Re: [Freeswitch-users] Passing user variables to mod_voicemail

[Phillip Jones]

Hi - sorry to go off topic - but we are looking for Voip supplier with SMS
capability. Would you mind telling me which Voip supplier you use?

On Thu, Dec 10, 2009 at 11:10 PM, Mark Campbell-Smith <
mcampbellsm...@gmail.com> wrote:

> Hi!
>
> My voip provider provides a SOAP interface to be able to send SMS's,
> so after a voicemail is left, I want to execute a 'send sms' script.
> I don't want a separate statement in the dialplan after the voicemail
> statement because I only want to send sms's when a voicemail is
> actually left.
>
> The way I was going to do this was to modify the mailer-app to point
> to a shell script and modify the mailer-app-args to include some user
> defined variables (in conf/directory/default/*.xml).
>
> value="/usr/local/freeswitch/scripts/emailvm.sh"/>
>
>
> The shell script would do the following:
>
> emailvm.sh
>
> #$1 $2 $3 = smsaccount smspassword textmessage
> tee /tmp/vmmail | /usr/sbin/sendmail -t
> exec /usr/local/freeswitch/scripts/sendsms.pl $1 $2 $3
> #echo $1 $2 $3 $4 $5 $6 >> /usr/local/freeswitch/scripts/log.log
>
> However, if I uncomment the last line, I never see the user variables
> being passed to the shell script.  The email is sucessfully sent, but
> the sms script doesnt work.  If fact, the output of log.log is (for
> example):
>
> -f 1...@192.168.1.120 email_addr...@domain.com
>
> Any ideas if it is possible to pass user variables via mod_voicemail
> in this way?
>
> Thanks
>
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Re: [Freeswitch-users] Setting up a FreeSwitch system on a pfSense firewall???

[Phillip Jones]

Also check out this great write up:

http://wiki.freeswitch.org/wiki/Multi_home_tutorial

This is pfSense specific.

On Sat, Dec 5, 2009 at 10:22 AM, ram  wrote:

>
>
> On Sat, Dec 5, 2009 at 3:16 PM, mailinglist wrote:
>
>>  Has anybody done this?
>>
>> I'm completely at a loss, having tinkered very little with Asterisk, and
>> giving up on that, I wonder if there's any help to be found on FreeSwitch?
>> Anybody that can give pointers to a good step-by-step instruction?
>>
>> I want to have it handle my two sip-phones (siemens dect ip and spa 901),
>> and handle a sip account at my provider.
>> Of course transferring calls between the two, as well as group calls would
>> be a nice benefit.
>>
>>
>>
>
> in short answer Fusionpbx.com
>
> Ram
>
>
>
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Re: [Freeswitch-users] Bridging to a non SIP based system

[Phillip Jones]

Ah guys - that was exactly the nudge I was looking for - I will take a look
at the other endpoint modules like mod_skypiax etc. I will also look at the
SDP - I see where you are going there - I might not even need the conference
in that case.

Question is - could I write an endpoint is C# !!!  :)

Thanks again - that's a great help.

On Fri, Dec 4, 2009 at 5:16 PM, Michael Giagnocavo wrote:

> I think you will need to sort out the signaling first, as you’ll have to
> tell the conference system to accept which RTP streams for which
> conferences, as well as tell it to transmit to your callers, no?
>
>
>
> After that, then I would imagine you just need to do SDP rewriting when a
> call hits FreeSWITCH.
>
>
>
> -Michael
>
>
>
> *From:* freeswitch-users-boun...@lists.freeswitch.org [mailto:
> freeswitch-users-boun...@lists.freeswitch.org] *On Behalf Of *Phillip
> Jones
> *Sent:* Friday, December 04, 2009 2:29 PM
> *To:* freeswitch-users@lists.freeswitch.org
> *Subject:* [Freeswitch-users] Bridging to a non SIP based system
>
>
>
> Hi All,
>
> Every so often you have to ask a question - where you know so little - it's
> hard to even now where to start. This is one of the times. I am not
> expecting an full answer here, just a gentle nudge in right direction to get
> me started.
>
> What I have is a propriety IP based conference system - who want to add the
> ability to have inbound PSTN callers join their conferences. All their
> signaling is propriety - no SIP - but I do have access to that signaling
> schema so can do some translation. Enough to get the IP / Port & CODEC of
> the RTP stream. They use speex rtp sessions over TCP.
>
> So from an architectural point of view I am thinking of having the callers
> enter a FS conference and than bridge that conference to their IP based
> conference room. That would do it.
>
> The problem is that because I can not bridge using SIP (through a Sofia
> gateway) to that IP based conference system I am kind of lost. But it seems
> reasonable that I should be able to get my head round this, because I know
> the IP / Port & CODEC of the RTP stream.
>
> But perhaps I missing a key bit of knowledge/understanding here.
>
> I would be grateful for any advise here.
>
> Thanks a lot,
>
>
> Phil
>
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[Freeswitch-users] Bridging to a non SIP based system

[Phillip Jones]

Hi All,

Every so often you have to ask a question - where you know so little - it's
hard to even now where to start. This is one of the times. I am not
expecting an full answer here, just a gentle nudge in right direction to get
me started.

What I have is a propriety IP based conference system - who want to add the
ability to have inbound PSTN callers join their conferences. All their
signaling is propriety - no SIP - but I do have access to that signaling
schema so can do some translation. Enough to get the IP / Port & CODEC of
the RTP stream. They use speex rtp sessions over TCP.

So from an architectural point of view I am thinking of having the callers
enter a FS conference and than bridge that conference to their IP based
conference room. That would do it.

The problem is that because I can not bridge using SIP (through a Sofia
gateway) to that IP based conference system I am kind of lost. But it seems
reasonable that I should be able to get my head round this, because I know
the IP / Port & CODEC of the RTP stream.

But perhaps I missing a key bit of knowledge/understanding here.

I would be grateful for any advise here.

Thanks a lot,


Phil
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[Freeswitch-users] Playing an rtp stream

[Phillip Jones]

Hi there,

It it possible do something like:


  


  



Basically I have need to connect to incoming calls listen to an existing rtp
stream - I know the IP and port.

Any hints on achieving this would be much appreciated.

Thanks


Phil
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Re: [Freeswitch-users] Holiday routing examples

[Phillip Jones]

Thanks for this goodness. I am sure to use it so it is appreciated.

On Mon, Nov 30, 2009 at 2:51 PM, Andrew Thompson  wrote:

> Tony committed my patch for doing 'week of month' conditions in the XML
> dialplan along with some holiday routing examples to the default
> dialplan. Now you can detect all the major US holidays in pure dialplan
> XML without having to do any nasty math or anything (I did it all for
> you).
>
> I've also added a page to the wiki describing how to use it for other
> dates (like non-US holidays):
>
> http://wiki.freeswitch.org/wiki/Holiday_Routing
>
> Hope this helps some people.
>
> Andrew
>
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[Freeswitch-users] register timeout / cisco 7960

[Phillip Jones]

hi there,

I have set up some cisco 7960 up with fs. They work fine - but the only way
I can keep them registered is to set the "timer_register_expires" in the
Cisco cfg file to something really short like 10s.

Does anyone know the default register timeout for fs? And where I might
change this in fs?

Thanks!


Phil
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Re: [Freeswitch-users] Simplest of Conference Setup questions

[Phillip Jones]

Anthony - setting



or



does not make a difference, even when the default profile has



un-commented.


Looks to me like that default group is ignored even when specifically
referred to?

As Michael says though, creating a specific group:



and adding

   in the default profile
works a charm.

I am good - but let me know if you want me to try anything else.

Phil



On Mon, Nov 23, 2009 at 3:27 PM, Michael Collins  wrote:

>
>
> On Mon, Nov 23, 2009 at 12:17 PM, Phillip Jones wrote:
>
>> Thanks for replying.
>>
>> Well in the log I see:
>>
>> 2009-11-23 15:13:22.015625 [DEBUG] switch_rtp.c:2282 RTP RECV DTMF #:760
>> 2009-11-23 15:13:22.062500 [DEBUG] mod_conference.c:2379 Channel leaving
>> conference, cause: NONE
>>
>> which make sense because just above I see:
>>
>> 009-11-23 15:13:08.171875 [DEBUG] mod_conference.c:5508 Installing default
>> caller control action 'hangup' bound to '#'.
>>
>> The question I have - is how do I change that default caller control
>> action if it is not in conference.conf.xml ??
>>
>> 
>> 
>> ...
>>
>> **
>>
>
> I believe that this is because the caller-controls param is commented out
> in the default profile config. I prefer not to mess w/ the default configs
> which is why I recommended the custom configs in my previous email...
>
> -MC
>
>
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Re: [Freeswitch-users] Simplest of Conference Setup questions

[Phillip Jones]

Michael that for the reply.

I created a new group with # unbound and referenced it from the default
profile:

  

And that worked fine.

Strangely though, changing the default group and referencing that from the
default profile does not.




Do you want me to test this on the latest trunk or is this as expected?

Phil


On Mon, Nov 23, 2009 at 3:12 PM, Michael Collins  wrote:

>
>
> On Mon, Nov 23, 2009 at 11:24 AM, Phillip Jones wrote:
>
>> Hi there,
>>
>> I have created a simple conference that works great. The only problem is,
>> when a participant press # it exits the call. So when a user enters a
>> conference with a PIN, and by habit they enter 12345 followed by pound, it
>> puts them in and then straight out.
>>
>> So I edited conference.conf.xml so:
>>
>> 
>>
>> and even assigned # to another function:
>>
>> 
>>
>> and the same occurs. Pressing # exits the conference.
>>
>> What am I missing here?
>>
>> tia - phil
>>
>>
>>
> Phil,
>
> I recommend that you create a custom profile and a custom caller control
> group. Just copy the defaults and rename them to something meaningful. In
> conference.conf.xml you can add a new call control group like this:
>
>   
>digits="0"/>
>
>digits="*"/>
>
>digits="9"/>
>
>digits="8"/>
>
>digits="7"/>
>
>digits="3"/>
>
>digits="2"/>
>
>digits="1"/>
>
>digits="6"/>
>
>digits="5"/>
>
>digits="4"/>
>
>   
> 
>
> Then make a copy of the default profile changing the profile name and the
> caller-controls parameter:
>  
>  
>
>
>  
>  
>
>
> 
>
> Give that a whirl and report back. :)
> -MC
>
>
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Re: [Freeswitch-users] Simplest of Conference Setup questions

[Phillip Jones]

Thanks for replying.

Well in the log I see:

2009-11-23 15:13:22.015625 [DEBUG] switch_rtp.c:2282 RTP RECV DTMF #:760
2009-11-23 15:13:22.062500 [DEBUG] mod_conference.c:2379 Channel leaving
conference, cause: NONE

which make sense because just above I see:

009-11-23 15:13:08.171875 [DEBUG] mod_conference.c:5508 Installing default
caller control action 'hangup' bound to '#'.

The question I have - is how do I change that default caller control action
if it is not in conference.conf.xml ??



...
**


On Mon, Nov 23, 2009 at 2:35 PM, Anthony Minessale <
anthony.miness...@gmail.com> wrote:

> issue
>
> console loglevel debug
> from the cli
>
> then try again and see if there is any hint
>
>
> On Mon, Nov 23, 2009 at 1:24 PM, Phillip Jones wrote:
>
>> Hi there,
>>
>> I have created a simple conference that works great. The only problem is,
>> when a participant press # it exits the call. So when a user enters a
>> conference with a PIN, and by habit they enter 12345 followed by pound, it
>> puts them in and then straight out.
>>
>> So I edited conference.conf.xml so:
>>
>> 
>>
>> and even assigned # to another function:
>>
>> 
>>
>> and the same occurs. Pressing # exits the conference.
>>
>> What am I missing here?
>>
>> tia - phil
>>
>>
>>
>> Conf Setup:
>>
>> 
>> 
>> 
>> 
>> 
>>   
>>
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>>
>
>
> --
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>
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[Freeswitch-users] Simplest of Conference Setup questions

[Phillip Jones]

Hi there,

I have created a simple conference that works great. The only problem is,
when a participant press # it exits the call. So when a user enters a
conference with a PIN, and by habit they enter 12345 followed by pound, it
puts them in and then straight out.

So I edited conference.conf.xml so:



and even assigned # to another function:



and the same occurs. Pressing # exits the conference.

What am I missing here?

tia - phil



Conf Setup:






  
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Re: [Freeswitch-users] Accessing Config Info From Database

[Phillip Jones]

Take a look at http://wiki.freeswitch.org/wiki/Mod_xml_curl to get started.


On Fri, Nov 13, 2009 at 4:59 PM, Jerry Richards
wrote:

> Is there a way to access configuration information from a database (e.g.
> SQL) rather than from the XML files?
>
> Best Regards,
> Jerry
>
>
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Re: [Freeswitch-users] skill-based ACD

[Phillip Jones]

I would be interested in this too

Concerning mod_fifo - can you restrict an agent to a slot. So lets say DNIS
A is for product A and DNIS B product B - some agents know both - some know
just 1 product  -  would that be possible?

