What are the ways that I can set up an outbound call to send an additional leg to an external logging recorder? Right now I'm trying to use mod_conference, but it just ends the call as soon as it begins, even though I have another dialplan extension that sets up several endpoints in a conference bridge just fine. I thought about using eavesdrop, but that is initiated by and sends audio to leg A. Plus it joins an existing call whereas in this application I'm establishing a new call, so that won't work. I need to send the audio to leg C from the start. I'm hoping I don't have to write a script, because I'm on the customer's site and I'm running out of time. Any ideas? Thanks. Bote
Use a media bug to record the call? API: http://wiki.freeswitch.org/wiki/Mod_commands#uuid_record Dialplan: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session Or did you mean something completely different? -Avi On Wed, Apr 4, 2012 at 7:03 AM, Bote Man <bote_radio@botecomm.com> wrote:
What are the ways that I can set up an outbound call to send an additional leg to an external logging recorder?
Right now I'm trying to use mod_conference, but it just ends the call as soon as it begins, even though I have another dialplan extension that sets up several endpoints in a conference bridge just fine.
I thought about using eavesdrop, but that is initiated by and sends audio to leg A. Plus it joins an existing call whereas in this application I'm establishing a new call, so that won't work. I need to send the audio to leg C from the start.
I'm hoping I don't have to write a script, because I'm on the customer's site and I'm running out of time.
Any ideas? Thanks.
Bote
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
It's a ridiculously expensive stand-alone recording system that can record RTP streams. It's not up to me to change it at this point. My leg C notion is a fallback in case it doesn't grok SIP calls as advertised; everything else in this project has fallen short of expectations (except FreeSWITCH) so I expect this to fall short as well. :- / Bote On Wed, Apr 4, 2012 at 2:05 AM, Avi Marcus <avi@avimarcus.net> wrote:
Use a media bug to record the call? API: http://wiki.freeswitch.org/wiki/Mod_commands#uuid_record Dialplan: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session
Or did you mean something completely different?
-Avi
On Wed, Apr 4, 2012 at 7:03 AM, Bote Man <bote_radio@botecomm.com> wrote:
What are the ways that I can set up an outbound call to send an additional leg to an external logging recorder?
Right now I'm trying to use mod_conference, but it just ends the call as soon as it begins, even though I have another dialplan extension that sets up several endpoints in a conference bridge just fine.
I thought about using eavesdrop, but that is initiated by and sends audio to leg A. Plus it joins an existing call whereas in this application I'm establishing a new call, so that won't work. I need to send the audio to leg C from the start.
I'm hoping I don't have to write a script, because I'm on the customer's site and I'm running out of time.
Any ideas? Thanks.
Bote
Usually an RTP stream recorder records RTP streams that pass through its network. Can you jigger your network to have it pass through that box? Or some other local packet capture should be able to grab and then forward to another machine. Also, mod_conference should work if the logger picks up the call.. What features does this external solution offer you, to make it worth the effort? -Avi On Wed, Apr 4, 2012 at 9:32 AM, Bote Man <bote_radio@botecomm.com> wrote:
It's a ridiculously expensive stand-alone recording system that can record RTP streams. It's not up to me to change it at this point.
My leg C notion is a fallback in case it doesn't grok SIP calls as advertised; everything else in this project has fallen short of expectations (except FreeSWITCH) so I expect this to fall short as well. :- /
Bote
On Wed, Apr 4, 2012 at 2:05 AM, Avi Marcus <avi@avimarcus.net> wrote:
Use a media bug to record the call? API: http://wiki.freeswitch.org/wiki/Mod_commands#uuid_record Dialplan: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session
Or did you mean something completely different?
-Avi
On Wed, Apr 4, 2012 at 7:03 AM, Bote Man <bote_radio@botecomm.com> wrote:
What are the ways that I can set up an outbound call to send an additional leg to an external logging recorder?
Right now I'm trying to use mod_conference, but it just ends the call as soon as it begins, even though I have another dialplan extension that sets up several endpoints in a conference bridge just fine.
I thought about using eavesdrop, but that is initiated by and sends audio to leg A. Plus it joins an existing call whereas in this application I'm establishing a new call, so that won't work. I need to send the audio to leg C from the start.
