Re: [Freeswitch-users] Conference dialing and uuid
On Tue, Feb 3, 2009 at 6:16 AM, Sias Mey wrote:
> Hmm ok ... Ill try that In my head though the api call to originate
> shouldent block? but I assume since it does my head is wrong.
You can use "bgapi originate" to do it in a non-blocking way. See the
very last example for the originate command:
http://wiki.freeswitch.org/wiki/Mod_commands#originate
-MC
>
> Thanks you for the explanation. I think you can put this one to bed now
> :-P
>
> On Tue, Feb 03, 2009 at 07:54:29AM -0600, Anthony Minessale wrote:
>>There is a file format called tone_stream that I was trying to explain
>>yesterday.
>>tone_stream://
>>or
>>tone_stream://path=/path/to/text_file.ttml
>>you can use this to play tones anywhere a filename is supposed to go.
>>I guess loopback really is your only option if you must generate
>>ringback.
>>Typically, whatever gateway you are calling out over will go into early
>>media and start playing the real ringback.
>>You should not execute any apps during the on_ring_execute that block,
>>(playing audio etc)
>>Media has not even been established at that point and you have nobody
>>to play the audio to anyway,
>>But you will block from that point until the application you chose has
>>ended so you should only execute small apps that
>>return immediately such as setting a variable etc.
>> As for ringback I think you have the whole thing reversed in your
>>head.
>>the ringback vars etc only apply to the origination (a) leg of a call.
>>If you make an inbound call set the ringback variable and then call
>>bridge, the ringback var is parsed on that inbound leg
>>and the dialout process of the bridge app involves 2 channels the A leg
>>and the B leg. When the B leg gets a ring indication and the A leg
>>detects it, it will begin to play the ringback sound you chose back to
>>the originator of that inbound leg.
>>In the conference or using originate situation, you are doing an
>>outbound call with no relevant inbound call, so there is nothing
>>to generate ringback to. That's why loopback works because it cross
>>connects an outbound call back to an inbound call which gives the
>>bridge app everything it needs to be able to generate artificial
>>ringback.
>>
>>On Tue, Feb 3, 2009 at 2:25 AM, Sias Mey <[1]s...@cpdata.co.za> wrote:
>>
>> Hmmm no MOH wont work... since I am planning on pulling more than
>> just 2
>> members into the conference and I still need ringback for the later
>> members as well.
>> Is there a direct way for me to use conference play
>>
>> to play teletone directly? or should I just records some ringing if
>> I
>> want to use that?
>> And lastly for my own sanity ;-) why would the following in a
>> on_ring_execute stop execution of the call at that point?
>> call = argv[1];
>> conf = argv[2];
>> consoleLog("info","Making ringback channel for uuid : "+
>> session.uuid
>> +"\n");
>> var ringuuid = apiExecute("originate","loopback/ringback-conf="+
>> conf +"-conf &park()")
>> //I tried with and without a exit() at the end
>> It seems to stop media detection??(not really sure about the term)
>> for the call that executes this
>> script.
>> Freeswitch doesent recognize the pickup of that call and thus it
>> doesent
>> get bridged into the conference. when I uuid_kill the call that gets
>> originated everything else starts happening again.
>> Oh Im running this in FS ver. 1.0.trunk (11226:11561M)
>> and that loopback points to
>>
>> > expression="^ringback-conf=(.*)$">
>>
>>
>>
>> and ringback.js is
>> use("TeleTone");
>> session.answer();
>> var tts = new TeleTone(session);
>> tts.addTone("u", 400.0, 450.0, 0.0);
>> tts.addTone("r", 440.0, 480.0, 0.0);
>> var RESET = "v=2000;>=0;+=0;";
>> var UK_RING = RESET + "L=2;u(400,200);u(400,2200)";
>> var US_RING = RESET + "r(2000,4000)";
>> while(session.ready()) {
>> console_log("making UK ring\n");
>> for (x = 0 ; x < 2 ; x++) {
>> tts.generate(UK_RING);
>> }
>> }
>> A slight bastardisation of the teletone JS example.
>> I would expected the new channel that is created via a api originate
>> to
>> be completely seperate from the JS I create it in. (thats why I use
>> api
>> instead of creating a new session, although I should probably try
>> that
>> as well).
>> I use some CoreDB stuff to keep tabs on the uuid for the originated
>> call
>> so that I can uuid_kill it in the on_answer_script but as
>> mentioned...
>> the on_answer only executes after I uuid_kill the originated channel
>> in
>> the cli...