On Thu, Sep 17, 2009 at 3:17 PM, Anthony Minessale <
anthony.miness...@gmail.com> wrote:

> I can tell you from years of painful experience, don't use asterisk for
> queues.
> see http://www.freeswitch.org/node/117
>
> You don't have to use FS, but please don't let the asterisk siren lure you
> to the rocks.
>
> mod_fifo is like a tool with basic functions you can exploit however you
> wish, it does not try to do high level
> features because those are best left in external logic.
>
>
> mod_fifo has priorities which means each individual fifo is really an array
> of 10 fifos
> when you set the priority you are choosing which index in the array to
> insert the caller.
> when an agent belongs to a queue he drills down the array from 0-9 so you
> could for instance put everyone in 5 by default and put more
> important people in 0 so they always go to the front
>
> when you assign an agent to take calls off hook you can set a
> fifo_pop_order variable that tells you which array indexes to service and in
> what order.
> so if you pretend slot 1 is for general problems and slot 2 is for hard
> problems you can put one agent in 1,2 and a more stupid agent in just 1
>
> *shrug*
>
>
>
> On Thu, Sep 17, 2009 at 1:56 PM, Christian Jensen <
> christian.jen...@teligence.net> wrote:
>
>>  This would be a fantastic addition – my company is currently looking to
>> Asterisk as a potential candidate for this if FS can’t do it.
>>
>>
>>
>> I want FS to win of course J
>>
>>
>>
>> *Christian Jensen*
>> Software Development Manager
>>
>> Back Office
>>   --
>>
>> *From:* freeswitch-users-boun...@lists.freeswitch.org [mailto:
>> freeswitch-users-boun...@lists.freeswitch.org] *On Behalf Of *Michael
>> Collins
>> *Sent:* Thursday, September 17, 2009 11:20 AM
>> *To:* freeswitch-users@lists.freeswitch.org
>> *Subject:* Re: [Freeswitch-users] skill-based ACD
>>
>>
>>
>>
>>
>> On Sun, Sep 13, 2009 at 8:01 AM, mark morreny 
>> wrote:
>>
>> Hello
>>
>>
>>
>> Has any tried setting up an ACD based on skillset?  The current out-of-box
>> version of fifo does not seem to support acd based on agent skillset.  Does
>> anyone have any experience in doing it with some external scripting using
>> lua or javascript?
>>
>>
>>
>> I am interested in hearing how others may have done it as I am trying to
>> implement one myself.
>>
>>
>>
>> thx,
>>
>>
>>
>> mark
>>
>>
>>
>>
>> I was curious about this myself. Even if someone has built a non-free
>> skills-based ACD using FS I'd like to know about it.
>> -MC
>>
>>
>>
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>
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Re: [Freeswitch-users] Callback in Javascript, session.destroy() does not free the channel!

[Phillip Jones]

session.dispose();

???

On Thu, Sep 17, 2009 at 5:20 PM, Alberto Escudero  wrote:

> We are trying to create a callback application in Javascript. We get the
> callerid from the unanswered call and after destroying the session, we
> initiate a callback to the user to conenct it to a local extension in the
> dialplan.
>
> Although we have tried to destroy the first session, or even invoke a
> second script using apiExecute("jsrun",dialer.js"), tried session.hangup()
> or exit()... the first session does not seem to close properly until the
> whole chain of scripts are completed.
>
> Here is a piece of code that shows the concept (yes!, the sleep function
> is far from ideal. CPU loves it! )
>
> function sleep(milliseconds) {
>  var start = new Date().getTime();
>  for (var i = 0; i < 1e7; i++) {
>if ((new Date().getTime() - start) > milliseconds){
>  break;
>}
>  }
> }
>
> if (session.ready()) {
>//We catch the caller_id
>caller_id_num = session.caller_id_num;
>
>console_log("Now we got your Caller ID\n");
>
>//How long we want to wait to trigger a call back
>session.execute("sleep",5000);
>
>console_log("We have waited a while... time to create the
> callback\n");
>
>//apiExecute("jsrun", "callback.js");
>}
>
> //Destroy the session...
> session.destroy();
> session=undefined;
>
> sleep(1);
>
> //Preparing callback
> session2 = new
> Session('{ignore_early_media=true}celliax/interface1/600464646');
> session2.setAutoHangup(false);
> session2.answer();
> exit();
>
> ++
> Wisdom thoughts?
>
> --
> Stopping junk mailers is good for the environment
>
>
>
>
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Re: [Freeswitch-users] skill-based ACD

[Phillip Jones]

see here: *http://en.wikipedia.org/wiki/Skills_based_routing*

On Thu, Sep 17, 2009 at 4:42 PM, Phillip Jones  wrote:

> >> My FS based ACD allows agents to log-in to multiple queues at once
>
> What is the difference between those queues? Does each q require that
> agents have a different skill? ie Agent A has skills x,y,z and queue 1,2,3
> have calls about x,y,z - there agent A logins to queue 1,2,3.
>
> 2009/9/17 Paweł Pierścionek 
>
> Michael Collins pisze:
>> >
>> >
>> > On Sun, Sep 13, 2009 at 8:01 AM, mark morreny > > <mailto:markmorr...@gmail.com>> wrote:
>> >
>> > Hello
>> >
>> > Has any tried setting up an ACD based on skillset?  The current
>> > out-of-box version of fifo does not seem to support acd based on
>> > agent skillset.  Does anyone have any experience in doing it with
>> > some external scripting using lua or javascript?
>> >
>> > I am interested in hearing how others may have done it as I am
>> > trying to implement one myself.
>> >
>> > thx,
>> >
>> > mark
>> >
>> >
>> > I was curious about this myself. Even if someone has built a non-free
>> > skills-based ACD using FS I'd like to know about it.
>> > -MC
>> >
>> What is a skills-based ACD ?
>> My FS based ACD allows agents to log-in to multiple queues at once and
>> have different priority setting in each of the queues - does this count
>> as skills-based ?
>> It does not allow to limit the number of calls an agent can answer from
>> a queue daily nor does it allow to set distribution of calls to an agent
>> from his queues (other the by priority).
>>
>> Pawel,
>>
>>
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Re: [Freeswitch-users] skill-based ACD

[Phillip Jones]

>> My FS based ACD allows agents to log-in to multiple queues at once

What is the difference between those queues? Does each q require that agents
have a different skill? ie Agent A has skills x,y,z and queue 1,2,3 have
calls about x,y,z - there agent A logins to queue 1,2,3.

2009/9/17 Paweł Pierścionek 

> Michael Collins pisze:
> >
> >
> > On Sun, Sep 13, 2009 at 8:01 AM, mark morreny  > > wrote:
> >
> > Hello
> >
> > Has any tried setting up an ACD based on skillset?  The current
> > out-of-box version of fifo does not seem to support acd based on
> > agent skillset.  Does anyone have any experience in doing it with
> > some external scripting using lua or javascript?
> >
> > I am interested in hearing how others may have done it as I am
> > trying to implement one myself.
> >
> > thx,
> >
> > mark
> >
> >
> > I was curious about this myself. Even if someone has built a non-free
> > skills-based ACD using FS I'd like to know about it.
> > -MC
> >
> What is a skills-based ACD ?
> My FS based ACD allows agents to log-in to multiple queues at once and
> have different priority setting in each of the queues - does this count
> as skills-based ?
> It does not allow to limit the number of calls an agent can answer from
> a queue daily nor does it allow to set distribution of calls to an agent
> from his queues (other the by priority).
>
> Pawel,
>
>
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Re: [Freeswitch-users] Pastebin Username/Password Not Accepted

[Phillip Jones]

it's a rite of passage  :)

On Mon, Sep 14, 2009 at 11:29 AM, Jerry Richards  wrote:

>
> Aha... I have been notified that I failed the test.  The username/password
> is given in the authentication pop-up itself.  My bad...
>
>
>
> -Original Message-
> From: Jerry Richards [mailto:jerry.richa...@teotech.com]
> Sent: Monday, September 14, 2009 8:13 AM
> To: 'freeswitch-users@lists.freeswitch.org'
> Subject: Pastebin Username/Password Not Accepted
>
> What account do I need to create to post logs in the Pastebin?  I tried my
> mailing list username/password, and also tried a jira.freeswitch.org
> username/password.  Neither of these were accepted.
>
> Best Regards,
> Jerry
>
>
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Re: [Freeswitch-users] uuid_exists - does it still exist?

[Phillip Jones]

Strangely - the uuid_getvar  uuid workaround does
not work for me.

This is the result of:

apiResult = fsApi.ExecuteString(string.Format("uuid_getvar {0} uuid",
call.Uuid));
Log.WriteLine(LogLevel.Alert, "RESULT: uuid_getvar {0} is: {1}", call.Uuid,
apiResult);

returns a different uuid?

2009-09-10 17:12:26.953125 [ALERT] switch_cpp.cpp:1130 RESULT: uuid_getvar
54dd24be-b0da-684f-acee-38c7530b4c2b is:
1a0e83db-240c-ac4e-ae45-bf5d5b46f5c3

the passed uuid is vaild however:

2009-09-10 17:12:26.953125 [ALERT] switch_cpp.cpp:1130 RESULT: uuid_kill
54dd24be-b0da-684f-acee-38c7530b4c2b is: +OK

Can a call leg have two uuids??


On Thu, Sep 3, 2009 at 5:56 PM, Michael Collins  wrote:

>
>
> On Thu, Sep 3, 2009 at 2:23 PM, Benedikt Fraunhofer <
> fraunhofer.lists.freeswitch-...@traced.net> wrote:
>
>> Hi,
>>
>>
>> > Usage: uuid_exists 
>> > However when I call via an API call I get:
>> > INVALID COMMAND!
>> > I also don't see it in MOD_COMMAND.C
>>
>> As a workaround or if your are unable to upgrade, you can use
>> "uuid_getvar [some_uuid] thisVariableDoesNotExist"
>> ("thisVariableDoesNotExist" is any variable you can think of, a valid
>> one or literally 'thisVariableDoesNotExist' :)
>> You'll either get an error that this channel does not exist any longer
>> or "undef" for the channel variable.
>> The api will return "+OK" in case the channel still exists, and
>> "ERROR" in case it does not.
>>
>> Beni.
>>
>>
> You could even do this:
> uuid_getvar  uuid
>
> If it exists then the return will be the uuid. :)
>
> Although I must say I recommend this instead:
> make current
>
> :)
> -MC
>
>
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Re: [Freeswitch-users] Subscribing to events in managed C# / .NET

[Phillip Jones]

Surely this is because you are calling ev.serialized_string and ev is null
if there is not event waiting?
i.e.  con.pop(0); returns null if no event is waiting.

Therefore

if(ev != null)
  Log.WriteLine(LogLevel.Notice, "Event: " + ev.serialized_string);

would work, alternatively calling

Event ev = con.pop(1);

would block until an event is available - and therefore ev cannot be
null



On Wed, Sep 9, 2009 at 2:01 PM, Josh Rivers  wrote:

> A new discovery:public bool Load()
> {
> ThreadPool.QueueUserWorkItem((o) =>
> {
> Log.WriteLine(LogLevel.Notice, "Thread Starting. ");
> EventConsumer con = new EventConsumer("all", "");
> while (true)
> {
> Event ev = con.pop(0);
> if (ev == null) continue;
> Log.WriteLine(LogLevel.Notice, "Event: " +
> ev.serialized_string);
> }
> });
> return true;
> }
> Does not crash. (Adding the null check prevents crash.) The backgrounded
> loop runs fine. Looks like the event object goes straight to pinvokes, so a
> null result just crashes?
>
> I like the idea of a 'startup-script' for mod_managed. It would also be
> excellent if there was an event or message  informing the background code to
> terminate nicely when the module reloads.
>
> --Josh
>
> On Wed, Sep 9, 2009 at 12:57 PM, Jeff Lenk  wrote:
>
>>
>> I think the problem here is that the loader only keeps this method in
>> scope
>> until completion then it drops the remoted connection. Therefore you
>> should
>> not use threads in this method. Michael please correct me if I am wrong
>> here.
>>
>> As an example of the failure simply just put a Sleep(1) call in the
>> thread and you will see the failure.
>>
>> As Michael said this method was only designed to allow the option to opt
>> out
>> of being loaded.
>>
>> In order to support this perhaps a configuration flag simular to the lua
>> "startup-script" should be added.
>>
>>
>>
>> Here is the error I get with the loop I mentioned. -Josh
>> [image: Capture.PNG]
>>
>> On Tue, Sep 8, 2009 at 5:05 AM, Michael Giagnocavo
>> wrote:
>>
>> >  Hi,
>> >
>> >
>> >
>> > Can you please elaborate on the crash you receive when
>> you
>> > queue a thread during load?
>> >
>> >
>> >
>> > Thanks,
>> >
>> > Michael
>> >
>> >
>>
>> --
>> View this message in context:
>> http://n2.nabble.com/Subscribing-to-events-in-managed-C-NET-tp3573619p3613195.html
>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>
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Re: [Freeswitch-users] Subscribing to events in managed C# / .NET

[Phillip Jones]

What is:

freeswitch.msleep(100);

Why aren't you using Thread.Sleep?