I'm hoping I don't have to write a script, because I'm on the customer's site and I'm running out of time.
Any ideas? Thanks.
Bote
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
I guess mod_spy will do the trick for you... Take a look at the wiki page and let us know... Best, Wesley Akio Em 04/04/2012 03:49, "Avi Marcus" <avi@avimarcus.net> escreveu:
Usually an RTP stream recorder records RTP streams that pass through its network. Can you jigger your network to have it pass through that box? Or some other local packet capture should be able to grab and then forward to another machine.
Also, mod_conference should work if the logger picks up the call..
What features does this external solution offer you, to make it worth the effort?
-Avi
On Wed, Apr 4, 2012 at 9:32 AM, Bote Man <bote_radio@botecomm.com> wrote:
It's a ridiculously expensive stand-alone recording system that can record RTP streams. It's not up to me to change it at this point.
My leg C notion is a fallback in case it doesn't grok SIP calls as advertised; everything else in this project has fallen short of expectations (except FreeSWITCH) so I expect this to fall short as well. :- /
Bote
On Wed, Apr 4, 2012 at 2:05 AM, Avi Marcus <avi@avimarcus.net> wrote:
Use a media bug to record the call? API: http://wiki.freeswitch.org/wiki/Mod_commands#uuid_record Dialplan: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session
Or did you mean something completely different?
-Avi
On Wed, Apr 4, 2012 at 7:03 AM, Bote Man <bote_radio@botecomm.com>wrote:
What are the ways that I can set up an outbound call to send an additional leg to an external logging recorder?
Right now I'm trying to use mod_conference, but it just ends the call as soon as it begins, even though I have another dialplan extension that sets up several endpoints in a conference bridge just fine.
I thought about using eavesdrop, but that is initiated by and sends audio to leg A. Plus it joins an existing call whereas in this application I'm establishing a new call, so that won't work. I need to send the audio to leg C from the start.
I'm hoping I don't have to write a script, because I'm on the customer's site and I'm running out of time.
Any ideas? Thanks.
Bote
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
You could do this doing it at router level. This kind of recorders just capture rtp streams but not get involved in the call flow. So if you router have the feature, just pick a port and reproduce all traffic in that port then connect it to your recorder. In that way, your recorder will see all rtp packet in your netwokr On 4/4/2012 2:02 AM, Bote Man wrote:
It's a ridiculously expensive stand-alone recording system that can record RTP streams. It's not up to me to change it at this point.
My leg C notion is a fallback in case it doesn't grok SIP calls as advertised; everything else in this project has fallen short of expectations (except FreeSWITCH) so I expect this to fall short as well. :- /
Bote
On Wed, Apr 4, 2012 at 2:05 AM, Avi Marcus <avi@avimarcus.net <mailto:avi@avimarcus.net>> wrote:
Use a media bug to record the call? API: http://wiki.freeswitch.org/wiki/Mod_commands#uuid_record Dialplan: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session
Or did you mean something completely different?
-Avi
On Wed, Apr 4, 2012 at 7:03 AM, Bote Man <bote_radio@botecomm.com <mailto:bote_radio@botecomm.com>> wrote:
What are the ways that I can set up an outbound call to send an additional leg to an external logging recorder?
Right now I'm trying to use mod_conference, but it just ends the call as soon as it begins, even though I have another dialplan extension that sets up several endpoints in a conference bridge just fine.
I thought about using eavesdrop, but that is initiated by and sends audio to leg A. Plus it joins an existing call whereas in this application I'm establishing a new call, so that won't work. I need to send the audio to leg C from the start.
I'm hoping I don't have to write a script, because I'm on the customer's site and I'm running out of time.
Any ideas? Thanks.