>> Thanks again guys,
>> Specially since it seems you two are always the
Re: [Freeswitch-users] Conference dialing and uuid
hmmm ok indeed.
small mods to js files to just play a lng tone_stream full of ringy
noises and then stop them in the on answer and I have what I wanted.
Thank you very very much for all your help.
On Tue, Feb 03, 2009 at 04:16:21PM +0200, Sias Mey wrote:
> Hmm ok ... Ill try that In my head though the api call to originate
> shouldent block? but I assume since it does my head is wrong.
>
> Thanks you for the explanation. I think you can put this one to bed now
> :-P
>
> On Tue, Feb 03, 2009 at 07:54:29AM -0600, Anthony Minessale wrote:
> >There is a file format called tone_stream that I was trying to explain
> >yesterday.
> >tone_stream://
> >or
> >tone_stream://path=/path/to/text_file.ttml
> >you can use this to play tones anywhere a filename is supposed to go.
> >I guess loopback really is your only option if you must generate
> >ringback.
> >Typically, whatever gateway you are calling out over will go into early
> >media and start playing the real ringback.
> >You should not execute any apps during the on_ring_execute that block,
> >(playing audio etc)
> >Media has not even been established at that point and you have nobody
> >to play the audio to anyway,
> >But you will block from that point until the application you chose has
> >ended so you should only execute small apps that
> >return immediately such as setting a variable etc.
> > As for ringback I think you have the whole thing reversed in your
> >head.
> >the ringback vars etc only apply to the origination (a) leg of a call.
> >If you make an inbound call set the ringback variable and then call
> >bridge, the ringback var is parsed on that inbound leg
> >and the dialout process of the bridge app involves 2 channels the A leg
> >and the B leg. When the B leg gets a ring indication and the A leg
> >detects it, it will begin to play the ringback sound you chose back to
> >the originator of that inbound leg.
> >In the conference or using originate situation, you are doing an
> >outbound call with no relevant inbound call, so there is nothing
> >to generate ringback to. That's why loopback works because it cross
> >connects an outbound call back to an inbound call which gives the
> >bridge app everything it needs to be able to generate artificial
> >ringback.
> >
> >On Tue, Feb 3, 2009 at 2:25 AM, Sias Mey <[1]s...@cpdata.co.za> wrote:
> >
> > Hmmm no MOH wont work... since I am planning on pulling more than
> > just 2
> > members into the conference and I still need ringback for the later
> > members as well.
> > Is there a direct way for me to use conference play
> >
> > to play teletone directly? or should I just records some ringing if
> > I
> > want to use that?
> > And lastly for my own sanity ;-) why would the following in a
> > on_ring_execute stop execution of the call at that point?
> > call = argv[1];
> > conf = argv[2];
> > consoleLog("info","Making ringback channel for uuid : "+
> > session.uuid
> > +"\n");
> > var ringuuid = apiExecute("originate","loopback/ringback-conf="+
> > conf +"-conf &park()")
> > //I tried with and without a exit() at the end
> > It seems to stop media detection??(not really sure about the term)
> > for the call that executes this
> > script.
> > Freeswitch doesent recognize the pickup of that call and thus it
> > doesent
> > get bridged into the conference. when I uuid_kill the call that gets
> > originated everything else starts happening again.
> > Oh Im running this in FS ver. 1.0.trunk (11226:11561M)
> > and that loopback points to
> >
> > > expression="^ringback-conf=(.*)$">
> >
> >
> >
> > and ringback.js is
> > use("TeleTone");
> > session.answer();
> > var tts = new TeleTone(session);
> > tts.addTone("u", 400.0, 450.0, 0.0);
> > tts.addTone("r", 440.0, 480.0, 0.0);
> > var RESET = "v=2000;>=0;+=0;";
> > var UK_RING = RESET + "L=2;u(400,200);u(400,2200)";
> > var US_RING = RESET + "r(2000,4000)";
> > while(session.ready()) {
> > console_log("making UK ring\n");
> > for (x = 0 ; x < 2 ; x++) {
> > tts.generate(UK_RING);
> > }
> > }
> > A slight bastardisation of the teletone JS example.
> > I would expected the new channel that is created via a api originate
> > to
> > be completely seperate from the JS I create it in. (thats why I use
> > api
> > instead of creating a new session, although I should probably try
> > that
> > as well).
> > I use some CoreDB stuff to keep tabs on the uuid for the originated
> > call
> > so that I can uuid_kill it in the on_answer_script but as
> > mentioned...
> > the on_answer only executes after I uuid_kil
Re: [Freeswitch-users] Conference dialing and uuid
Hmm ok ... Ill try that In my head though the api call to originate
shouldent block? but I assume since it does my head is wrong.