On Tue, Sep 8, 2009 at 2:21 AM, Josh Rivers  wrote:

> Thanks for the response!
> I have tried putting a long-running loop here, but then it blocks anything
> else managed from happening:
>
>public class TestLoop : ILoadNotificationPlugin
> {
> public bool Load()
> {
> EventConsumer con = new EventConsumer("all", "");
> while (true)
> {
> Event ev = con.pop(0);
> Log.WriteLine(LogLevel.Notice, "Event: " +
> ev.serialized_string);
> freeswitch.msleep(100);
> }
> }
> }
>
> However, if I fork off a thread here, freeswitch crashes:
> public class TestLoop : ILoadNotificationPlugin
> {
> public bool Load()
> {
> ThreadPool.QueueUserWorkItem((o) =>
> {
> Log.WriteLine(LogLevel.Notice, "Thread Starting. ");
> EventConsumer con = new EventConsumer("all", "");
> while (true)
> {
> Event ev = con.pop(0);
> Log.WriteLine(LogLevel.Notice, "Event: " +
> ev.serialized_string);
> freeswitch.msleep(100);
> }
> });
> return true;
> }
> }
>
> It doesn't look like this is a good place to start a long-running process?
>
> Thanks!
> Josh
>
> On Mon, Sep 7, 2009 at 11:05 PM, Raffaele P. Guidi <
> raffaele.p.gu...@gmail.com> wrote:
>
>> Yes!
>> public class LoadDemo : ILoadNotificationPlugin {
>> public bool Load() {
>> Log.WriteLine(LogLevel.Notice, "LoadDemo running.");
>> return true;
>> }
>> }
>>
>> this example is from Michael Giagnocavo's Demo.csx which you can find into
>> the mod_managed svn.
>>
>> And let me add that works like a charm :)
>>
>> Ciao,
>>Raffaele
>>
>> On Sun, Sep 6, 2009 at 22:50, Josh Rivers  wrote:
>>
>>> Is there a way to start this when FreeSWITCH starts? The lua and perl
>>> modules have a 'startup-script' configuration preference. Is there something
>>> similar in mod_managed? Or is there a way to have an api command executed at
>>> a startup?
>>>
>>> 
>>> Exactly what I was after - thank you!
>>>
>>> On Thu, Sep 3, 2009 at 1:54 PM, Jeff Lenk  wrote:
>>>
>>> >
>>> > try something like this
>>> >
>>> > EventConsumer con = new EventConsumer("all", "");
>>> > Event ev = con.pop(0);
>>> >
>>> > see lua sample -
>>> > http://wiki.freeswitch.org/wiki/Lua#freeswitch.EventConsumer
>>> >
>>> >
>>> > Phillip Jones-2 wrote:
>>> > >
>>> > > Hi there,
>>> > >
>>> > > mod_managed exposes EventReceivedFunction such that:
>>> > >
>>> > >  Session.EventReceivedFunction = (e) =>
>>> > >  {
>>> > >Log.WriteLine(LogLevel.Alert, "Received Event {0}",
>>> e.ToString());
>>> > >return "";
>>> > >  };
>>> > >
>>> > > should trap all events to which i subscribe.
>>> > >
>>> > >
>>> > > But how do I subscribe to events? What is the .NET / managed
>>> equivalent
>>> > > of:
>>> > >
>>> > > switch_event_bind(const char *id, switch_event_types_t event, const
>>> char
>>> > > *subclass_name, switch_event_callback_t callback, void *user_data);
>>> > >
>>> > >
>>> > >
>>> > > Thank you!
>>> > >
>>> > >
>>> > >
>>>
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>>>
>>
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[Freeswitch-users] uuid_exists - does it still exist?

[Phillip Jones]

Hi there:

The wiki states:

uuid_exists

Checks whether a given UUID exists.

Usage: uuid_exists 


However when I call via an API call I get:

INVALID COMMAND!

I also don't see it in MOD_COMMAND.C


Has it gone?


Thanks


Phillip Jones.
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Re: [Freeswitch-users] Subscribing to events in managed C# / .NET

[Phillip Jones]

Exactly what I was after - thank you!

On Thu, Sep 3, 2009 at 1:54 PM, Jeff Lenk  wrote:

>
> try something like this
>
> EventConsumer con = new EventConsumer("all", "");
> Event ev = con.pop(0);
>
> see lua sample -
> http://wiki.freeswitch.org/wiki/Lua#freeswitch.EventConsumer
>
>
> Phillip Jones-2 wrote:
> >
> > Hi there,
> >
> > mod_managed exposes EventReceivedFunction such that:
> >
> >  Session.EventReceivedFunction = (e) =>
> >  {
> >Log.WriteLine(LogLevel.Alert, "Received Event {0}", e.ToString());
> >return "";
> >  };
> >
> > should trap all events to which i subscribe.
> >
> >
> > But how do I subscribe to events? What is the .NET / managed equivalent
> > of:
> >
> > switch_event_bind(const char *id, switch_event_types_t event, const char
> > *subclass_name, switch_event_callback_t callback, void *user_data);
> >
> >
> >
> > Thank you!
> >
> >
> >
> > Phillip Jones
> >
> > ___
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> >
> >
>
> --
> View this message in context:
> http://n2.nabble.com/Subscribing-to-events-in-managed-C-NET-tp3573619p3574945.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
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Re: [Freeswitch-users] Subscribing to events in managed C# / .NET

[Phillip Jones]

Where can I find EventConsumer?? I search the wiki and contrib\

Thanks

On Thu, Sep 3, 2009 at 11:12 AM, Mathieu Rene  wrote:

> Check out EventConsumer
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mr...@avgs.ca
>
>
>
>
> On 3-Sep-09, at 8:05 AM, Michael Giagnocavo wrote:
>
> You can call switch_event_bind directly, but I doubt that’ll achieve what
> you want.
>
> My guess is that it works the same as LUA or any of the other plugins –
> whatever higher level API is exposed for setting up events should make
> things just work.
>
> -Michael
>
> *From:* freeswitch-users-boun...@lists.freeswitch.org [
> mailto:freeswitch-users-boun...@lists.freeswitch.org
> ] *On Behalf Of *Phillip Jones
> *Sent:* Thursday, September 03, 2009 8:27 AM
> *To:* freeswitch-users@lists.freeswitch.org
> *Subject:* [Freeswitch-users] Subscribing to events in managed C# / .NET
>
> Hi there,
>
> mod_managed exposes EventReceivedFunction such that:
>
>  Session.EventReceivedFunction = (e) =>
>  {
>Log.WriteLine(LogLevel.Alert, "Received Event {0}", e.ToString());
>return "";
>  };
>
> should trap all events to which i subscribe.
>
>
> But how do I subscribe to events? What is the .NET / managed equivalent of:
>
> switch_event_bind(const char *id, switch_event_types_t event, const char
> *subclass_name, switch_event_callback_t callback, void *user_data);
>
>
>
> Thank you!
>
>
>
> Phillip Jones
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Re: [Freeswitch-users] 2009-09-02 15:21:44.665608 [CRIT] switch_time.c:454 Reverse Clock Skew Detected!

[Phillip Jones]

Yeah - temporal worm holes and the like are my other project :) Or may be
its the full moon.

Thanks for the info.

On Wed, Sep 2, 2009 at 3:52 PM, Brian West  wrote:

> well your clock shouldn't be going back in time... that is unless you
> have figured out time travel or passed thru some star trekish temporal
> wake.
>
> For the most part its a harmless warning unless its happening every
> second or so.
>
> /b
>
>
> On Sep 2, 2009, at 2:43 PM, Phillip Jones wrote:
>
> > Hi there,
> >
> > Can anyone give any insight to this following message:
> >
> > 2009-09-02 15:21:44.665608 [CRIT] switch_time.c:454 Reverse Clock
> > Skew Detected!
> >
> > This is on a WIN2003 machine with the last call hangup exactly 20
> > minutes and 20 seconds earlier.
> >
> > Just wondering how CRITICAL this really is?
> >
> > Many thanks
> >
> >
> > Phillip Jones
>
>
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[Freeswitch-users] Subscribing to events in managed C# / .NET

[Phillip Jones]

Hi there,

mod_managed exposes EventReceivedFunction such that:

 Session.EventReceivedFunction = (e) =>
 {
   Log.WriteLine(LogLevel.Alert, "Received Event {0}", e.ToString());
   return "";
 };

should trap all events to which i subscribe.


But how do I subscribe to events? What is the .NET / managed equivalent of:

switch_event_bind(const char *id, switch_event_types_t event, const char
*subclass_name, switch_event_callback_t callback, void *user_data);



Thank you!



Phillip Jones
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[Freeswitch-users] 2009-09-02 15:21:44.665608 [CRIT] switch_time.c:454 Reverse Clock Skew Detected!

[Phillip Jones]

Hi there,

Can anyone give any insight to this following message:

2009-09-02 15:21:44.665608 [CRIT] switch_time.c:454 Reverse Clock Skew
Detected!

This is on a WIN2003 machine with the last call hangup exactly 20 minutes
and 20 seconds earlier.

Just wondering how CRITICAL this really is?

Many thanks


Phillip Jones
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Re: [Freeswitch-users] How can I make more sounds that sound like "Callie"?

[Phillip Jones]

Taken from a similar recent posting:

"Callie" is one of the voices from GM Voices. She is definitely available
for custom work. Visit www.gmvoices.com for more info. Tell them that the
FreeSWITCH project sent you. :) MC

On Wed, Sep 2, 2009 at 3:14 PM, Nick Lemberger wrote:

> I'm guessing she's a Cepstral voice, but can I ask what version, khz and
> settings the 8khz sounds are recorded with (I ask about the khz because
> perhaps they were pre-transcoded)?  I tried downloading 5.1 from Cepstral
> but they don't sound at all alike.  I'd like to replace a few prompts with
> words more accurate for my installation but I'm having trouble getting close
> to the default voice.
>
> Best Regards,
> Nicholas Lemberger
> Lakefield Communications
> 920.973.6873
>
>
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Re: [Freeswitch-users] Can I stream a file to a parked call.

[Phillip Jones]

That makes perfect sense - I now send the call to a managed app instead of
park and every thing works great. thanks.



On Fri, Aug 28, 2009 at 5:12 PM, Anthony Minessale <
anthony.miness...@gmail.com> wrote:

> Taking over the session while it's parked like that from your code and
> asking it to play a file is making it do 2 things at once.
> The session's thread is already busy in the park loop so you would have to
> change it's state to something passive like soft_execute so
> it's thread was not doing anything or send it a message so the park loop
> would pick it up by using the uuid_broadcast app::arg
>
> there are variables you can set that begin group_confirm_* that you can set
> to do this all without even using a script.
>
>
> http://wiki.freeswitch.org/wiki/Channel_Variables#Answer_confirmation_variables
>
>
>
>
>
> On Fri, Aug 28, 2009 at 3:53 PM, Phillip Jones wrote:
>
>> The dialplan is is kicking off the C# script:
>>
>> 
>> * *
>> 
>>
>> I launch a thread and specify the uuid to originate:
>>
>> fsApi.Execute("originate",
>> string.Format("{{ignore_early_media=true,absolute_codec_string='PCMU'}}[*
>> origination_uuid={0}*,origination_caller_id_number={1}]sofia/gateway/broadvox/{2}
>> &park", *call.Uuid*, call.OutgoingCallerID, call.NumberToDial));
>>
>> in the callback i try and play the wav file:
>>
>> FreeSWITCH.Native.ManagedSession blegSession = new
>> FreeSWITCH.Native.ManagedSession(callID);
>>
>> if (blegSession.Ready() && blegSession.mediaReady())
>> {
>>   blegSession.Answer();
>>   blegSession.Execute("playback",
>> "prompts/press-1-to-accept-call-from.wav");
>> }
>>
>> I go on to bridge the leg a and b - that bits works fine.
>>
>>
>>
>> On Fri, Aug 28, 2009 at 4:40 PM, Anthony Minessale<
>> anthony.miness...@gmail.com> wrote:
>> > nevermind,
>> >
>> > I didn't look closely at your code.
>> >
>> > how did you end up with blegSession ?
>> > Are you running this script from the CLI or via the application
>> interface.
>> >
>> >
>> >
>> >
>> >
>> >
>> >
>> > On Fri, Aug 28, 2009 at 3:33 PM, Anthony Minessale
>> >  wrote:
>> >>
>> >> the L16 codec is used because the wav files is raw PCM audio
>> >> which is being encoded to PCMU
>> >>
>> >> Have you tried doing this to a locally registered phone?
>> >>
>> >>
>> >>
>> >> On Fri, Aug 28, 2009 at 3:13 PM, Phillip Jones 
>> >> wrote:
>> >>>
>> >>> thanks for the reply.
>> >>>
>> >>> Tried this and exactly the same result. The prompt is slowed - as if
>> >>> the person is drunk.
>> >>>
>> >>> I do notice that the originating call SDP shows:
>> >>>
>> >>> a=rtpmap:0 PCMU/8000
>> >>>
>> >>> but the play tries to use:
>> >>>
>> >>> EXECUTE sofia/external/6093693828
>> >>> playback(prompts/press-1-to-accept-call-from.wav)
>> >>> 2009-08-28 16:03:14.507375 [DEBUG] switch_ivr_play_say.c:1097 Codec
>> >>> Activated l...@8000hz 1 channels 20ms
>> >>>
>> >>>
>> >>> Should the codec be p...@8000hz as opposed to l...@8000hz? If so of
>> >>> there a way of forcing this codec for play.
>> >>>
>> >>> Thanks
>> >>>
>> >>>
>> >>> Phillip
>> >>>
>> >>>
>> >>> Full code & logs
>> >>> http://pastebin.freeswitch.org/10147
>> >>> http://pastebin.freeswitch.org/10150
>> >>>
>> >>>
>> >>>
>> >>> On Thu, Aug 27, 2009 at 4:30 PM, Anthony
>> >>> Minessale wrote:
>> >>> > try wedging {ignore_early_media=true} before the first [ in your
>> dial
>> >>> > string
>> >>> > and eliminate the code waiting for answer.
>> >>> >
>> >>> >
>> >>> > On Thu, Aug 27, 2009 at 3:09 PM, Phillip Jones <
>> pjinthe...@gmail.com>
>> >>> > wrote:
>> >>> >>
>> >>> >> Hi there,
>> >>> >>
>> >>> >> I know there are other ways of doin

Re: [Freeswitch-users] Can I stream a file to a parked call.

