaps post it on the bounty section in jira.


On Tue, Feb 24, 2009 at 2:02 PM, <freeswitch-users@digitaldan.com (freeswitch-users@digitaldan.com)> wrote:
Quote:
no, I'm matching the incoming sip call via the destination number in my public context and executing the javascript appliaction. This app directly answers the call and records it until the user hangs up.
D-


----- Original Message -----
From: "Anthony Minessale" <anthony.minessale@gmail.com (anthony.minessale@gmail.com)>
To: freeswitch-users@lists.freeswitch.org (freeswitch-users@lists.freeswitch.org)
Sent: Tuesday, February 24, 2009 12:05:33 PM GMT -07:00 US/Canada Mountain
Subject: Re: [Freeswitch-users] Recording and outbound rtp

is it during a bridged call?


On Tue, Feb 24, 2009 at 11:49 AM, Dan <freeswitch-users@digitaldan.com (freeswitch-users@digitaldan.com)> wrote:
Quote:
Hi,

I have a small javascript application that accepts a call, retrieves some dtmf digits and then records the call to an icecast server. This works great.

The problem I'm having is that when the call is being recorded freeswitch is no longer sending rtp packets back to the originating caller, in my case a Cisco 5300 with a bunch of T1 voice circuits in it. This makes sense, since no voice data back is being generated. Unfortunately my Cisco gear has rtp inactivity timers set up to hang up a call after 3 minutes of no incoming rtp packets, this is a global setting that cannot be configured for a single dial peer. Does anyone have a suggestion to generate rtp packets every once in a while? I tried setting comfort noise which did not seem to send anything. I could try playing a empty/short wav file every minute or so but the javascript call session.record is blocking, would a traditional javascript timer and callback to play a wav file be my best bet or is there a better approach? I'm using FreeSWITCH Version 1.0.trunk (12108M) on debian etch.

Thanks!
Dan-




_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org (Freeswitch-users@lists.freeswitch.org)
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




--
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale@hotmail.com ([email]MSN%3Aanthony_minessale@hotmail.com[/email])
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])
pstn:213-799-1400



_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org (Freeswitch-users@lists.freeswitch.org) http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org


_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org (Freeswitch-users@lists.freeswitch.org)
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




--
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/

AIM: anthm
MSN:anthony_minessale@hotmail.com ([email]MSN%3Aanthony_minessale@hotmail.com[/email])
GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com ([email]PAYPAL%3Aanthony.minessale@gmail.com[/email])
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888@conference.freeswitch.org ([email]sip%3A888@conference.freeswitch.org[/email])
iax:guest@conference.freeswitch.org/888
googletalk:conf+888@conference.freeswitch.org ([email]googletalk%3Aconf%2B888@conference.freeswitch.org[/email])
pstn:213-799-1400

_______________________________________________ Freeswitch-users mailing list Freeswitch-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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