Thanks you for the explanation. I think you can put this one to bed now
:-P
On Tue, Feb 03, 2009 at 07:54:29AM -0600, Anthony Minessale wrote:
>There is a file format called tone_stream that I was trying to explain
>yesterday.
>tone_stream://
>or
>tone_stream://path=/path/to/text_file.ttml
>you can use this to play tones anywhere a filename is supposed to go.
>I guess loopback really is your only option if you must generate
>ringback.
>Typically, whatever gateway you are calling out over will go into early
>media and start playing the real ringback.
>You should not execute any apps during the on_ring_execute that block,
>(playing audio etc)
>Media has not even been established at that point and you have nobody
>to play the audio to anyway,
>But you will block from that point until the application you chose has
>ended so you should only execute small apps that
>return immediately such as setting a variable etc.
> As for ringback I think you have the whole thing reversed in your
>head.
>the ringback vars etc only apply to the origination (a) leg of a call.
>If you make an inbound call set the ringback variable and then call
>bridge, the ringback var is parsed on that inbound leg
>and the dialout process of the bridge app involves 2 channels the A leg
>and the B leg. When the B leg gets a ring indication and the A leg
>detects it, it will begin to play the ringback sound you chose back to
>the originator of that inbound leg.
>In the conference or using originate situation, you are doing an
>outbound call with no relevant inbound call, so there is nothing
>to generate ringback to. That's why loopback works because it cross
>connects an outbound call back to an inbound call which gives the
>bridge app everything it needs to be able to generate artificial
>ringback.
>
>On Tue, Feb 3, 2009 at 2:25 AM, Sias Mey <[1]s...@cpdata.co.za> wrote:
>
> Hmmm no MOH wont work... since I am planning on pulling more than
> just 2
> members into the conference and I still need ringback for the later
> members as well.
> Is there a direct way for me to use conference play
>
> to play teletone directly? or should I just records some ringing if
> I
> want to use that?
> And lastly for my own sanity ;-) why would the following in a
> on_ring_execute stop execution of the call at that point?
> call = argv[1];
> conf = argv[2];
> consoleLog("info","Making ringback channel for uuid : "+
> session.uuid
> +"\n");
> var ringuuid = apiExecute("originate","loopback/ringback-conf="+
> conf +"-conf &park()")
> //I tried with and without a exit() at the end
> It seems to stop media detection??(not really sure about the term)
> for the call that executes this
> script.
> Freeswitch doesent recognize the pickup of that call and thus it
> doesent
> get bridged into the conference. when I uuid_kill the call that gets
> originated everything else starts happening again.
> Oh Im running this in FS ver. 1.0.trunk (11226:11561M)
> and that loopback points to
>
>expression="^ringback-conf=(.*)$">
>
>
>
> and ringback.js is
> use("TeleTone");
> session.answer();
> var tts = new TeleTone(session);
> tts.addTone("u", 400.0, 450.0, 0.0);
> tts.addTone("r", 440.0, 480.0, 0.0);
> var RESET = "v=2000;>=0;+=0;";
> var UK_RING = RESET + "L=2;u(400,200);u(400,2200)";
> var US_RING = RESET + "r(2000,4000)";
> while(session.ready()) {
> console_log("making UK ring\n");
> for (x = 0 ; x < 2 ; x++) {
> tts.generate(UK_RING);
> }
> }
> A slight bastardisation of the teletone JS example.
> I would expected the new channel that is created via a api originate
> to
> be completely seperate from the JS I create it in. (thats why I use
> api
> instead of creating a new session, although I should probably try
> that
> as well).
> I use some CoreDB stuff to keep tabs on the uuid for the originated
> call
> so that I can uuid_kill it in the on_answer_script but as
> mentioned...
> the on_answer only executes after I uuid_kill the originated channel
> in
> the cli...
> Thanks again guys,
> Specially since it seems you two are always the ones that get back
> to
> me.
>
>On Tue, Feb 03, 2009 at 09:22:21AM +0200, Sias Mey wrote:
>> Actually loopback does work.
>> however as I said it generates a pair of extra channels.
>>
>> Hmmm I was trying to generate and extra call to a JS script that
>> generated a teletone ring in an on_ring_exec
Re: [Freeswitch-users] Conference dialing and uuid
There is a file format called tone_stream that I was trying to explain
yesterday.
tone_stream://
or
tone_stream://path=/path/to/text_file.ttml
you can use this to play tones anywhere a filename is supposed to go.
I guess loopback really is your only option if you must generate ringback.
Typically, whatever gateway you are calling out over will go into early
media and start playing the real ringback.