[Phillip Jones]

The dialplan is is kicking off the C# script:


* *


I launch a thread and specify the uuid to originate:

fsApi.Execute("originate",
string.Format("{{ignore_early_media=true,absolute_codec_string='PCMU'}}[*
origination_uuid={0}*,origination_caller_id_number={1}]sofia/gateway/broadvox/{2}
&park", *call.Uuid*, call.OutgoingCallerID, call.NumberToDial));

in the callback i try and play the wav file:

FreeSWITCH.Native.ManagedSession blegSession = new
FreeSWITCH.Native.ManagedSession(callID);

if (blegSession.Ready() && blegSession.mediaReady())
{
  blegSession.Answer();
  blegSession.Execute("playback",
"prompts/press-1-to-accept-call-from.wav");
}

I go on to bridge the leg a and b - that bits works fine.



On Fri, Aug 28, 2009 at 4:40 PM, Anthony Minessale<
anthony.miness...@gmail.com> wrote:
> nevermind,
>
> I didn't look closely at your code.
>
> how did you end up with blegSession ?
> Are you running this script from the CLI or via the application interface.
>
>
>
>
>
>
>
> On Fri, Aug 28, 2009 at 3:33 PM, Anthony Minessale
>  wrote:
>>
>> the L16 codec is used because the wav files is raw PCM audio
>> which is being encoded to PCMU
>>
>> Have you tried doing this to a locally registered phone?
>>
>>
>>
>> On Fri, Aug 28, 2009 at 3:13 PM, Phillip Jones 
>> wrote:
>>>
>>> thanks for the reply.
>>>
>>> Tried this and exactly the same result. The prompt is slowed - as if
>>> the person is drunk.
>>>
>>> I do notice that the originating call SDP shows:
>>>
>>> a=rtpmap:0 PCMU/8000
>>>
>>> but the play tries to use:
>>>
>>> EXECUTE sofia/external/6093693828
>>> playback(prompts/press-1-to-accept-call-from.wav)
>>> 2009-08-28 16:03:14.507375 [DEBUG] switch_ivr_play_say.c:1097 Codec
>>> Activated l...@8000hz 1 channels 20ms
>>>
>>>
>>> Should the codec be p...@8000hz as opposed to l...@8000hz? If so of
>>> there a way of forcing this codec for play.
>>>
>>> Thanks
>>>
>>>
>>> Phillip
>>>
>>>
>>> Full code & logs
>>> http://pastebin.freeswitch.org/10147
>>> http://pastebin.freeswitch.org/10150
>>>
>>>
>>>
>>> On Thu, Aug 27, 2009 at 4:30 PM, Anthony
>>> Minessale wrote:
>>> > try wedging {ignore_early_media=true} before the first [ in your dial
>>> > string
>>> > and eliminate the code waiting for answer.
>>> >
>>> >
>>> > On Thu, Aug 27, 2009 at 3:09 PM, Phillip Jones 
>>> > wrote:
>>> >>
>>> >> Hi there,
>>> >>
>>> >> I know there are other ways of doing this. I am just trying get to
>>> >> know have fun with the FreeSWITCH API.
>>> >>
>>> >> I am using originate and park a call:
>>> >>
>>> >> fsApi.Execute("originate",
>>> >>
>>> >>
>>> >>
string.Format("[origination_uuid={0},origination_caller_id_number={1}]sofia/gateway/broadvox/{2}
>>> >> &park", blegSession.Uuid, OutgoingCallerID, NumberToDial));
>>> >>
>>> >> That works great, the phone rings. I want to play something to this
>>> >> called party when they pick up.
>>> >>
>>> >>  while (!blegSession.answered())
>>> >>  {
>>> >>Log.WriteLine(LogLevel.Alert,
>>> >> "Inside::CallReturns:!Session.answered::Loop");
>>> >>blegSession.sleep(500, 1);
>>> >>  }
>>> >>
>>> >> string promptFile = "prompts/whisper.wav";
>>> >> blegSession.StreamFile(promptFile, 0);
>>> >>
>>> >>
>>> >> This works - but the audio is choppy and slow.
>>> >>
>>> >> Is there something I need to do to that parked call before streaming
>>> >> that
>>> >> file?
>>> >>
>>> >>
>>> >> Thanks for any input.
>>> >>
>>> >> Phil
>>> >>
>>> >> ___
>>> >> FreeSWITCH-users mailing list
>>> >> FreeSWITCH-users@lists.freeswitch.org
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>
>>> >> UNSUBSCRIBE:
http://lists.frees

Re: [Freeswitch-users] Can I stream a file to a parked call.

[Phillip Jones]

thanks for the reply.

Tried this and exactly the same result. The prompt is slowed - as if
the person is drunk.

I do notice that the originating call SDP shows:

a=rtpmap:0 PCMU/8000

but the play tries to use:

EXECUTE sofia/external/6093693828
playback(prompts/press-1-to-accept-call-from.wav)
2009-08-28 16:03:14.507375 [DEBUG] switch_ivr_play_say.c:1097 Codec
Activated l...@8000hz 1 channels 20ms


Should the codec be p...@8000hz as opposed to l...@8000hz? If so of
there a way of forcing this codec for play.

Thanks


Phillip


Full code & logs
http://pastebin.freeswitch.org/10147
http://pastebin.freeswitch.org/10150



On Thu, Aug 27, 2009 at 4:30 PM, Anthony
Minessale wrote:
> try wedging {ignore_early_media=true} before the first [ in your dial string
> and eliminate the code waiting for answer.
>
>
> On Thu, Aug 27, 2009 at 3:09 PM, Phillip Jones  wrote:
>>
>> Hi there,
>>
>> I know there are other ways of doing this. I am just trying get to
>> know have fun with the FreeSWITCH API.
>>
>> I am using originate and park a call:
>>
>> fsApi.Execute("originate",
>>
>> string.Format("[origination_uuid={0},origination_caller_id_number={1}]sofia/gateway/broadvox/{2}
>> &park", blegSession.Uuid, OutgoingCallerID, NumberToDial));
>>
>> That works great, the phone rings. I want to play something to this
>> called party when they pick up.
>>
>>  while (!blegSession.answered())
>>  {
>>        Log.WriteLine(LogLevel.Alert,
>> "Inside::CallReturns:!Session.answered::Loop");
>>        blegSession.sleep(500, 1);
>>  }
>>
>> string promptFile = "prompts/whisper.wav";
>> blegSession.StreamFile(promptFile, 0);
>>
>>
>> This works - but the audio is choppy and slow.
>>
>> Is there something I need to do to that parked call before streaming that
>> file?
>>
>>
>> Thanks for any input.
>>
>> Phil
>>
>> ___
>> FreeSWITCH-users mailing list
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>
>
>
> --
> Anthony Minessale II
>
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> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
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> 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
>
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[Freeswitch-users] Can I stream a file to a parked call.

[Phillip Jones]

Hi there,

I know there are other ways of doing this. I am just trying get to
know have fun with the FreeSWITCH API.

I am using originate and park a call:

fsApi.Execute("originate",
string.Format("[origination_uuid={0},origination_caller_id_number={1}]sofia/gateway/broadvox/{2}
&park", blegSession.Uuid, OutgoingCallerID, NumberToDial));

That works great, the phone rings. I want to play something to this
called party when they pick up.

 while (!blegSession.answered())
 {
Log.WriteLine(LogLevel.Alert,
"Inside::CallReturns:!Session.answered::Loop");
blegSession.sleep(500, 1);
 }

string promptFile = "prompts/whisper.wav";
blegSession.StreamFile(promptFile, 0);


This works - but the audio is choppy and slow.

Is there something I need to do to that parked call before streaming that file?


Thanks for any input.

Phil

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Re: [Freeswitch-users] how to avoid many "|" in bridge application?

[Phillip Jones]

Take a look at http://jira.freeswitch.org/browse/FSCORE-422. This a
feature request I submitted. This problem it solves is different - but
the solution is the same. Perhaps you add your take to the comments
there.

On Tue, Aug 25, 2009 at 10:06 AM, Carlos S. Antunes wrote:
> Max,
>
> I would like to see something similar too. For example, it would be
> wonderful if one could specify multiple gateways to try like this or
> something similar:
>
> 
>    
>    
>    
>    
>        
>    
>    
> 
>
> One would be able to avoid the "[]" and "{}" hacks and combine
> sequential and simultaneous trying of gateways.
>
> What do the developers think of this?
>
> Carlos
>
> Max Ivanov wrote:
>> Nowdays I 'm forced to put multiple "|" to find first free gateway, ie
>> sofia/gateway/panas111/1000|sofia/gateway/panas112/1000|sofia/gateway/panas113/1000
>> ,
>> the whole sting is tooo long, is there any shorter way to write same thing? 
>> Like
>> "sofia/gateway/panas*/1000" will try all gateways matching the pattern.
>>
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>
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Re: [Freeswitch-users] Loopback and bypass_media

[Phillip Jones]

Hi there,

I created a feature request to cover this issue:

http://jira.freeswitch.org/browse/FSCORE-422 - The ability to support
Call FollowMe (or Call Blast) and multiple termination carriers -
without Loopback

If anybody wants comment on its merits and/or make the request clearer
- that would be great.

Thanks for every bodies help on this.

Phillip Jones

On Fri, Aug 14, 2009 at 3:17 PM, Phillip Jones wrote:
> Hi Rupa,
>
> What about my suggestion above introduce a "api_after_bridge" event
> that fires when the switch_ivr_uuid_bridge() bridges to the two sofia
> channels that Mathieu mentioned?
>
> Is that suggestion just way off the mark? If possible that would allow
> me to move forward - although I agree that supporting groupings of
> carriers is would be the most elegant solution.
>
> Let me know if I am just talking rubbish re the api_after_bridge" event.
>
> Thanks!
>
>
> Phillip Jones
>
>
> On Fri, Aug 14, 2009 at 2:33 AM, Rupa Schomaker wrote:
>> On Thu, Aug 13, 2009 at 6:54 PM, Mathieu Rene wrote:
>>> Hi All,
>>>
>>> The reason it works when you wait 3 seconds is that mod_loopback bails
>>
>> [snip]
>>
>> Thanks for that explanation.  It umm.. explains a lot. :)
>>
>>> On another note, mod_sofia will behave differently when it detects its
>>> being bridge with another sofia channel, providing optimizations when
>>> both call legs are SIP.
>>>
>>> My personal opinion is not to use mod_loopback unless absolutely
>>> necessary, FreeSWITCH's core is very flexible and there's often a
>>> (better) way than using mod_loopback.
>>
>> So, I think the temp solution is to use loopback+delayed no media.
>>
>> but the real "solution" is to either drive the forked dialing logic
>> externally (event socket) or consider supporting groupings in the
>> bridge which.. umm...  is gonna be a pain and will need buy in from
>> from Tony and other core devs since that is a core (no pun intended)
>> piece of code that nearly everything uses.
>>
>> I'm not sure I want to take a wack at it.
>>
>> --
>> -Rupa
>>
>> ___
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>

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Re: [Freeswitch-users] Loopback and bypass_media

[Phillip Jones]

Hi Rupa,

What about my suggestion above introduce a "api_after_bridge" event
that fires when the switch_ivr_uuid_bridge() bridges to the two sofia
channels that Mathieu mentioned?

Is that suggestion just way off the mark? If possible that would allow
me to move forward - although I agree that supporting groupings of
carriers is would be the most elegant solution.

Let me know if I am just talking rubbish re the api_after_bridge" event.

Thanks!


Phillip Jones


On Fri, Aug 14, 2009 at 2:33 AM, Rupa Schomaker wrote:
> On Thu, Aug 13, 2009 at 6:54 PM, Mathieu Rene wrote:
>> Hi All,
>>
>> The reason it works when you wait 3 seconds is that mod_loopback bails
>
> [snip]
>
> Thanks for that explanation.  It umm.. explains a lot. :)
>
>> On another note, mod_sofia will behave differently when it detects its
>> being bridge with another sofia channel, providing optimizations when
>> both call legs are SIP.
>>
>> My personal opinion is not to use mod_loopback unless absolutely
>> necessary, FreeSWITCH's core is very flexible and there's often a
>> (better) way than using mod_loopback.
>
> So, I think the temp solution is to use loopback+delayed no media.
>
> but the real "solution" is to either drive the forked dialing logic
> externally (event socket) or consider supporting groupings in the
> bridge which.. umm...  is gonna be a pain and will need buy in from
> from Tony and other core devs since that is a core (no pun intended)
> piece of code that nearly everything uses.
>
> I'm not sure I want to take a wack at it.
>
> --
> -Rupa
>
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Re: [Freeswitch-users] Loopback and bypass_media

[Phillip Jones]

> The reason it works when you wait 3 seconds is that mod_loopback bails
> out of the equation as soon as it detects a bridge.
> It'll call switch_ivr_uuid_bridge() and you'll end up with 2 sofia
> channels.

Is there a hook that is fired when when that switch_ivr_uuid_bridge()
successfully executes? So the uuid_media off is called on the
appropriate sofia channels? Is "api_after_bridge" behaving correctly -
should that only be called on the sofia channels and not the loopback?
Is it being fired to early?



