Hello, Can someone tell me what clients can be used to test the conference module with the 16k Speex wideband codec? What was used in the testing so far? Thanks Tom __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com
Hello, would be nice to hear some resume, if someone have tested freeswitch. especially i interesting in: h323 -> sip and sip -> h323; codecs (especially g723 and g729 - is available (testing, commercial?); ivr? something like AGI in asterisk (external dialplan applications). Right now i use asterisk pbx and yate, together, at one machine, both giving me freedom :)
On Wed, 2006-07-19 at 10:31 +0100, AmberVoIP wrote:
Hello,
would be nice to hear some resume, if someone have tested freeswitch. especially i interesting in:
h323 -> sip and sip -> h323; you cna do that via woomera easily.
codecs (especially g723 and g729 - is available (testing, commercial?); g.723.1 and g.729 can be written there are plans to provide those codecs soon, however there is a little bit of stuff that has to be done regarding the patent licenses and liability. That should be taken care of shortly however.
ivr? You can do this using a variety of methods, so far I personally have only dont this type of stuff in javascript.
something like AGI in asterisk (external dialplan applications).
everything is intended to be an external dialplan function, I have done some stuff using mod_spidermonkey as have others, and that works quite well. Asterisk tries to get you to do everything in the dialplan by default, and freeswitch is designed to make that hard on purpose. Once the opal stuff is done you will have even more languages to choose from in freeswitch as well.
On Wed, 2006-07-19 at 06:14 -0700, trixter wrote:
On Wed, 2006-07-19 at 10:31 +0100, AmberVoIP wrote:
Hello,
would be nice to hear some resume, if someone have tested freeswitch. especially i interesting in:
h323 -> sip and sip -> h323; you cna do that via woomera easily.
Is woomera included to FreeSwitch? It can route calls without media (RTP forward) ? Possible right now to route call h323 -> sip which coming in g729 or g723?
codecs (especially g723 and g729 - is available (testing, commercial?); g.723.1 and g.729 can be written there are plans to provide those codecs soon, however there is a little bit of stuff that has to be done regarding the patent licenses and liability. That should be taken care of shortly however.
On Wed, 2006-07-19 at 14:16 +0100, AmberVoIP wrote:
On Wed, 2006-07-19 at 06:14 -0700, trixter wrote:
On Wed, 2006-07-19 at 10:31 +0100, AmberVoIP wrote:
Hello,
would be nice to hear some resume, if someone have tested freeswitch. especially i interesting in:
h323 -> sip and sip -> h323; you cna do that via woomera easily.
Is woomera included to FreeSwitch? It can route calls without media (RTP forward) ?
woomera is included with freeswitch, and provides an interface to connect to other protocols/devices in a fairly generic means. it works with openh323 from what I understand and keeps the licensing part of it clean so exceptions arent even needed for that. It has both media and signalling.
Possible right now to route call h323 -> sip which coming in g729 or g723?
That I am unsure of, potentially you should be able to where codec translations occur before hitting woomera and then they are sent out using a non patented codec.
mod_woomera is the woomera client in freeswitch you need to setup the woomera server from the openh323 project which runs in its own process and forwards h323 calls to freeswitch as woomera. This is still under development on both the client and server end but it does currently work. Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com JABBER:anthony.minessale@gmail.com FreeSWITCH Developer Conference sip:888@66.250.68.194 pstn:712-432-7800 ----- Original Message ---- From: AmberVoIP <ambervoip@gmail.com> To: freeswitch-users@lists.freeswitch.org Sent: Wednesday, July 19, 2006 8:16:29 AM Subject: Re: [Freeswitch-users] Codecs and Freeswitch features On Wed, 2006-07-19 at 06:14 -0700, trixter wrote:
On Wed, 2006-07-19 at 10:31 +0100, AmberVoIP wrote:
Hello,
would be nice to hear some resume, if someone have tested freeswitch. especially i interesting in:
h323 -> sip and sip -> h323; you cna do that via woomera easily.
Is woomera included to FreeSwitch? It can route calls without media (RTP forward) ? Possible right now to route call h323 -> sip which coming in g729 or g723?
codecs (especially g723 and g729 - is available (testing, commercial?); g.723.1 and g.729 can be written there are plans to provide those codecs soon, however there is a little bit of stuff that has to be done regarding the patent licenses and liability. That should be taken care of shortly however.
_______________________________________________ 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 can write IVR in javascript with the mod_spidermonkey module there are some unofficial examples at http://www.freeswitch.org/eg/js the syntax may be subject to change a bit but it's the general idea. you can do h323 with woomera from the openh323 project. A new version is due soon to support opal so we will be improving the mod_woomera to support that too. Both codecs are possible it's a matter of getting licenses. and to get something like AGI would be possible in a module but one does not currently exist but you will find the javascript to be very powerful. Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com JABBER:anthony.minessale@gmail.com FreeSWITCH Developer Conference sip:888@66.250.68.194 pstn:712-432-7800 ----- Original Message ---- From: AmberVoIP <ambervoip@gmail.com> To: freeswitch-users@lists.freeswitch.org Sent: Wednesday, July 19, 2006 4:31:50 AM Subject: [Freeswitch-users] Codecs and Freeswitch features Hello, would be nice to hear some resume, if someone have tested freeswitch. especially i interesting in: h323 -> sip and sip -> h323; codecs (especially g723 and g729 - is available (testing, commercial?); ivr? something like AGI in asterisk (external dialplan applications). Right now i use asterisk pbx and yate, together, at one machine, both giving me freedom :) _______________________________________________ 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
In our testing we used the speex codec and mod_exosip talking to eyebeam sip client in exosip.conf section of the freeswitch.xml change the codec_prefs to look like this: <param name="codec-prefs" value="SPEEX@16000k"/> This indicates you want speex at 16khz then look for conference.conf and edit the "default" profile to operate at 16khz <param name="rate" value="16000"/> Then configure eyebeam or whatever sip client to use wideband speex codec and you should be set. If you have mod_cepstral you can load in that and use it at 16k too to announce when you join etc . Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com JABBER:anthony.minessale@gmail.com FreeSWITCH Developer Conference sip:888@66.250.68.194 pstn:712-432-7800 ----- Original Message ---- From: Tom Chez <tj501blues@yahoo.com> To: freeswitch-users@lists.freeswitch.org Sent: Wednesday, July 19, 2006 3:38:16 AM Subject: [Freeswitch-users] 16 K Speex wideband client Hello, Can someone tell me what clients can be used to test the conference module with the 16k Speex wideband codec? What was used in the testing so far? Thanks Tom __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.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
participants (4)
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AmberVoIP
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Anthony Minessale
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Tom Chez
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trixter