You should not execute any apps during the on_ring_execute that block,
(playing audio etc)
Media has not even been established at that point and you have nobody to
play the audio to anyway,
But you will block from that point until the application you chose has ended
so you should only execute small apps that
return immediately such as setting a variable etc.
As for ringback I think you have the whole thing reversed in your head.
the ringback vars etc only apply to the origination (a) leg of a call.
If you make an inbound call set the ringback variable and then call bridge,
the ringback var is parsed on that inbound leg
and the dialout process of the bridge app involves 2 channels the A leg and
the B leg. When the B leg gets a ring indication and the A leg detects it,
it will begin to play the ringback sound you chose back to the originator of
that inbound leg.
In the conference or using originate situation, you are doing an outbound
call with no relevant inbound call, so there is nothing
to generate ringback to. That's why loopback works because it cross
connects an outbound call back to an inbound call which gives the bridge app
everything it needs to be able to generate artificial ringback.
On Tue, Feb 3, 2009 at 2:25 AM, Sias Mey wrote:
> Hmmm no MOH wont work... since I am planning on pulling more than just 2
> members into the conference and I still need ringback for the later
> members as well.
>
> Is there a direct way for me to use conference play
> to play teletone directly? or should I just records some ringing if I
> want to use that?
>
> And lastly for my own sanity ;-) why would the following in a
> on_ring_execute stop execution of the call at that point?
>
> call = argv[1];
> conf = argv[2];
>
> consoleLog("info","Making ringback channel for uuid : "+ session.uuid
> +"\n");
> var ringuuid = apiExecute("originate","loopback/ringback-conf="+ conf
> +"-conf &park()")
>
> //I tried with and without a exit() at the end
>
> It seems to stop media detection??(not really sure about the term) for the
> call that executes this
> script.
>
> Freeswitch doesent recognize the pickup of that call and thus it doesent
> get bridged into the conference. when I uuid_kill the call that gets
> originated everything else starts happening again.
>
> Oh Im running this in FS ver. 1.0.trunk (11226:11561M)
>
> and that loopback points to
>
>
>
>
>
>
> and ringback.js is
>
> use("TeleTone");
> session.answer();
> var tts = new TeleTone(session);
>
> tts.addTone("u", 400.0, 450.0, 0.0);
> tts.addTone("r", 440.0, 480.0, 0.0);
>
> var RESET = "v=2000;>=0;+=0;";
> var UK_RING = RESET + "L=2;u(400,200);u(400,2200)";
> var US_RING = RESET + "r(2000,4000)";
>
> while(session.ready()) {
> console_log("making UK ring\n");
>for (x = 0 ; x < 2 ; x++) {
>tts.generate(UK_RING);
>}
> }
>
> A slight bastardisation of the teletone JS example.
>
> I would expected the new channel that is created via a api originate to
> be completely seperate from the JS I create it in. (thats why I use api
> instead of creating a new session, although I should probably try that
> as well).
>
> I use some CoreDB stuff to keep tabs on the uuid for the originated call
> so that I can uuid_kill it in the on_answer_script but as mentioned...
> the on_answer only executes after I uuid_kill the originated channel in
> the cli...
>
> Thanks again guys,
> Specially since it seems you two are always the ones that get back to
> me.
>
> On Tue, Feb 03, 2009 at 09:22:21AM +0200, Sias Mey wrote:
> > Actually loopback does work.
> > however as I said it generates a pair of extra channels.
> >
> > Hmmm I was trying to generate and extra call to a JS script that
> > generated a teletone ring in an on_ring_execute for the second call
> > however it seems to stop execution of the call itself. Event though I
> > use api commands to originate and then transfer it into the conference
> > so that I have direct access to its uuid.
> >
> > I think changeing the moh might be a bit simpler however and elimite
> > some CoreDB stuff I was doing to keep track of the calls ring generating
> > call (what a sentance).
> >
> > On Mon, Feb 02, 2009 at 08:01:25AM -0600, Anthony Minessale wrote:
> > >you could set the conference moh sound to be tone_stream::// with
> the
> > >teletone spec for ring sound and it use ignore_early_media=true in
> your
> > >originates so the first caller would hear ringback until the 2nd one
> > >arrived.
> > >
> > >On Mon, Feb 2, 2009 at 4:29 AM, Brian West <[1]br...@freeswitch.org
> >
> > >
Re: [Freeswitch-users] Conference dialing and uuid
Hmmm no MOH wont work... since I am planning on pulling more than just 2
members into the conference and I still need ringback for the later
members as well.
Is there a direct way for me to use conference play
to play teletone directly? or should I just records some ringing if I
want to use that?