On Thu, Aug 13, 2009 at 4:54 PM, Mathieu Rene wrote:
> Hi All,
>
> The reason it works when you wait 3 seconds is that mod_loopback bails
> out of the equation as soon as it detects a bridge.
> It'll call switch_ivr_uuid_bridge() and you'll end up with 2 sofia
> channels.
>
> Now the reason why you can't do uuid_media over a loopback channel is
> because it doesn't pass on SWITCH_MESSAGE_INDICATE_MEDIA and
> SWITCH_MESSAGE_INDICATE_NOMEDIA onto the underlying channel.
> The handler for those two events require accessing channel variables
> on the both channels to get the ip+port of where the audio should go
> through, so that mod_sofia can send a re-invite.
> Since mod_loopback is a completely different channel, it has its own
> channel variables, independent from mod_sofia (provided you have sofia
> channels on both side).  That's why even sofia<>loopback won't do
> bypass media.
>
> On another note, mod_sofia will behave differently when it detects its
> being bridge with another sofia channel, providing optimizations when
> both call legs are SIP.
>
> My personal opinion is not to use mod_loopback unless absolutely
> necessary, FreeSWITCH's core is very flexible and there's often a
> (better) way than using mod_loopback.
>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mr...@avgs.ca
>
>
>
>
> On 13-Aug-09, at 6:59 PM, Phillip Jones wrote:
>
>> Rupa / all,
>>
>> Just a quick follow up to this.
>>
>> This is appears to a timing issue. If I try and do a "uuid_media off +
>> uuid"  in  "api_after_bridge"  it fails with "CHAN_NOT_IMPLEMENTED"
>> and  the call is dropped.
>>
>> If appears to be trying to do a SIP reinvite on the loopback channel
>> which is of course just about to / has disappear/ed.
>>
>> So I tried this, after the call is established, at the commend line, I
>> do "show calls" and using the uuid shown, type "uuid_media off  uuid".
>> The SIP REINVITE is issued and works.
>>
>> I think the switch_ivr_nomedia function in switch_ivr_c is getting the
>> loopback uuid when it calls "other_uuid =
>> switch_channel_get_variable(channel, SWITCH_BRIDGE_VARIABLE)"
>>
>> That's why the SIP REINVITE fails.
>>
>> So... in api_after_bridge I issue:
>>
>> "sched_api", "+3 none uuid_media off " + uuid. This calls the
>> switch_ivr_nomedia function 3 seconds after the calls bridge is
>> established.
>>
>>
>> And it works, Not nice - not scalable or production ready - but the
>> SIP-REINVITE is successful and at least now I understand what is going
>> on.
>>
>> Make sense?
>>
>> Thanks
>>
>>
>> Phil
>>
>>
>> On Wed, Aug 12, 2009 at 12:21 PM, Rupa Schomaker wrote:
>>> On Wed, Aug 12, 2009 at 10:22 AM, Phillip
>>> Jones wrote:
>>>> Hi there,
>>>>
>>>>>> application="originate" data="(sofia/foo/bar|sofia/baz/bar),
>>>>>> (sofia/foo/yum|sofia/baz/yum)"
>>>>
>>>> I agree. However, perhaps the ideal is not to specify the carriers
>>>> at
>>>> this level, as carriers are added and removed fairly often as
>>>> costings
>>>> change. So it would be nice to have some sort of proxy that resolves
>>>> to a list of carriers:
>>>>
>>>> application="originate" data="sofia/MyCarriers/bar,sofia/
>>>> MyCarriers/yum"
>>>
>>>> 
>>>> 
>>>> 
>>>> 
>>>> 
>>>>
>>>>
>>>> or something similar. This would achieve the same as loopback in
>>>> this
>>>> use case but without dangers of looping? Complicated stuff though.
>>>
>>> Well, that is all done by mod_lcr.  I was simplifying to narrow down
>>> to just originate.
>>>
>>> First we need to see if this 

Re: [Freeswitch-users] Loopback and bypass_media

[Phillip Jones]

Rupa / all,

Just a quick follow up to this.

This is appears to a timing issue. If I try and do a "uuid_media off +
uuid"  in  "api_after_bridge"  it fails with "CHAN_NOT_IMPLEMENTED"
and  the call is dropped.

If appears to be trying to do a SIP reinvite on the loopback channel
which is of course just about to / has disappear/ed.

So I tried this, after the call is established, at the commend line, I
do "show calls" and using the uuid shown, type "uuid_media off  uuid".
The SIP REINVITE is issued and works.

I think the switch_ivr_nomedia function in switch_ivr_c is getting the
loopback uuid when it calls "other_uuid =
switch_channel_get_variable(channel, SWITCH_BRIDGE_VARIABLE)"

That's why the SIP REINVITE fails.

So... in api_after_bridge I issue:

"sched_api", "+3 none uuid_media off " + uuid. This calls the
switch_ivr_nomedia function 3 seconds after the calls bridge is
established.


And it works, Not nice - not scalable or production ready - but the
SIP-REINVITE is successful and at least now I understand what is going
on.

Make sense?

Thanks


Phil


On Wed, Aug 12, 2009 at 12:21 PM, Rupa Schomaker wrote:
> On Wed, Aug 12, 2009 at 10:22 AM, Phillip Jones wrote:
>> Hi there,
>>
>>>> application="originate" 
>>>> data="(sofia/foo/bar|sofia/baz/bar),(sofia/foo/yum|sofia/baz/yum)"
>>
>> I agree. However, perhaps the ideal is not to specify the carriers at
>> this level, as carriers are added and removed fairly often as costings
>> change. So it would be nice to have some sort of proxy that resolves
>> to a list of carriers:
>>
>> application="originate" data="sofia/MyCarriers/bar,sofia/MyCarriers/yum"
>
>> 
>> 
>> 
>> 
>> 
>>
>>
>> or something similar. This would achieve the same as loopback in this
>> use case but without dangers of looping? Complicated stuff though.
>
> Well, that is all done by mod_lcr.  I was simplifying to narrow down
> to just originate.
>
> First we need to see if this is worth pursuing over fixing (modifying,
> whatever) loopback to handle bypass media.  If it is, then I'll modify
> mod_lcr to deal with the situation in question (comma or pipe sep list
> of numbers to call.  mod_lcr would then group as appropriate).
>
> Right now, my bridge is setup in a small javascript script that builds
> the appropriate dialstring (using loopback for external calls, user/
> for internal calls) and then when doing the loopback call to mod_lcr
> to get the dialstring with all providers in the right order.
>
>>>>Perhaps have an on answer hook that tries to enable bypass media 
>>>>(re-invite) after the call is setup?
>>
>> That's a good idea - I will look into that.
>>
>>
>> Thanks again.
>>
>>
>> Phillip
>
> Let us know how it works for you...
>
> --
> -Rupa
>
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Re: [Freeswitch-users] mod_managed users?

[Phillip Jones]

Hey Michael,

I am a little late to the party I know - but just want to say thanks
for your latest efforts.

I updated my dev environment with the latest managed mod and swapped
my app to the latest plugin architecture last night and all is working
well.

Love the dynamic loading of my dll into freeswitch - no more starting
and stopping freeswitch!

Also appreciate the f# example.

Thanks again.


Phillip Jones

On Wed, Jul 29, 2009 at 8:04 PM, Michael Giagnocavo wrote:
> Which directory ends with ; ?
>
>
>
> I’m not following – if you want email me off list and we can work together
> on it .
>
>
>
> From: freeswitch-users-boun...@lists.freeswitch.org
> [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Diego
> Toro
> Sent: Wednesday, July 29, 2009 5:06 PM
> To: freeswitch-users@lists.freeswitch.org
> Subject: Re: [Freeswitch-users] mod_managed users?
>
>
>
> Hi,
>
>
>
> My dll's are loaded correctly, I have a trouble becouse the directory where
> are loaded it finish with ";" (whitout ""), i mean the default path assembly
> finish with ";"
>
>
>
> Is possible remove the character ";"  ?
>
>
>
> thanks
>
>
>
> Diego
>
> --- On Wed, 7/29/09, Michael Giagnocavo  wrote:
>
> From: Michael Giagnocavo 
> Subject: Re: [Freeswitch-users] mod_managed users?
> To: "freeswitch-users@lists.freeswitch.org"
> 
> Date: Wednesday, July 29, 2009, 2:32 PM
>
> The error is that it just doesn’t find that alias to call the plugin.
> Assuming everything is spelled correctly, this probably means the DLL did
> not load.
>
>
>
> I just checked in a fix for dlls that don’t have both Api and App interfaces
> – it would not load them at all. Try with it now and see if that’s the
> problem. If it is, I apologize.
>
>
>
> If it still doesn’t, paste the full log of when it loads your file.
>
>
>
> Thanks,
>
> Michael
>
>
>
> From: freeswitch-users-boun...@lists.freeswitch.org
> [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Diego
> Toro
> Sent: Wednesday, July 29, 2009 12:44 PM
> To: freeswitch-users@lists.freeswitch.org
> Subject: Re: [Freeswitch-users] mod_managed users?
>
>
>
> Hi Michael,
>
>
>
> I am working with lastest managed module, I have assemblies wich dependen of
> others assemblies (dll's), the past version it works fine, but I have now
> next the error message:
>
>
>
> EXECUTE sofia/internal/10...@192.168.27.10 managed(CIV_BPFSProcess)
> 2009-07-29 13:31:27.718750 [DEBUG] switch_cpp.cpp:1130 FreeSWITCH.Managed:
> attempting to run application 'CIV_BPFSProcess'.
> 2009-07-29 13:31:27.718750 [ERR] switch_cpp.cpp:1130 App plugin
> CIV_BPFSProcess not found.
> 2009-07-29 13:31:27.718750 [ERR] mod_managed.cpp:405 Application run failed
> for CIV_BPFSProcess (unknown module or exception).
>
>
>
> My declaration class  is:
> public class CIV_BPFSProcess : FreeSWITCH.IAppPlugin
>
> It has public method:
> public void Run(AppContext context).
>
>
>
> I have VS2008 on Windows
>
> Thanks
>
>
>
> Diego
>
> --- On Wed, 7/29/09, Michael Giagnocavo  wrote:
>
> From: Michael Giagnocavo 
> Subject: Re: [Freeswitch-users] mod_managed users?
> To: "freeswitch-users@lists.freeswitch.org"
> 
> Date: Wednesday, July 29, 2009, 1:46 AM
>
> Hi Łukasz,
>
>         Would you please send me the DLL offlist and I'll figure it out?
>
>         The new session you create is the b-leg. The parameter it takes in
> originate is the a-leg. So you'd do:
>
> var session = new ManagedSession();
> session.Originate(context.Session, "sofia/default/1000",10);
>
>         As to non-blocking, I'm quite sure it's possible, but I don't recall
> offhand which functions. This should be the same as in any other language
> for FreeSWITCH -- these functions are just passthrough from the FS C++ API.
>
> -Michael
>
> -Original Message-
> From: freeswitch-users-boun...@lists.freeswitch.org
> [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Lukasz
> Zwierko
> Sent: Wednesday, July 29, 2009 12:13 AM
> To: freeswitch-users@lists.freeswitch.org
> Subject: Re: [Freeswitch-users] mod_managed users?
>
> Hi Michael ,
>
> thanks a lot for support on this.
>
>> As to the main problem of your DLL not working, can you send me the full
>> source code, or all the logging output from loading it? Try "managedreload
>> my.dll" to reload the DLL and see how it is registering them. It should
>> output something like "Registering API Full

Re: [Freeswitch-users] Loopback and bypass_media

[Phillip Jones]

Hi there,

>> application="originate" 
>> data="(sofia/foo/bar|sofia/baz/bar),(sofia/foo/yum|sofia/baz/yum)"

I agree. However, perhaps the ideal is not to specify the carriers at
this level, as carriers are added and removed fairly often as costings
change. So it would be nice to have some sort of proxy that resolves
to a list of carriers:

application="originate" data="sofia/MyCarriers/bar,sofia/MyCarriers/yum"








or something similar. This would achieve the same as loopback in this
use case but without dangers of looping? Complicated stuff though.

>>Perhaps have an on answer hook that tries to enable bypass media (re-invite) 
>>after the call is setup?

That's a good idea - I will look into that.


Thanks again.


Phillip

On Wed, Aug 12, 2009 at 10:22 AM, Rupa Schomaker wrote:
> perhaps we need to add some syntax + logic to originate:
>
> application="originate"
> data="(sofia/foo/bar|sofia/baz/bar),(sofia/foo/yum|sofia/baz/yum)"
>
> This would acomplish the equiv of
>
> loopback/bar,loopback/yum  where bar and yum are then further expanded
> in the dialplan as
>
> sofia/foo/bar|sofia/baz/bar and sofia/foo/yum|sofia/baz/yum
>
> except that the threads of execution are handled directly by
> originate.  I'm not sure that is really the "solution" since each ()
> group would still have to be a separate thread to run independently.
>
> To me, loopback is the way to accomplish this issue (how I've done it
> with the same requirements that you have) since all the hard work is
> layered and works.  The problem is that you require bypass_media which
> doesn't play nice with loopback.
>
> Perhaps have an on answer hook that tries to enable bypass media
> (re-invite) after the call is setup?
>
> On Wed, Aug 12, 2009 at 8:29 AM, Phillip Jones wrote:
>> David / Michael - thanks for your your replies. The SoftIVR example is
>> particularly useful. Must admit though - I was hoping not to have to
>> do any custom stuff at this stage.
>>
>> It does appear there is no method to do this by staking bridge lines
>> so I will put an issue in jira to try and get loopback working with
>> bypass_media.
>>
>> In the meantime I will also start looking to build a custom bridging
>> app. As I said though - not a road I wanted to go down.
>>
>> Thanks for your help!
>>
>>
>> Phillip Jones
>>
>> On Tue, Aug 11, 2009 at 6:13 PM, Michael Giagnocavo 
>> wrote:
>>> It's also simple enough to write a plugin in one of the scripting languages 
>>> to add an app to do exactly what you want...
>>>
>>> -Original Message-
>>> From: freeswitch-users-boun...@lists.freeswitch.org 
>>> [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of David 
>>> Knell
>>> Sent: Tuesday, August 11, 2009 1:55 PM
>>> To: freeswitch-users@lists.freeswitch.org
>>> Subject: Re: [Freeswitch-users] Loopback and bypass_media
>>>
>>> Just to add my $0.02-worth (if you're feeling generous..) - I don't
>>> think that the dialplan is expressive enough to do what's needed here,
>>> and that's where the trouble's coming from.  It's not enormously tricky
>>> to build a generic "dial this set of numbers according to these rules"
>>> service using something hanging off the event socket - there's a writeup
>>> here: http://www.softivr.com/wiki/index.php/Find_me showing how it could
>>> be done on SoftIVR.
>>>
>>> To roll something similar yourself using the event socket, you'd need to
>>> map the dial function to 'originate', bridge to (IIRC) 'uuid_bridge',
>>> and have some way of passing messages around between the threads
>>> handling the different call legs, assuming that you're using one thread
>>> per leg.
>>>
>>> --Dave
>>>
>>>
>>> ___
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>>> http://www.fr

Re: [Freeswitch-users] Loopback and bypass_media

[Phillip Jones]

David / Michael - thanks for your your replies. The SoftIVR example is
particularly useful. Must admit though - I was hoping not to have to
do any custom stuff at this stage.

