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[peter.olsson at vision...]

Anthony,

Yes, it seems to work correct now. I did a couple of test calls, and tha audio was good – thanks!

Another question about this scenario...

When doing a session.transfer(”5000”), this will transfer the call directly into the dialplan without the use of loopback-channels. But that way it’s not possible to do it in a controlled way. Shouldn’t it be possible to do the same thing with a bridge? As soon as the call is bridged, it gets ”rid of” unneccecary loopback channels, and connecting the two endpoints directly – cause by then it should be two ”normal” endpoints talking?

Regards,

Peter

Från: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] För Anthony Minessale
Skickat: den 13 april 2009 20:38
Till: freeswitch-users@lists.freeswitch.org
Ämne: Re: [Freeswitch-users] Use of loopback channels and bridge() in scripts...


see how it works in latest trunk 13011

nontheless you can just say

session.execute("bridge", "loopback/5000");

and get the same result without touching that other channel.

when the call fails, you will have an originate_disposition variable in session you can check.



On Mon, Apr 13, 2009 at 11:21 AM, Peter Olsson <peter.olsson@visionutveckling.se (peter.olsson@visionutveckling.se)> wrote:
1. The latest trunk I've tried with is 13008. Since I'm not doing anything for production yet (just testing/evaluating), so I tend to update as soon as there is new version available..
2. Yep, you will find it below. In javascript - my sample for .NET does basically the same thing, with the same result, except that it also won't drop the loopback-a call leg.
3. Hmm.. Not really - I'm just in the middle of learning FS, so I guess I'm not 100% sure what I'm doing.. What I want to be able to do is to dial into a script, let the script dial another extension, and bridge them together when the other party answers the call. I also need to take care of call setup problems - if the other part doesn't respond, is unavailable or busy in the phone - so I though this was the only way? If I use the session.execute("bridge"..), will I be able to control the call if it couldn't be connected?

---

if (session.ready()) {

session.answer();

new_session = new Session("loopback/5000", session);
new_session.waitForAnswer();

bridge(session, new_session);

// Not sure if this is needed - I've tried with it both enabled and disabled
session.hangup();
new_session.hangup();
}

Peter


On 09-04-13 17.54, "Anthony Minessale" <anthony.minessale@gmail.com (anthony.minessale@gmail.com)> wrote:

1) When you say latest, which rev does that mean? we change revs pretty often.
2) Do you have a minimal script that reproduces your issue.
3) is there a reason you cannot just session.execute("bridge", dest);
instead of doing it manually (which is a process not for the faint at heart)?



On Mon, Apr 13, 2009 at 10:29 AM, Peter Olsson <peter.olsson@visionutveckling.se (peter.olsson@visionutveckling.se)> wrote:
I have two problems that I haven't been able to solve. I've done the same tests in both javascript, and in .NET.

The two scripts are pretty simple, they just answer an incomming call, creates a new session, wait for an answer on the second call leg, and then bridge the two channels together.

In both cases everything works just fine, but the audio is distorted. The destination I'm calling is "loopback/5000" - the sample IVR application included in FreeSWITCH. I first thought it was a codec issue, but even after trying to switch to different codecs the problem was the same. It more sounds like it's a timestamping issue - the voice is not distorted enough to be a bad codec, but it reads way to fast (mayby twice the "normal" speed). When doing a direct transfer() to the other destination this works just fine, but I need to be able to have some extra logic to tell if the destination is available or not.

The second problem occurs only in .NET. After doing this sample there is as loopback channel still hanging around. It seems like the call creates a loopback-a and loopback-b, the loopback-b dissapears as it should (when the call has been disconnected), but the other one stays there. When doing the same in javascript this doesn't seem to occur.

I'm using the latest SVN trunk, and my OS is Windows XP.

I found bug FSCORE-349 in Jira, which seems to point in to the direction that there might be a bug with the loopback channels in some cases, but I could not find anything about the audio which plays too fast.

Has anyone else experienced this?

Regards,

Peter Olsson

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--
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])
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[peter.olsson at vision...]

Yes, I’m starting to realize that... but you to get everything right – if I want to bridge a call, using the dialplan, then the only way is to use loopback, right? If I don’t want a loopback I’m able to bridge to the destination directly?