And lastly for my own sanity ;-) why would the following in a
on_ring_execute stop execution of the call at that point?
call = argv[1];
conf = argv[2];
consoleLog("info","Making ringback channel for uuid : "+ session.uuid
+"\n");
var ringuuid = apiExecute("originate","loopback/ringback-conf="+ conf +"-conf
&park()")
//I tried with and without a exit() at the end
It seems to stop media detection??(not really sure about the term) for the call
that executes this
script.
Freeswitch doesent recognize the pickup of that call and thus it doesent
get bridged into the conference. when I uuid_kill the call that gets
originated everything else starts happening again.
Oh Im running this in FS ver. 1.0.trunk (11226:11561M)
and that loopback points to
and ringback.js is
use("TeleTone");
session.answer();
var tts = new TeleTone(session);
tts.addTone("u", 400.0, 450.0, 0.0);
tts.addTone("r", 440.0, 480.0, 0.0);
var RESET = "v=2000;>=0;+=0;";
var UK_RING = RESET + "L=2;u(400,200);u(400,2200)";
var US_RING = RESET + "r(2000,4000)";
while(session.ready()) {
console_log("making UK ring\n");
for (x = 0 ; x < 2 ; x++) {
tts.generate(UK_RING);
}
}
A slight bastardisation of the teletone JS example.
I would expected the new channel that is created via a api originate to
be completely seperate from the JS I create it in. (thats why I use api
instead of creating a new session, although I should probably try that
as well).
I use some CoreDB stuff to keep tabs on the uuid for the originated call
so that I can uuid_kill it in the on_answer_script but as mentioned...
the on_answer only executes after I uuid_kill the originated channel in
the cli...
Thanks again guys,
Specially since it seems you two are always the ones that get back to
me.
On Tue, Feb 03, 2009 at 09:22:21AM +0200, Sias Mey wrote:
> Actually loopback does work.
> however as I said it generates a pair of extra channels.
>
> Hmmm I was trying to generate and extra call to a JS script that
> generated a teletone ring in an on_ring_execute for the second call
> however it seems to stop execution of the call itself. Event though I
> use api commands to originate and then transfer it into the conference
> so that I have direct access to its uuid.
>
> I think changeing the moh might be a bit simpler however and elimite
> some CoreDB stuff I was doing to keep track of the calls ring generating
> call (what a sentance).
>
> On Mon, Feb 02, 2009 at 08:01:25AM -0600, Anthony Minessale wrote:
> >you could set the conference moh sound to be tone_stream::// with the
> >teletone spec for ring sound and it use ignore_early_media=true in your
> >originates so the first caller would hear ringback until the 2nd one
> >arrived.
> >
> >On Mon, Feb 2, 2009 at 4:29 AM, Brian West <[1]br...@freeswitch.org>
> >wrote:
> >
> > Loopback will not work in that case either. If the far end plays
> > ringback inband you should hear that if you use the conference dial
> > api call.
> > /b
> >
> >On Feb 2, 2009, at 4:24 AM, Sias Mey wrote:
> >> Aaah ok.
> >>
> >> Thanks for clearing that up.
> >>
> >> So using loopback is still the only real workable sollution for me,
> >> since that generates ringback from and alternative endpoint and
> >> plays it
> >> into the conference.
> >>
> >> I might play with some javascript that streams ring into the channel
> >> eventually but for now the string comparisons at least get me the
> >> right
> >> uuid.