It does appear there is no method to do this by staking bridge lines
so I will put an issue in jira to try and get loopback working with
bypass_media.

In the meantime I will also start looking to build a custom bridging
app. As I said though - not a road I wanted to go down.

Thanks for your help!


Phillip Jones

On Tue, Aug 11, 2009 at 6:13 PM, Michael Giagnocavo wrote:
> It's also simple enough to write a plugin in one of the scripting languages 
> to add an app to do exactly what you want...
>
> -Original Message-
> From: freeswitch-users-boun...@lists.freeswitch.org 
> [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of David 
> Knell
> Sent: Tuesday, August 11, 2009 1:55 PM
> To: freeswitch-users@lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Loopback and bypass_media
>
> Just to add my $0.02-worth (if you're feeling generous..) - I don't
> think that the dialplan is expressive enough to do what's needed here,
> and that's where the trouble's coming from.  It's not enormously tricky
> to build a generic "dial this set of numbers according to these rules"
> service using something hanging off the event socket - there's a writeup
> here: http://www.softivr.com/wiki/index.php/Find_me showing how it could
> be done on SoftIVR.
>
> To roll something similar yourself using the event socket, you'd need to
> map the dial function to 'originate', bridge to (IIRC) 'uuid_bridge',
> and have some way of passing messages around between the threads
> handling the different call legs, assuming that you're using one thread
> per leg.
>
> --Dave
>
>
> ___
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Re: [Freeswitch-users] Loopback and bypass_media

[Phillip Jones]

Thanks Michael,

pseudo code would look this.

1) Call 609555 and 732555 simultaneously.
2a) As you dial 609555 go to the next carrier if you receive
NO_ROUTE_DESTINATION.
2b) As you dial 732555 go to the next carrier if you receive
NO_ROUTE_DESTINATION
3) The first number to answer and accept the call - bridge to leg a.
Drop the other.


Brian maintains that this can be done without loopback and loopback
should never be used. Mike Jerris believes that loop back handles this
and should be used. I am happy for all the help, but I am a bit
confused at this stage.

If this can not be done in the dailplan then I will need to pursue
loopback and see how far I get with that. At the moment it works but
not with bypass_media.

As I said "call followme" and carrier failover are fairly common
requirements so I do feel I am missing something. This must be
implemented elsewhere.

Again - thank you all for your help. You do a great job of getting
getting your heads around all this stuff day in day out.

Phillip Jones


On Tue, Aug 11, 2009 at 3:19 PM, Michael Collins wrote:
>
>
> On Tue, Aug 11, 2009 at 1:43 PM, Phillip Jones  wrote:
>>
>> Hi Michael,
>>
>> Thanks for trying to help with this.
>>
>> > 
>> > > >
>> > data="sofia/gateway/broadvox/6095553828,sofia/gateway/broadvox/7325553828"/>
>> > 
>> > 
>>
>> In the dial plan above, consider that target is sitting at their
>> 6095553828 phone. And that broadvox does not route to 6095553828  and
>> passes me a NO_ROUTE_DESTINATION.
>>
>> The target will never get the call.
>>
>> So if I extend this to:
>>
>> 
>> 
>>
>> In this plan you are not calling the numbers simultaneously if the
>> broadvox is not able to route both calls. I can not see that you can
>> stack this way and get the ability to call multiple destinations, so I
>> must be missing something.
>
> When you say "call multiple destinations" what exactly do you mean? If the
> Broadvox is unable to connect either call then the dialplan moves on to the
> Quest dial attempts. If that isn't what you need then can you clarify? Under
> what condition(s) would the dialplan need to attempt to call via Quest?
>
>>
>>
>> "Call blast" or "Followme" is basically the functionality I am trying
>> to implement. Separately we use multiple carriers. This is fairly
>> common requirement.
>
> Yep, makes sense. Could you maybe write it out in pseudo code? Something
> like this:
> #1 Call phone numbers ABC and XYZ simultaneously via Broadvox
> #2 If BOTH ABC AND XYZ fail, then call ABC and XYZ via Quest
>
>>
>> I am using .NET by the way - but point taken.
>
> My mistake. :)
>
>>
>> Any other ideas on how to achieve multiple destinations with carrier
>> failover would great!
>
> I'm 100% certain that this is possible with FreeSWITCH. You might just need
> a different approach, depending upon the circumstances with your carriers.
> Try the above suggestions first and let's see what happens. We'll take it
> from there.
> -MC
>
>
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Re: [Freeswitch-users] hangup_after_bridge=false only works if bypass_media=false

[Phillip Jones]

Hi there,

I the 'does not work' example your media stream is back up at the
carrier as the called party hangs up.

You would have to cause another SIP REINVITE and re capture the media.

See http://wiki.freeswitch.org/wiki/Bypass_Media and the "How to
disable/enable it on the fly?" section as a start.




On Tue, Aug 11, 2009 at 2:29 PM, Jeremiah Johnson wrote:
> This is an integral part of my application.  I need to have FreeSWITCH
> outside of the media path as well as be able to do multiple bridges for the
> same "A" leg.
> /*WORKS*/
> 
>  data="sofia/gateway/${mygateway}/1${client_one}"/>
>  data="sofia/gateway/${mygateway}/1${client_two}"/>
> /*DOES NOT WORK*/
> 
> 
>  data="sofia/gateway/${mygateway}/1${client_one}"/>
>  data="sofia/gateway/${mygateway}/1${client_two}"/>
> In the "DOES NOT WORK" example, the "A" leg hangs up as soon as the leg for
> client_one hangs up.
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Re: [Freeswitch-users] Loopback and bypass_media

[Phillip Jones]

Hi Michael,

Thanks for trying to help with this.

> 
>  data="sofia/gateway/broadvox/6095553828,sofia/gateway/broadvox/7325553828"/>
> 
> 

In the dial plan above, consider that target is sitting at their
6095553828 phone. And that broadvox does not route to 6095553828  and
passes me a NO_ROUTE_DESTINATION.

The target will never get the call.

So if I extend this to:




In this plan you are not calling the numbers simultaneously if the
broadvox is not able to route both calls. I can not see that you can
stack this way and get the ability to call multiple destinations, so I
must be missing something.

"Call blast" or "Followme" is basically the functionality I am trying
to implement. Separately we use multiple carriers. This is fairly
common requirement.

I am using .NET by the way - but point taken.

Any other ideas on how to achieve multiple destinations with carrier
failover would great!

Thanks again.


On Tue, Aug 11, 2009 at 2:07 PM, Michael Collins wrote:
>
>
> On Tue, Aug 11, 2009 at 11:08 AM, Phillip Jones 
> wrote:
>>
>> Can you use this method to make simultaneous calls though while
>> preserving carrier order? i.e. each call must try carrier 1 first then
>> carrier 2 (because each carrier terminates a different subset of
>> numbers)
>>
>> So - if I did NOT want to support failover I would use:
>>
>> session:execute("bridge",
>> "sofia/gateway/broadvox/6095553828,sofia/gateway/broadvox/7325553828");
>>  << call 6095553828 and 7325553828 at the same time
>>
>> If I were to use failover and not need multiple destinations then I would
>> use:
>>
>> session:execute("bridge",
>> "sofia/gateway/broadvox/6095553828|sofia/gateway/quest/6095553828");
>> << call using broadvox first and quest it that fails
>>
>> I need to combine:
>>
>> Call 6095553828  and 7325553828 simultaneously and each trying
>> broadvox then on fail quest.
>>
>> session:execute("bridge",
>>
>> "(sofia/gateway/broadvox/6095553828|sofia/gateway/quest/6095553828),(sofia/gateway/broadvox/7325553828,sofia/gateway/quest/7325553828)");
>>
>> But of course that syntax does not work.
>>
>> I can not see another syntax that would achieve this?
>
> Just to make sure I'm reading you correctly I want to clarify...
>
> You are trying to maintain carrier order and also trying to dial
> simultaneously. If I understand what you mean, you have two separate
> dialstrings that you want to dial simultaneously and if BOTH fail, THEN you
> want to dial the third dialstring?
>
> If the above is correct then this dialplan snippet SHOULD work:
>
> 
>  data="sofia/gateway/broadvox/6095553828,sofia/gateway/broadvox/7325553828"/>
> 
> 
>
> I also noticed that you're using js in your example... I hope you will
> consider using the dialplan instead. The DP is quick and clean. At the very
> least, if you really need js for some logic then let the js do the logic and
> then transfer back out to the dialplan. That way you will use fewer system
> resources and your project can scale more easily.
>
> -MC
>
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Re: [Freeswitch-users] Loopback and bypass_media

[Phillip Jones]

Can you use this method to make simultaneous calls though while
preserving carrier order? i.e. each call must try carrier 1 first then
carrier 2 (because each carrier terminates a different subset of
numbers)

So - if I did NOT want to support failover I would use:

session:execute("bridge",
"sofia/gateway/broadvox/6095553828,sofia/gateway/broadvox/7325553828");
 << call 6095553828 and 7325553828 at the same time

If I were to use failover and not need multiple destinations then I would use:

session:execute("bridge",
"sofia/gateway/broadvox/6095553828|sofia/gateway/quest/6095553828");
<< call using broadvox first and quest it that fails

I need to combine:

Call 6095553828  and 7325553828 simultaneously and each trying
broadvox then on fail quest.

session:execute("bridge",
"(sofia/gateway/broadvox/6095553828|sofia/gateway/quest/6095553828),(sofia/gateway/broadvox/7325553828,sofia/gateway/quest/7325553828)");

But of course that syntax does not work.

I can not see another syntax that would achieve this?


On Tue, Aug 11, 2009 at 10:58 AM, Brian West wrote:
> This can all be accomplished with standard bridge lines stacked one on top
> of the other with continue_on_fail set to the list of codes you wish to fail
> over with.  You would also want to set hangup_after_bridge=true.  Loopback
> should NEVER be used in this case.
> http://wiki.freeswitch.org/wiki/Channel_Variables#continue_on_fail
> http://wiki.freeswitch.org/wiki/Channel_Variables#hangup_after_bridge
> /b
> On Aug 11, 2009, at 9:46 AM, Phillip Jones wrote:
>
> Thanks for the reply Brian.
>
> I went through this fairly carefully with Mike Jerris - please see
> http://www.nabble.com/Calling-Multiple-Destinations-with-Failover-td24877157.html
>
> I am very open to any other mechanism that allows the calling multiple
> destinations with carrier failover support. I would have thought this
> is a fairly common requirement and is a fundamental requirement in my
> application.
>
> As always - your help is appreciated.
>
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Re: [Freeswitch-users] Loopback and bypass_media

[Phillip Jones]

Thanks for the reply Brian.

I went through this fairly carefully with Mike Jerris - please see
http://www.nabble.com/Calling-Multiple-Destinations-with-Failover-td24877157.html

I am very open to any other mechanism that allows the calling multiple
destinations with carrier failover support. I would have thought this
is a fairly common requirement and is a fundamental requirement in my
application.

As always - your help is appreciated.

On Tue, Aug 11, 2009 at 10:07 AM, Brian West wrote:
> loopback requires media you can't be using it in this manner.  There
> is NO reason to use loopback in the first place.  Can you show me what
> you're doing in the dialplan that requires you to use loopback?
>
> We gave you the rope... now you just have to stop from hanging
> yourself!  :P
>
> /b
>
> On Aug 11, 2009, at 8:59 AM, Phillip Jones wrote:
>
>> It occurred to me that I better check that this still occurs using lua
>> and that this is not a mod_manged issue.
>>
>> The lua script:
>>
>> session:answer();
>> session:execute("set", "bypass_media_after_bridge=true");
>> session:execute("bridge", "loopback/6095553828/default");
>>
>> does produce the same issue. The bridge fails.
>>
>> I am wondering whether this is something I should just put straight
>> into jira.
>
>
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Re: [Freeswitch-users] Loopback and bypass_media

[Phillip Jones]

It occurred to me that I better check that this still occurs using lua
and that this is not a mod_manged issue.