//Peter


Från: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] För Anthony Minessale
Skickat: den 14 april 2009 17:27
Till: freeswitch-users@lists.freeswitch.org
Ämne: Re: [Freeswitch-users] Use of loopback channels and bridge() in scripts...


yes,

But if you plan is to bridge the call, the loopback channel is completely unnecessary.
Be careful how much control you want =D getting a phone call up and running is more work
than you think (see switch_ivr_originate.c)


On Tue, Apr 14, 2009 at 8:24 AM, Peter Olsson <peter.olsson@visionutveckling.se (peter.olsson@visionutveckling.se)> wrote:
Anthony,

Yes, it seems to work correct now. I did a couple of test calls, and tha audio was good – thanks!

Another question about this scenario...

When doing a session.transfer(”5000”), this will transfer the call directly into the dialplan without the use of loopback-channels. But that way it’s not possible to do it in a controlled way. Shouldn’t it be possible to do the same thing with a bridge? As soon as the call is bridged, it gets ”rid of” unneccecary loopback channels, and connecting the two endpoints directly – cause by then it should be two ”normal” endpoints talking?

Regards,

Peter

Från: freeswitch-users-bounces@lists.freeswitch.org (freeswitch-users-bounces@lists.freeswitch.org) [mailto:freeswitch-users-bounces@lists.freeswitch.org (freeswitch-users-bounces@lists.freeswitch.org)] För Anthony Minessale
Skickat: den 13 april 2009 20:38
Till: freeswitch-users@lists.freeswitch.org (freeswitch-users@lists.freeswitch.org)
Ämne: Re: [Freeswitch-users] Use of loopback channels and bridge() in scripts...


see how it works in latest trunk 13011

nontheless you can just say

session.execute("bridge", "loopback/5000");

and get the same result without touching that other channel.

when the call fails, you will have an originate_disposition variable in session you can check.


On Mon, Apr 13, 2009 at 11:21 AM, Peter Olsson <peter.olsson@visionutveckling.se (peter.olsson@visionutveckling.se)> wrote:
1. The latest trunk I've tried with is 13008. Since I'm not doing anything for production yet (just testing/evaluating), so I tend to update as soon as there is new version available..
2. Yep, you will find it below. In javascript - my sample for .NET does basically the same thing, with the same result, except that it also won't drop the loopback-a call leg.
3. Hmm.. Not really - I'm just in the middle of learning FS, so I guess I'm not 100% sure what I'm doing.. What I want to be able to do is to dial into a script, let the script dial another extension, and bridge them together when the other party answers the call. I also need to take care of call setup problems - if the other part doesn't respond, is unavailable or busy in the phone - so I though this was the only way? If I use the session.execute("bridge"..), will I be able to control the call if it couldn't be connected?

---

if (session.ready()) {

session.answer();

new_session = new Session("loopback/5000", session);
new_session.waitForAnswer();

bridge(session, new_session);

// Not sure if this is needed - I've tried with it both enabled and disabled
session.hangup();
new_session.hangup();
}

Peter


On 09-04-13 17.54, "Anthony Minessale" <anthony.minessale@gmail.com (anthony.minessale@gmail.com)> wrote:

1) When you say latest, which rev does that mean? we change revs pretty often.
2) Do you have a minimal script that reproduces your issue.
3) is there a reason you cannot just session.execute("bridge", dest);
instead of doing it manually (which is a process not for the faint at heart)?



On Mon, Apr 13, 2009 at 10:29 AM, Peter Olsson <peter.olsson@visionutveckling.se (peter.olsson@visionutveckling.se)> wrote:
I have two problems that I haven't been able to solve. I've done the same tests in both javascript, and in .NET.

The two scripts are pretty simple, they just answer an incomming call, creates a new session, wait for an answer on the second call leg, and then bridge the two channels together.

In both cases everything works just fine, but the audio is distorted. The destination I'm calling is "loopback/5000" - the sample IVR application included in FreeSWITCH. I first thought it was a codec issue, but even after trying to switch to different codecs the problem was the same. It more sounds like it's a timestamping issue - the voice is not distorted enough to be a bad codec, but it reads way to fast (mayby twice the "normal" speed). When doing a direct transfer() to the other destination this works just fine, but I need to be able to have some extra logic to tell if the destination is available or not.