> >>
> >> Thank you again,
> >> Sias
> >
> >___
> >Freeswitch-users mailing list
> >[2]freeswitch-us...@lists.freeswitch.org
> >[3]http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >UNSUBSCRIBE:[4]http://lists.freeswitch.org/mailman/options/freeswitch-u
> >sers
> >[5]http://www.freeswitch.org
> >
> >--
> >Anthony Minessale II
> >FreeSWITCH [6]http://www.freeswitch.org/
> >ClueCon [7]http://www.cluecon.com/
> >AIM: anthm
> >[8]MSN:anthony_miness...@hotmail.com
> >GTALK/JABBER/[9]PAYPAL:anthony.miness...@gmail.com
> >IRC: [10]irc.freenode.net #freeswitch
> >FreeSWITCH Developer Conference
> >[11]sip:8...@conference.freeswitch.org
> >[12]iax:gu...@conference.freeswitch.org/888
> >[13]googletalk:conf+...@conference.freeswitch.org
> >pstn:213-799-1400
> >
> > References
> >
> >1. mailto:br...@freeswitch.org
> >2. mailto:Freeswitch-users@lists.freeswitch.org
> >3. http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >4. http://lists.frees
Re: [Freeswitch-users] Conference dialing and uuid
Actually loopback does work. however as I said it generates a pair of extra channels. Hmmm I was trying to generate and extra call to a JS script that generated a teletone ring in an on_ring_execute for the second call however it seems to stop execution of the call itself. Event though I use api commands to originate and then transfer it into the conference so that I have direct access to its uuid. I think changeing the moh might be a bit simpler however and elimite some CoreDB stuff I was doing to keep track of the calls ring generating call (what a sentance). On Mon, Feb 02, 2009 at 08:01:25AM -0600, Anthony Minessale wrote: >you could set the conference moh sound to be tone_stream::// with the >teletone spec for ring sound and it use ignore_early_media=true in your >originates so the first caller would hear ringback until the 2nd one >arrived. > >On Mon, Feb 2, 2009 at 4:29 AM, Brian West <[1]br...@freeswitch.org> >wrote: > > Loopback will not work in that case either. If the far end plays > ringback inband you should hear that if you use the conference dial > api call. > /b > >On Feb 2, 2009, at 4:24 AM, Sias Mey wrote: >> Aaah ok. >> >> Thanks for clearing that up. >> >> So using loopback is still the only real workable sollution for me, >> since that generates ringback from and alternative endpoint and >> plays it >> into the conference. >> >> I might play with some javascript that streams ring into the channel >> eventually but for now the string comparisons at least get me the >> right >> uuid. >> >> Thank you again, >> Sias > >___ >Freeswitch-users mailing list >[2]freeswitch-us...@lists.freeswitch.org >[3]http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:[4]http://lists.freeswitch.org/mailman/options/freeswitch-u >sers >[5]http://www.freeswitch.org > >-- >Anthony Minessale II >FreeSWITCH [6]http://www.freeswitch.org/ >ClueCon [7]http://www.cluecon.com/ >AIM: anthm >[8]MSN:anthony_miness...@hotmail.com >GTALK/JABBER/[9]PAYPAL:anthony.miness...@gmail.com >IRC: [10]irc.freenode.net #freeswitch >FreeSWITCH Developer Conference >[11]sip:8...@conference.freeswitch.org >[12]iax:gu...@conference.freeswitch.org/888 >[13]googletalk:conf+...@conference.freeswitch.org >pstn:213-799-1400 > > References > >1. mailto:br...@freeswitch.org >2. mailto:Freeswitch-users@lists.freeswitch.org >3. http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >4. http://lists.freeswitch.org/mailman/options/freeswitch-users >5. http://www.freeswitch.org/ >6. http://www.freeswitch.org/ >7. http://www.cluecon.com/ >8. mailto:msn%3aanthony_miness...@hotmail.com >9. mailto:paypal%3aanthony.miness...@gmail.com > 10. http://irc.freenode.net/ > 11. mailto:sip%3a...@conference.freeswitch.org > 12. http://iax:gu...@conference.freeswitch.org/888 > 13. mailto:googletalk%3aconf%2b...@conference.freeswitch.org > ___ > 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 ___ 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
Re: [Freeswitch-users] Conference dialing and uuid
you could set the conference moh sound to be tone_stream::// with the teletone spec for ring sound and it use ignore_early_media=true in your originates so the first caller would hear ringback until the 2nd one arrived. On Mon, Feb 2, 2009 at 4:29 AM, Brian West wrote: > Loopback will not work in that case either. If the far end plays > ringback inband you should hear that if you use the conference dial > api call. > > /b > > On Feb 2, 2009, at 4:24 AM, Sias Mey wrote: > > > Aaah ok. > > > > Thanks for clearing that up. > > > > So using loopback is still the only real workable sollution for me, > > since that generates ringback from and alternative endpoint and > > plays it > > into the conference. > > > > I might play with some javascript that streams ring into the channel > > eventually but for now the string comparisons at least get me the > > right > > uuid. > > > > Thank you again, > > Sias > > > ___ > 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 > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.org 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
Re: [Freeswitch-users] Conference dialing and uuid
Loopback will not work in that case either. If the far end plays ringback inband you should hear that if you use the conference dial api call. /b On Feb 2, 2009, at 4:24 AM, Sias Mey wrote: > Aaah ok. > > Thanks for clearing that up. > > So using loopback is still the only real workable sollution for me, > since that generates ringback from and alternative endpoint and > plays it > into the conference. > > I might play with some javascript that streams ring into the channel > eventually but for now the string comparisons at least get me the > right > uuid. > > Thank you again, > Sias ___ 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
Re: [Freeswitch-users] Conference dialing and uuid
Aaah ok.
Thanks for clearing that up.