The lua script:

session:answer();
session:execute("set", "bypass_media_after_bridge=true");
session:execute("bridge", "loopback/6095553828/default");

does produce the same issue. The bridge fails.

I am wondering whether this is something I should just put straight into jira.

On Mon, Aug 10, 2009 at 4:35 PM, Phillip Jones wrote:
> Hi there,
>
> I am attempting to do a simple bridge. Leg A is an inbound DID. Leg B
> is terminated through a SIP carrier.
>
> I am unable to use loopback AND bypass_media_after_bridge=true. The
> bridge fails.
>
> Here is my three line application:
>
> Session.Answer();
>
> Session.Execute("set", "bypass_media_after_bridge=true");
>
> Session.Execute("bridge", "loopback/6095553828/default");
>
> FreeSWITCH log: http://pastebin.freeswitch.org/9949
>
> I believe the bridge fails as the SIP-REINVITE is issued and fails.
>
> Using loopback and bypass_media_after_bridge=false works fine. As does
> using bypass_media_after_bridge=true and sending call directly through
> the gateway to which the loopback points.
>
> e.g.  http://pastebin.freeswitch.org/9948
>
> Has anyone else encountered this? Am I missing something?
>
> Any help would be much appreciated!
>
> Thanks
>
>
> Phillip Jones
>

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[Freeswitch-users] Loopback and bypass_media

[Phillip Jones]

Hi there,

I am attempting to do a simple bridge. Leg A is an inbound DID. Leg B
is terminated through a SIP carrier.

I am unable to use loopback AND bypass_media_after_bridge=true. The
bridge fails.

Here is my three line application:

Session.Answer();

Session.Execute("set", "bypass_media_after_bridge=true");

Session.Execute("bridge", "loopback/6095553828/default");

FreeSWITCH log: http://pastebin.freeswitch.org/9949

I believe the bridge fails as the SIP-REINVITE is issued and fails.

Using loopback and bypass_media_after_bridge=false works fine. As does
using bypass_media_after_bridge=true and sending call directly through
the gateway to which the loopback points.

e.g.  http://pastebin.freeswitch.org/9948

Has anyone else encountered this? Am I missing something?

Any help would be much appreciated!

Thanks


Phillip Jones

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Re: [Freeswitch-users] Calling Multiple Destinations with Failover

[Phillip Jones]

Ok - that's great. I will build this out - thanks both for your help
on this. Much appreciated.

On Sat, Aug 8, 2009 at 1:32 PM, Michael Jerris wrote:
>  I think that summary is totally wrong.  Loopback should be used
> here, and this should work to do what you want, just be aware of what
> that means.
>
> Mike
>
>
> On Aug 8, 2009, at 4:24 PM, Phillip Jones wrote:
>
>> Mike/Rupa ,
>>
>> Thanks for your help on this. So I am correct that summarizing that
>> FreeSWITCH does not really support fail over and multiple call
>> destinations because the same mechanism is used to achieve both? And
>> that loopback as a solution is possible but not recommended?
>>
>> Is there any other solution to this? Perhaps a second FS box in the
>> mix?
>>
>> Phil
>>
>>
>> On Sat, Aug 8, 2009 at 1:00 PM, Michael Jerris wrote:
>>> loopback ends up using extra threads which we are only able to drop
>>> later in certain situations so it will decrease your total amount of
>>> calls you can do if your not careful with them.
>>>
>>> Mike
>>>
>>> On Aug 8, 2009, at 1:44 PM, Phillip Jones wrote:
>>>
>>>> Thanks very much for that - very help.
>>>>
>>>> Why would loopback be considered "abuse"? What would be the
>>>> downsides
>>>> of doing this?
>>>>
>>>> On Sat, Aug 8, 2009 at 7:18 AM, Rupa Schomaker wrote:
>>>>>
>>>>>
>>>>> On Sat, Aug 8, 2009 at 9:16 AM, Rupa Schomaker 
>>>>> wrote:
>>>>>>
>>>>>> 
>>>>>
>>>>> That of course, should be:
>>>>>
>>>>>  
>>>>> --
>>>>> -Rupa
>
>
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Re: [Freeswitch-users] Calling Multiple Destinations with Failover

[Phillip Jones]

Mike/Rupa ,

Thanks for your help on this. So I am correct that summarizing that
FreeSWITCH does not really support fail over and multiple call
destinations because the same mechanism is used to achieve both? And
that loopback as a solution is possible but not recommended?

Is there any other solution to this? Perhaps a second FS box in the mix?

Phil


On Sat, Aug 8, 2009 at 1:00 PM, Michael Jerris wrote:
> loopback ends up using extra threads which we are only able to drop
> later in certain situations so it will decrease your total amount of
> calls you can do if your not careful with them.
>
> Mike
>
> On Aug 8, 2009, at 1:44 PM, Phillip Jones wrote:
>
>> Thanks very much for that - very help.
>>
>> Why would loopback be considered "abuse"? What would be the downsides
>> of doing this?
>>
>> On Sat, Aug 8, 2009 at 7:18 AM, Rupa Schomaker wrote:
>>>
>>>
>>> On Sat, Aug 8, 2009 at 9:16 AM, Rupa Schomaker  wrote:
>>>>
>>>> 
>>>
>>> That of course, should be:
>>>
>>>  
>>> --
>>> -Rupa
>>>
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Re: [Freeswitch-users] Call transfer/forward PSTN-SIP-PSTN

[Phillip Jones]

Are your calls coming in on TDM or SIP trunks? Are your calls answered
by FreeSWITCH before you need to redirect them?

On Sat, Aug 8, 2009 at 11:52 AM, Vladimir
Rodionov wrote:
> Actually, this is what I need
>
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_redirect
>
> Will it work with PSTN? Can I redirect incoming PSTN call to another PSTN
> number?
>
> -Vladimir Rodionov
>
>
> On Sat, Aug 8, 2009 at 11:39 AM, Vladimir Rodionov 
> wrote:
>>
>> Yes, I need SIP-REFER (pass back media and signaling to PSTN). I am pretty
>> sure it is doable because voxeo offers this
>> option for their Voice XML customers but I am not interested in a hosted
>> solution at the time - it is quite expensive. As far as I understood, Voip
>> provider MUST have pstn call transfer feature enabled by telecom provider
>> (AT&T for example) and this should work fine with SIP.
>>
>> -Vladimir Rodionov
>>
>>
>> On Sat, Aug 8, 2009 at 10:40 AM, Phillip Jones 
>> wrote:
>>>
>>> Hi there,
>>>
>>> Are you wanting to keep the SIP signaling in FreeSWITCH (SIP re
>>> INVITE) and only pass back the media to the network, or pass back
>>> signaling also (SIP REFER)?
>>>
>>> I know several suppliers who support SIP re INVITE but none that
>>> support SIP REFER.
>>>
>>> Check out http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_deflect
>>> and http://wiki.freeswitch.org/wiki/Bypass_Media
>>>
>>>
>>>
>>> On Sat, Aug 8, 2009 at 9:26 AM, Vladimir Rodionov
>>> wrote:
>>> > Good morning,
>>> > This is the scenario: User 1 (PSTN) calls freeSWITCH (DID). Calls goes
>>> > through PSTN Gateway (1) to freeSWITCH application server (AS) (2).
>>> > AS does some logic and transfers call (or forward) out of Voip provider
>>> > network to another PSTN number User2.
>>> >
>>> >
>>> > This is call bridge
>>> >
>>> >
>>> >     UA1 (PSTN) - ->  UA2 (PSTN)
>>> >   -    -
>>> >     -  (1)   -  (4)
>>> >   -> PSTN Gateway->
>>> >     -    -
>>> >    (2) -    - (3)
>>> >   -> FreeSWITCH ->
>>> >
>>> >
>>> > This is what I want to acomplish
>>> >     (4)
>>> >     UA1 (PSTN) --- ->  UA2 (PSTN)
>>> >     -
>>> >    -   (1)
>>> >     ->  PSTN Gateway->
>>> >  -    -
>>> >    (2)  -    - (3)
>>> >     -> FreeSWITCH ->
>>> >
>>> >
>>> > 1. Can it be implemented in FreeSWITCH?
>>> > 2. Does anybody know Voip providers which support out of network call
>>> > transfer/forwarding to PSTN?
>>> >
>>> > TIA
>>> >
>>> > -Vladimir Rodionov
>>> >
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Re: [Freeswitch-users] Calling Multiple Destinations with Failover

[Phillip Jones]

Thanks very much for that - very help.

Why would loopback be considered "abuse"? What would be the downsides
of doing this?

On Sat, Aug 8, 2009 at 7:18 AM, Rupa Schomaker wrote:
>
>
> On Sat, Aug 8, 2009 at 9:16 AM, Rupa Schomaker  wrote:
>>
>> 
>
> That of course, should be:
>
>  
> --
> -Rupa
>
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Re: [Freeswitch-users] Call transfer/forward PSTN-SIP-PSTN

[Phillip Jones]

Hi there,

Are you wanting to keep the SIP signaling in FreeSWITCH (SIP re
INVITE) and only pass back the media to the network, or pass back
signaling also (SIP REFER)?

I know several suppliers who support SIP re INVITE but none that
support SIP REFER.

Check out http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_deflect
and http://wiki.freeswitch.org/wiki/Bypass_Media



On Sat, Aug 8, 2009 at 9:26 AM, Vladimir Rodionov wrote:
> Good morning,
> This is the scenario: User 1 (PSTN) calls freeSWITCH (DID). Calls goes
> through PSTN Gateway (1) to freeSWITCH application server (AS) (2).
> AS does some logic and transfers call (or forward) out of Voip provider
> network to another PSTN number User2.
>
>
> This is call bridge
>
>
>     UA1 (PSTN) - ->  UA2 (PSTN)
>   -    -
>     -  (1)   -  (4)
>   -> PSTN Gateway->
>     -    -
>    (2) -    - (3)
>   -> FreeSWITCH ->
>
>
> This is what I want to acomplish
>     (4)
>     UA1 (PSTN) --- ->  UA2 (PSTN)
>     -
>    -   (1)
>     ->  PSTN Gateway->
>  -    -
>    (2)  -    - (3)
>     -> FreeSWITCH ->
>
>
> 1. Can it be implemented in FreeSWITCH?
> 2. Does anybody know Voip providers which support out of network call
> transfer/forwarding to PSTN?
>
> TIA
>
> -Vladimir Rodionov
>
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[Freeswitch-users] Calling Multiple Destinations with Failover

[Phillip Jones]

Hi there,

I am trying to implement a scenario where I can terminate calls to
multiple destinations AND have termination carrier fail over.
Currently I can see how to do one or the other. But not both.

Multiple destinations is easy:

 

Failover appears to use the same mechanism:



I can not get my head around how use these together, such that:


- 6095551234 is dialed through SIP_PROVIDER_1
- if NO_ROUTE_DESTINATION then dial 6095551234  through SIP_PROVIDER_2
- Called party does not answer
7325551234 is dialed through SIP_PROVIDER_1
- if NO_ROUTE_DESTINATION then dial 7325551234 through SIP_PROVIDER_2
- Called party answers


This must be a fairly common requirement so any ideas on what I might
be missing would be very welcome.

Thanks


Phillip Jones

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Re: [Freeswitch-users] Fwd: strange behaviour with scheduled jobs, uuid_transfer hangs up one of the legs if freeswitch is not in the media path, lua freeswitch.Session(uuid) tries to call out if chan

[Phillip Jones]

Hi there,

Not sure whether this helps but test this without set bypass_media. In
my setup I have noticed the leg A session ends when bypass_media is
true. Call/bridge continue successfully.