Bote
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
No virus found in this message. Checked by AVG - www.avg.com <http://www.avg.com> Version: 2012.0.1913 / Virus Database: 2114/4913 - Release Date: 04/03/12
-- Atentamente, *Dario García* Consultor. CCCT, Nivel C2, Sector Yarey, Mz, Ofc. MZ03a. Caracas-Venezuela. Teléfono: +58 212 9081842 Cel: +58 412 2221515 dgarcia@anew.com.ve http://www.anew.com.ve
Yes, I am hoping that this Cisco supports SPAN (Switched Port Analyzer) because that is exactly what I need. Unfortunately, that is a job for someone else involved in this project due to political and business reasons, so I can not implement it or even check if it has that feature. The problem with eavesdrop and mod_spy is that they depend on the listener/recorder initiating the dialplan action. The recorder is passive so the stream must be sent to it as another party to the call, unless I am missing something here. Thanks. Bote On Wed, Apr 4, 2012 at 7:20 AM, Saugort Dario Garcia Tovar < dgarcia@anew.com.ve> wrote:
You could do this doing it at router level. This kind of recorders just capture rtp streams but not get involved in the call flow. So if you router have the feature, just pick a port and reproduce all traffic in that port then connect it to your recorder. In that way, your recorder will see all rtp packet in your netwokr
On 4/4/2012 2:02 AM, Bote Man wrote:
It's a ridiculously expensive stand-alone recording system that can record RTP streams. It's not up to me to change it at this point.
My leg C notion is a fallback in case it doesn't grok SIP calls as advertised; everything else in this project has fallen short of expectations (except FreeSWITCH) so I expect this to fall short as well. :- /
Bote
On Wed, Apr 4, 2012 at 2:05 AM, Avi Marcus <avi@avimarcus.net> wrote:
Use a media bug to record the call? API: http://wiki.freeswitch.org/wiki/Mod_commands#uuid_record Dialplan: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session
Or did you mean something completely different?
-Avi
On Wed, Apr 4, 2012 at 7:03 AM, Bote Man <bote_radio@botecomm.com>wrote:
What are the ways that I can set up an outbound call to send an additional leg to an external logging recorder?
Right now I'm trying to use mod_conference, but it just ends the call as soon as it begins, even though I have another dialplan extension that sets up several endpoints in a conference bridge just fine.
I thought about using eavesdrop, but that is initiated by and sends audio to leg A. Plus it joins an existing call whereas in this application I'm establishing a new call, so that won't work. I need to send the audio to leg C from the start.
I'm hoping I don't have to write a script, because I'm on the customer's site and I'm running out of time.
Any ideas? Thanks.
Bote
_________________________________________________________________________ Professional FreeSWITCH Consulting Services:consulting@freeswitch.orghttp://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Serverhttp://www.cudatel.com
Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com
FreeSWITCH-users mailing listFreeSWITCH-users@lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.frees...
No virus found in this message. Checked by AVG - www.avg.com Version: 2012.0.1913 / Virus Database: 2114/4913 - Release Date: 04/03/12
-- Atentamente, *Dario García* Consultor.
CCCT, Nivel C2, Sector Yarey, Mz, Ofc. MZ03a. Caracas-Venezuela. Teléfono: +58 212 9081842 Cel: +58 412 2221515 dgarcia@anew.com.ve http://www.anew.com.ve
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
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
I assume your FreeSWITCH system is Linux? You could try using iptables tee functionality: http://www.highonphp.com/packet-cloning-with-iptables http://www.bjou.de/blog/2008/05/howto-copyteeclone-network-traffic-using-ipt... This will increase the network and cpu load on your FreeSWITCH machine somewhat but if your traffic rate isn't very high it should work. You'll probably want to use multiple rules and --dport/--sport matches to only include SIP and RTP traffic. On Wed, Apr 4, 2012 at 8:53 AM, Bote Man <bote_radio@botecomm.com> wrote:
Yes, I am hoping that this Cisco supports SPAN (Switched Port Analyzer) because that is exactly what I need. Unfortunately, that is a job for someone else involved in this project due to political and business reasons, so I can not implement it or even check if it has that feature.
The problem with eavesdrop and mod_spy is that they depend on the listener/recorder initiating the dialplan action. The recorder is passive so the stream must be sent to it as another party to the call, unless I am missing something here.
Thanks.
Bote
-- Kristian Kielhofner
participants (5)
-
Avi Marcus
-
Bote Man
-
Kristian Kielhofner
-
Saugort Dario Garcia Tovar
-
Wesley Akio