So using loopback is still the only real workable sollution for me,
since that generates ringback from and alternative endpoint and plays it
into the conference.
I might play with some javascript that streams ring into the channel
eventually but for now the string comparisons at least get me the right
uuid.
Thank you again,
Sias
On Mon, Feb 02, 2009 at 02:55:42AM -0600, Brian West wrote:
>You can't get ringback dialing out from a conference its not possible
>as it is now.
>
>/b
>
>On Feb 2, 2009, at 2:51 AM, Sias Mey wrote:
>
> Yes ... yes indeed I can.
> That works quite a bit better than generating 4 channels and getting
> massively confused with what uuid does what... but now im stuck
> without
> ringback again :-(.
> In my conference dial string I send:
> {ringback=\'%(400,200,400,450)\',transfer_ringback=\'%(400,200,400,4
> 50)\',
> [1]}sofia/internal/1...@xxx.xxx.xxx.xxx
> A dump of all the channel variables shows ringback is set to
> %25(400,200,400,450)%3B%25(400,2200,400,450)
> %25(400,200,400,450)%3B%25(400,2200,400,450) transfer_ringback>
> This seems ok to me but I still dont get any ringback.
> Thanks again for answering all the anoying questions from the same
> guy
> :-P,
> Sias
>
> References
>
>1. mailto:}sofia/internal/1...@xxx.xxx.xxx.xxx
> ___
> 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
___
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
Re: [Freeswitch-users] Conference dialing and uuid
You can't get ringback dialing out from a conference its not possible
as it is now.
/b
On Feb 2, 2009, at 2:51 AM, Sias Mey wrote:
Yes ... yes indeed I can.
That works quite a bit better than generating 4 channels and getting
massively confused with what uuid does what... but now im stuck
without
ringback again :-(.
In my conference dial string I send:
{ringback=\'%(400,200,400,450)\',transfer_ringback=
\'%(400,200,400,450)\',
}sofia/internal/1...@xxx.xxx.xxx.xxx
A dump of all the channel variables shows ringback is set to
%25(400,200,400,450)%3B%25(400,2200,400,450)
%25(400,200,400,450)%3B%25(400,2200,400,450)transfer_ringback>
This seems ok to me but I still dont get any ringback.
Thanks again for answering all the anoying questions from the same guy
:-P,
Sias
___
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
Re: [Freeswitch-users] Conference dialing and uuid
Yes ... yes indeed I can.
That works quite a bit better than generating 4 channels and getting
massively confused with what uuid does what... but now im stuck without
ringback again :-(.
In my conference dial string I send:
{ringback=\'%(400,200,400,450)\',transfer_ringback=\'%(400,200,400,450)\',
}sofia/internal/1...@xxx.xxx.xxx.xxx
A dump of all the channel variables shows ringback is set to
%25(400,200,400,450)%3B%25(400,2200,400,450)
%25(400,200,400,450)%3B%25(400,2200,400,450)
This seems ok to me but I still dont get any ringback.
Thanks again for answering all the anoying questions from the same guy
:-P,
Sias
On Fri, Jan 30, 2009 at 08:05:07AM -0600, Anthony Minessale wrote:
>you should be able to use {} in the dial command
>you also should be able to do
>originate {...}sofia/profile/[1]u...@domain.com
>conference:@ inline
>to the api interface
>
>On Fri, Jan 30, 2009 at 7:33 AM, Sias Mey <[2]s...@cpdata.co.za> wrote:
>
> Hi Brian,
> Hmmm Ill do some more testing on it later. But I got a destination
> out
> of order when I tried. Right now Im busy implementing the string
> checking. Which seems like it will work out ok, but is clearly not
> ideal.
> Thanks for the replay
>
>On Fri, Jan 30, 2009 at 04:45:54AM -0600, Brian West wrote:
>> What wasn't working about this? The {} can be used everywhere
>without
>> a problem... Maybe you can provide more details on this.
>>
>> /b
>>
>>
>>
>>
>> On Jan 30, 2009, at 4:39 AM, Sias Mey wrote:
>>
>> >
>> > I couldent find a way of setting channel variables or executing
>> > javascript directly on the conference dial since it expects and
>> > endpoint
>> > and the {} syntax produced an error. So now I am using the Loopback
>> > inteface to register some values.