The second problem occurs only in .NET. After doing this sample there is as loopback channel still hanging around. It seems like the call creates a loopback-a and loopback-b, the loopback-b dissapears as it should (when the call has been disconnected), but the other one stays there. When doing the same in javascript this doesn't seem to occur.

I'm using the latest SVN trunk, and my OS is Windows XP.

I found bug FSCORE-349 in Jira, which seems to point in to the direction that there might be a bug with the loopback channels in some cases, but I could not find anything about the audio which plays too fast.

Has anyone else experienced this?

Regards,

Peter Olsson

_______________________________________________
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 (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
!DSPAM:49e4ade432931915915389!

[peter.olsson at vision...]

Allright – last question I’ll try to be a little more specific. Lets say I whant  to do the following;

<![if !supportLists]>1. <![endif]>Dial into FreeSWITCH, to some kind of application (javascript or whatever).
<![if !supportLists]>2. <![endif]>Answer that call, and let the user choose what to do; 1: record message, 2: transfer to XXX etc. The user presses 2.
<![if !supportLists]>3. <![endif]>I don’t want to release the first call leg yet, since I need to be really sure that 2 is reachable (or else I will give the user choices again, with som kind of ”the call could not be transferred”). So lets say I play some music for the user while trying to connect the call.
<![if !supportLists]>4. <![endif]>I originate another call – now I understand I have two choices, either I originate directly to a SIP phone (sofia/internal...), or I let the dialplan do the work – and if I want the dialplan to be the one to transfer the call somewhere (maybe to the same extension), I must use loopback – right?
<![if !supportLists]>5. <![endif]>If the new call answers, bridge the two calls, if it fails, start over again, after reading an error message.

Whould this also be possible with transfer? If I understand everything right I loose control of the call, and won’t be able to handle the failed transfer? Or is it possible to solve in a better way?

What I guess I’d really want to do is to ask the dialplan ”hey, I want to dial XXXX – give me the full sofia profile string” so I can originate the call directly, and I won’t need a loopback. I could of course connect to the sofia string directly, but it would be nice to leave that kind of lookup logic to the dialplan.

Thanks for staying with me – I hope you understand my problem

Regards,

Peter


Från: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] För Anthony Minessale
Skickat: den 14 april 2009 18:26
Till: freeswitch-users@lists.freeswitch.org
Ämne: Re: [Freeswitch-users] Use of loopback channels and bridge() in scripts...


The bridge application will let you bridge right to a destination on *another* box.
If you want to connect to a local extension like 5000 you can use the transfer application or method.

session.transfer("5000");
exit();

or

session.execute("transfer", "5000");
exit();



On Tue, Apr 14, 2009 at 10:59 AM, Peter Olsson <peter.olsson@visionutveckling.se (peter.olsson@visionutveckling.se)> wrote:
Yes, I’m starting to realize that... but you to get everything right – if I want to bridge a call, using the dialplan, then the only way is to use loopback, right? If I don’t want a loopback I’m able to bridge to the destination directly?

//Peter


Från: freeswitch-users-bounces@lists.freeswitch.org (freeswitch-users-bounces@lists.freeswitch.org) [mailto:freeswitch-users-bounces@lists.freeswitch.org (freeswitch-users-bounces@lists.freeswitch.org)] För Anthony Minessale
Skickat: den 14 april 2009 17:27

Till: freeswitch-users@lists.freeswitch.org (freeswitch-users@lists.freeswitch.org)
Ämne: Re: [Freeswitch-users] Use of loopback channels and bridge() in scripts...




yes,

But if you plan is to bridge the call, the loopback channel is completely unnecessary.
Be careful how much control you want =D getting a phone call up and running is more work
than you think (see switch_ivr_originate.c)
On Tue, Apr 14, 2009 at 8:24 AM, Peter Olsson <peter.olsson@visionutveckling.se (peter.olsson@visionutveckling.se)> wrote:
Anthony,

Yes, it seems to work correct now. I did a couple of test calls, and tha audio was good – thanks!

Another question about this scenario...

When doing a session.transfer(”5000”), this will transfer the call directly into the dialplan without the use of loopback-channels. But that way it’s not possible to do it in a controlled way. Shouldn’t it be possible to do the same thing with a bridge? As soon as the call is bridged, it gets ”rid of” unneccecary loopback channels, and connecting the two endpoints directly – cause by then it should be two ”normal” endpoints talking?

Regards,

Peter

Från: freeswitch-users-bounces@lists.freeswitch.org (freeswitch-users-bounces@lists.freeswitch.org) [mailto:freeswitch-users-bounces@lists.freeswitch.org (freeswitch-users-bounces@lists.freeswitch.org)] För Anthony Minessale
Skickat: den 13 april 2009 20:38
Till: freeswitch-users@lists.freeswitch.org (freeswitch-users@lists.freeswitch.org)
Ämne: Re: [Freeswitch-users] Use of loopback channels and bridge() in scripts...


see how it works in latest trunk 13011

nontheless you can just say

session.execute("bridge", "loopback/5000");

and get the same result without touching that other channel.

when the call fails, you will have an originate_disposition variable in session you can check.
On Mon, Apr 13, 2009 at 11:21 AM, Peter Olsson <peter.olsson@visionutveckling.se (peter.olsson@visionutveckling.se)> wrote:
1. The latest trunk I've tried with is 13008. Since I'm not doing anything for production yet (just testing/evaluating), so I tend to update as soon as there is new version available..
2. Yep, you will find it below. In javascript - my sample for .NET does basically the same thing, with the same result, except that it also won't drop the loopback-a call leg.
3. Hmm.. Not really - I'm just in the middle of learning FS, so I guess I'm not 100% sure what I'm doing.. What I want to be able to do is to dial into a script, let the script dial another extension, and bridge them together when the other party answers the call. I also need to take care of call setup problems - if the other part doesn't respond, is unavailable or busy in the phone - so I though this was the only way? If I use the session.execute("bridge"..), will I be able to control the call if it couldn't be connected?

---

if (session.ready()) {

session.answer();

new_session = new Session("loopback/5000", session);
new_session.waitForAnswer();

bridge(session, new_session);

// Not sure if this is needed - I've tried with it both enabled and disabled
session.hangup();
new_session.hangup();
}

Peter


On 09-04-13 17.54, "Anthony Minessale" <anthony.minessale@gmail.com (anthony.minessale@gmail.com)> wrote:

1) When you say latest, which rev does that mean? we change revs pretty often.
2) Do you have a minimal script that reproduces your issue.
3) is there a reason you cannot just session.execute("bridge", dest);
instead of doing it manually (which is a process not for the faint at heart)?



On Mon, Apr 13, 2009 at 10:29 AM, Peter Olsson <peter.olsson@visionutveckling.se (peter.olsson@visionutveckling.se)> wrote:
I have two problems that I haven't been able to solve. I've done the same tests in both javascript, and in .NET.

The two scripts are pretty simple, they just answer an incomming call, creates a new session, wait for an answer on the second call leg, and then bridge the two channels together.

In both cases everything works just fine, but the audio is distorted. The destination I'm calling is "loopback/5000" - the sample IVR application included in FreeSWITCH. I first thought it was a codec issue, but even after trying to switch to different codecs the problem was the same. It more sounds like it's a timestamping issue - the voice is not distorted enough to be a bad codec, but it reads way to fast (mayby twice the "normal" speed). When doing a direct transfer() to the other destination this works just fine, but I need to be able to have some extra logic to tell if the destination is available or not.

The second problem occurs only in .NET. After doing this sample there is as loopback channel still hanging around. It seems like the call creates a loopback-a and loopback-b, the loopback-b dissapears as it should (when the call has been disconnected), but the other one stays there. When doing the same in javascript this doesn't seem to occur.

I'm using the latest SVN trunk, and my OS is Windows XP.

I found bug FSCORE-349 in Jira, which seems to point in to the direction that there might be a bug with the loopback channels in some cases, but I could not find anything about the audio which plays too fast.

Has anyone else experienced this?

Regards,

Peter Olsson

_______________________________________________
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




_______________________________________________
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Freeswitch-users@lists.freeswitch.org (Freeswitch-users@lists.freeswitch.org)
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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





--
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)
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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
!DSPAM:49e4bcb132932104520616!