>>
>>
>> ___
>> Freeswitch-users mailing list
>> [3]freeswitch-us...@lists.freeswitch.org
>> [4]http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>
>UNSUBSCRIBE:[5]http://lists.freeswitch.org/mailman/options/freeswitch-u
>sers
>> [6]http://www.freeswitch.org
>___
>Freeswitch-users mailing list
>[7]freeswitch-us...@lists.freeswitch.org
>[8]http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>UNSUBSCRIBE:[9]http://lists.freeswitch.org/mailman/options/freeswitch-u
>sers
>[10]http://www.freeswitch.org
>
>--
>Anthony Minessale II
>FreeSWITCH [11]http://www.freeswitch.org/
>ClueCon [12]http://www.cluecon.com/
>AIM: anthm
>[13]MSN:anthony_miness...@hotmail.com
>GTALK/JABBER/[14]PAYPAL:anthony.miness...@gmail.com
>IRC: [15]irc.freenode.net #freeswitch
>FreeSWITCH Developer Conference
>[16]sip:8...@conference.freeswitch.org
>[17]iax:gu...@conference.freeswitch.org/888
>[18]googletalk:conf+...@conference.freeswitch.org
>pstn:213-799-1400
>
> References
>
>1. mailto:u...@domain.com
>2. mailto:s...@cpdata.co.za
>3. mailto:Freeswitch-users@lists.freeswitch.org
>4. http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>5. http://lists.freeswitch.org/mailman/options/freeswitch-users
>6. http://www.freeswitch.org/
>7. mailto:Freeswitch-users@lists.freeswitch.org
>8. http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>9. http://lists.freeswitch.org/mailman/options/freeswitch-users
> 10. http://www.freeswitch.org/
> 11. http://www.freeswitch.org/
> 12. http://www.cluecon.com/
> 13. mailto:msn%3aanthony_miness...@hotmail.com
> 14. mailto:paypal%3aanthony.miness...@gmail.com
> 15. http://irc.freenode.net/
> 16. mailto:sip%3a...@conference.freeswitch.org
> 17. http://iax:gu...@conference.freeswitch.org/888
> 18. mailto:googletalk%3aconf%2b...@conference.freeswitch.org
> ___
> 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
___
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
Re: [Freeswitch-users] Conference dialing and uuid
you should be able to use {} in the dial command
you also should be able to do
originate
{...}sofia/profile/u...@domain.comconference:@
inline
to the api interface
On Fri, Jan 30, 2009 at 7:33 AM, Sias Mey wrote:
> Hi Brian,
>
> Hmmm Ill do some more testing on it later. But I got a destination out
> of order when I tried. Right now Im busy implementing the string
> checking. Which seems like it will work out ok, but is clearly not
> ideal.
>
> Thanks for the replay
>
> On Fri, Jan 30, 2009 at 04:45:54AM -0600, Brian West wrote:
> > What wasn't working about this? The {} can be used everywhere without
> > a problem... Maybe you can provide more details on this.
> >
> > /b
> >
> >
> >
> >
> > On Jan 30, 2009, at 4:39 AM, Sias Mey wrote:
> >
> > >
> > > I couldent find a way of setting channel variables or executing
> > > javascript directly on the conference dial since it expects and
> > > endpoint
> > > and the {} syntax produced an error. So now I am using the Loopback
> > > inteface to register some values.
> >
> >
> > ___
> > 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
>
> ___
> 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
>
--
Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
AIM: anthm
MSN:anthony_miness...@hotmail.com
GTALK/JABBER/PAYPAL:anthony.miness...@gmail.com
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:8...@conference.freeswitch.org
iax:gu...@conference.freeswitch.org/888
googletalk:conf+...@conference.freeswitch.org
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
Re: [Freeswitch-users] Conference dialing and uuid
Hi Brian,
Hmmm Ill do some more testing on it later. But I got a destination out
of order when I tried. Right now Im busy implementing the string
checking. Which seems like it will work out ok, but is clearly not
ideal.
Thanks for the replay
On Fri, Jan 30, 2009 at 04:45:54AM -0600, Brian West wrote:
> What wasn't working about this? The {} can be used everywhere without
> a problem... Maybe you can provide more details on this.
>
> /b
>
>
>
>
> On Jan 30, 2009, at 4:39 AM, Sias Mey wrote:
>
> >
> > I couldent find a way of setting channel variables or executing
> > javascript directly on the conference dial since it expects and
> > endpoint
> > and the {} syntax produced an error. So now I am using the Loopback
> > inteface to register some values.
>
>
> ___
> 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
___
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
Re: [Freeswitch-users] Conference dialing and uuid
What wasn't working about this? The {} can be used everywhere without
a problem... Maybe you can provide more details on this.
/b
On Jan 30, 2009, at 4:39 AM, Sias Mey wrote:
>
> I couldent find a way of setting channel variables or executing
> javascript directly on the conference dial since it expects and
> endpoint
> and the {} syntax produced an error. So now I am using the Loopback
> inteface to register some values.
___
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