Phillip Jones

On Thu, Aug 6, 2009 at 1:28 PM, Benedikt
Fraunhofer wrote:
> Hello List, Hello *,
>
> First of all the usual excuses: sorry for the bad english and the long
> email, no native speaker and i really tried to make it shorter, but i
> guess this would result in even more "check back"s than it already
> does :)
>
> we're currently running in a weird "lockup"-scenario in our loadtests.
>
> Our setup is the following:
>
> three freeswitch servers, let's call them A(-leg), M(aster), and
> B(-leg) with the goal in mind to initiate calls on M which calls A,
> play some file, bridge to B, limit call length and play (different)
> prompts to A and B if they exceed that limit.
>
> (Note that A and B work fine, regardless of the amount of load we put on them)
> A and B are silly dialplan logic, accepting calls on a certain
> extension after a random delay and playing moh. Before calling
> playback to a localstream they call a lua script which schedules
> hangup somewhere in future (which works flawlessly)
>
> Calls are initiated on M using some hacked up loadgen-script issuing
> http requests like
>   originate [sofiaSyntaxToExtensionOn_A] 6000
> . The 6000 extension on M has the following (xml) dialplan which
> essentially does the following:
> --
> answer()
> ...playback file...
> ...set some callerid stuff
> set bypass_media
> bridge to extension 6009 on B
> --
> we use "execute_on_answer" on the b-leg to run a script which limits
> the length of the call (doesn't matter if it's done via "export
> nolocal" or "inlined" into the data part of the bridge application
> "{execute_on_answer=lua ...}")
>
> 
>
> the lua script "schedula-hangup.lua" does essentially the following:
>
> --
> api = freeswitch.API();
> local res = api:execute("sched_api", "+10 none lua
> lua/c2c-hangup-timeout.lua " .. argv[1]);
> --
>
> the 10 seconds are just to speed up the time until it gets stuck.
>
> this is where things start to go wrong. if I comment out the call to
> the "schedule-hangup" script, everything works fine, even if it's
> under heavy load.
>
> c2c-hangup-timeout.lua does the following:
> --
> local sess = argv[1];
> if(sess)
> then
>    freeswitch.consoleLog("INFO", "c2c-hangup-timeout.lua for uuid "
> .. sess .. "\n");
>
>    api = freeswitch.API();
>    local stillValid = api:execute("uuid_getvar", sess .. "
> Dummy-DoesChannelExists");
>    if(stillValid:sub(1,4) == "-ERR")
>    then
>        log("session uuid " .. sess .. " disappeared (nothing bad)");
>    else
>        -- this is important!!! Otherwise the aleg get's just hung up!
>        api:execute("uuid_media", sess);
>        api:execute("uuid_transfer", sess .. " -both timeout");
>    end
> else -- /if(sess)
>    log("called with nil session?");
> end -- /if(sess)
>
> --
>
> i guess this needs some explanation:
> we get the uuid of the channel as argument in argv[1]. We don't use
>   local session = freeswitch.Session(uuid);
> since if the channel referenced by "uuid" does not exist any longer,
> freeswitch (or the lua bindings) try to interpret the uuid as an
> "originate string" and can't figure out how to call that. So we use a
> dummy api call to get some channel variable. If the channel does not
> exist any longer (A or B already hung up), we get an error message
> starting with "-ERR", otherwise the channel still exists (we get
> "_unset_" as the value, if it's not set) and we continue by getting
> freeswitch back in the media path (uuid_media) and then transferring
> both legs to an extension called "timeout" which plays some prompt and
> finally calls hangup().
>
> If we don't do the uuid_media call, one of the legs gets hung up when
> we transfer them to the extension. This looks like the following on
> the console after issuing "uuid_transfer [uuid] -both timeout"
> (extensions are not the same as in our loadgen example above)
>
>
> --
> 2009-07-23 19:57:19.865703 [NOTICE] switch_ivr.c:1334 Hangup (*)
> sofia/internal/1000 [CS_HIBERNATE] [BLIND_TRANSFER]
> 2009-07-23 19:57:19.865703 [NOTICE] switch_ivr.c:1349 Transfer
> sofia/internal/1...@192.168.179.177:50

[Freeswitch-users] Calling multiple destinations with fail over

[Phillip Jones]

Hi there,

I am trying to implement a scenario where I can terminate calls to
multiple destinations AND have termination carrier fail over.
Currently I can see how to do one or the other. But not both.

Multiple destinations is easy:

 

Failover appears to use the same mechanism:



I can not get my head around how use these together, such that:


- 6095551234 is dialed through SIP_PROVIDER_1
- if NO_ROUTE_DESTINATION then dial 6095551234  through SIP_PROVIDER_2
- Called party does not answer
7325551234 is dialed through SIP_PROVIDER_1
- if NO_ROUTE_DESTINATION then dial 7325551234 through SIP_PROVIDER_2
- Called party answers


This must be a fairly comment requirement so any ideas on what I might
be missing would be very welcome.

Thanks


Phillip Jones

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Re: [Freeswitch-users] Which event contains ORIGINATOR_CANCEL?

[Phillip Jones]

What does

bridge_hangup_cause

give you?

On Fri, Aug 7, 2009 at 12:43 PM, Nicolas Brenner  wrote:
>
> I changed the script to set hangup_after_bridge to false, but still the same 
> thing happens, I get this on the console:
>
> 2009-08-07 12:27:44.229091 [NOTICE] sofia.c:322 Hangup 
> sofia/external/00569xxx [CS_SOFT_EXECUTE] [NORMAL_CLEARING]
> 2009-08-07 12:27:44.229091 [DEBUG] switch_channel.c:1683 Send signal 
> sofia/external/00569xxx [KILL]
> 2009-08-07 12:27:44.229091 [DEBUG] switch_core_session.c:932 Send signal 
> sofia/external/00569xxx [BREAK]
> 2009-08-07 12:27:44.231471 [NOTICE] switch_ivr_originate.c:1994 Hangup 
> sofia/external/005622170039 [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL]
> 2009-08-07 12:27:44.231471 [DEBUG] switch_channel.c:1683 Send signal 
> sofia/external/005622170039 [KILL]
> 2009-08-07 12:27:44.231471 [DEBUG] switch_core_session.c:932 Send signal 
> sofia/external/005622170039 [BREAK]
> 2009-08-07 12:27:44.231471 [DEBUG] switch_ivr_originate.c:2134 Originate 
> Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL]
> 2009-08-07 12:27:44.231471 [DEBUG] switch_core_state_machine.c:398 
> (sofia/external/00569xxx) Running State Change CS_HANGUP
> 2009-08-07 12:27:44.231471 [INFO] mod_dptools.c:2092 Originate Failed.  
> Cause: ORIGINATOR_CANCEL
> 2009-08-07 12:27:44.231471 [NOTICE] c2c.js:1 *** Leg2: 
> NORMAL_CLEARING ***
>
>
> The second to last line comes from the script, and prints the hangup_cause of 
> he session, instead of getting ORIGINATOR_CANCEL, I'm getting 
> NORMAL_CLEARING. Where is the ORIGINATOR_CANCEL value set?
>
>
> Thanks!
>
> Nicolas
>
> On Thu, Aug 6, 2009 at 3:45 PM, Nicolas Brenner  wrote:
>>
>> Hi Matt,
>>
>> Actually I'm explicitly setting hangup_after_bridge to true, think setting 
>> it to false would help? I'm going to try that.
>>
>> Here's the JS code:
>> (Note: session.getVariable() doesn't work, FS complains saying it is not a 
>> function, also tried self.session.getVariable() - that's what the wiki says 
>> - and FS complains that self does not exist)
>>
>> 
>> var uuid = argv[0]; // Call identifier
>> var dialstr1 = argv[1]; // Dial string obtained from previous call to LCR
>> var dialstr2 = argv[2]; // Dial string obtained from previous call to LCR
>> var greeting_snd = "/var/audio/alert.wav";
>>
>> console_log("notice", "*** STARTING C2C Call ***\n");
>> timeout = 30;
>>
>> console_log("notice", "*** DIALING "+dialstr1+" ***\n");
>>
>> //var stUsRing = session.getVariable("us-ring");  // This doesn't work, 
>> self.session.getVariable doesn't work either
>> var stUsRing = "%(2000,4000,440,480)";
>>
>> // Create new_session
>> new_session = new Session(originate_str1);
>> console_log("notice", "*** Leg1: " + new_session.cause + " 
>> ***\n");
>>
>> if (new_session.ready()) {
>>     // log to the console
>>     console_log("notice", "*** Leg1 ("+dialstr1+") CONNECTED! 
>> ***\n");
>>     console_log("notice", "*** Playing greeting sound: 
>> "+greeting_snd+" ***\n");
>>
>>     new_session.execute("sleep", 100);
>>     new_session.execute("playback", greeting_snd);
>>
>>     // Originate second call and bridge
>>     originate_str2 = 
>> "{ignore_early_media=true,originate_timeout="+timeout+",hangup_after_bridge=true,medularis_uuid="+uuid+",c2c_call=true,leg=2}"+dialstr2;
>>
>>     // Create new_session
>>     new_session.execute("bridge", originate_str2);
>>     console_log("notice", "*** Leg2: " + new_session.cause + " 
>> ***\n");
>>
>>     if (new_session.ready()) {
>>     console_log("notice", "*** Leg2 ("+dialstr2+") 
>> CONNECTED! ***\n");
>>     }
>> }
>>
>> exit();
>> 
>>
>> Thanks!
>>
>>
>> Nicolas
>>
>>
>> On Thu, Aug 6, 2009 at 2:25 PM, Matthew Fong  wrote:
>>>
>>> Hi Nicolas,
>>> do you have a copy of the .js code you can paste. I would guess tho, that 
>>> ORIGINATOR_CANCLE might be related to not setting hangup_after_bridge to 
>>> false. Just a guess tho.
>>> Hangup causes can be found here:
>>> http://wiki.freeswitch.org/wiki/Hangup_causes
>>> --matt
>>> hello hunter - hosted predictive dialer & voice broadcasting
>>> http://www.hellohunter.com
>>>
>>> On Thu, Aug 6, 2009 at 9:38 AM, Nicolas Brenner  
>>> wrote:

 I'm bridging 2 calls in a javascript file, I originate the first call and 
 then execute a bridge with an origination string for the second call. If I 
 hangup the first call while trying to make the second call, I get this on 
 the console:

 2009-08-05 16:44:05.69122 [NOTICE] switch_ivr_originate.c:1994 Hangup 
 sofia/external/005622170039 [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL]
 2009-08-05 16:44:05.69122 [DEBUG] switch_channel.c:1683 Send signal 
 sofia/external/005622170039 [KILL]
 2009-08-05 16:44:05.69122 [DEBUG] switch_core_sess

Re: [Freeswitch-users] FreeSWITCH code to emulate Asterisk dialing announcement

[Phillip Jones]

Mike,

I am not familiar with Asterisk so I am not 100% sure this is what you are
looking for, but check out
http://wiki.freeswitch.org/wiki/Freeswitch_IVR_Originate#Answer_confirmation

It may contain what you need

Phillip Jones


On Mon, Aug 3, 2009 at 3:23 PM, Michael Frager wrote:

> Hello,
>
> I'm in the process of moving my VOIP application from Asterisk to
> FreeSWITCH.
>
> I was wondering if it is possible to emulate the call announcement feature
> that is available on Asterisk.
>
> On Asterisk it looks like this, with the "A(...)" parameter:
>
> Dial(SIP/1551212|180|A(connecttone1))
>
> Note that this announcement is only played for the called party, the
> calling party does NOT hear the tone.
>
> I'm guessing this can be done with FreeSWITCH.
>
> Does anyone know how I might accomplish this?
>
>
> Thanks in advance,
>
> -Mike Fragre
>
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Re: [Freeswitch-users] LUA: Independent control of each call leg.

[Phillip Jones]

 >>a prime example is a b-leg "whisper" where you are still playing
"ringing" to the a-party while
>>the b-party answers and hears a message just before connection to the
>>a-party.

You should be able to do this particular function using group_confirm. See
this page in the wiki.

http://wiki.freeswitch.org/wiki/Freeswitch_IVR_Originate#exec_in_answer_confirm

In your case the test.js would contain the whisper. Note that you can send
args to this script.


On Fri, Jul 31, 2009 at 11:01 AM, peely  wrote:

>
> Hi,
>
> I'm trying to develop an application using lua and need to control the
> inbound and outbound legs independently, even when they are switched
> together.
>
> I can initiate the outbound session but I can't seem to bridge without
> losing control of the script.
>
> For example, if I use:
>
>
>local api = freeswitch.API();
>inSession = session;
>inSession:answer();
>inSession:setAutoHangup(false);
>
>
>egSession = freeswitch.Session("sofia/default/my...@mydomain.com");
>egSession:setAutoHangup(false);
>
>if egSession:ready() then
>api:execute("uuid_bridge",inSession.uuid .. " " ..
> egSession.uuid);
>end
>
>while egSession:ready() do
>inSession:sleep(1000);
>end
>
> Then I lose the script entirely, and if I use:
>
>inSession:execute("bridge", "sofia/default/my...@mydomain.com")
>
> Then I lose the ability to control the call whilst the outbound is in
> progress.
>
> Does anyone know a way I can allow ingress to egress calling whilst still
> maintaining script control mid-call? I also need to ingress to hear
> provisional speech during outbound connect. I've looked at conferencing but
> there seems to be quite a lot of automated messaging.
>
>
>
> Thanks,
>
>
> Neil.
>
> --
> View this message in context:
> http://www.nabble.com/LUA%3A-Independent-control-of-each-call-leg.-tp24744087p24744087.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
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Re: [Freeswitch-users] Barge on on prompts

[Phillip Jones]

Ah! That you very much.

Not Session.Break() but:

Session.DtmfReceivedFunction = (d, t) =>
{
  Log.WriteLine(LogLevel.Info, "Received {0} for {1}.", d, t);
  CollectedDigits = d.ToString().Trim();
  return "break";
}


Thanks to you both for your help on this.

On Thu, Jul 23, 2009 at 4:17 PM, David Knell  wrote:

> Hi Phillip,
>
> You need to call FreeSWITCH's break function - I'd guess
> Session.Break(); might do it for you, but no guarantees.
>
> --Dave
>
> > Hi there,
> >
> > Thanks for the reply. That information is extremely useful.
> >
> > Given the code below though - when if I press '1' when the phrase is
> > playing - playing does not stop. It continues. I am looking for a
> > method to barge in and collect & react to digits immediately.
> >
> >
> > Session.DtmfReceivedFunction = (d, t) =>
> > {
> > Log.WriteLine(LogLevel.Info, "Received {0} for
> > {1}.", d, t);
> > Session.StreamFile("", 0);
> > CollectedDigits = d.ToString().Trim();
> > return "";
> >
> > };
> >
> > Session.SayPhrase("msgcount", "187346", "en");
> >
> >
> > Any ideas? I am sure I must be missing something simple.
> >
> > Thanks a lot.
> >
> >
> > Phillip Jones
> >
> >
> >
> >
> >
> > On Thu, Jul 23, 2009 at 2:40 PM, Michael Collins 
> > wrote:
> > I think you might want to check out phrase macros...
> > http://wiki.freeswitch.org/wiki/Speech_Phrase_Management
> > -MC
> >
> >
> > On Thu, Jul 23, 2009 at 11:13 AM, Phillip Jones
> >  wrote:
> >
> >
> > Hi there,
> >
> > Very simple scenario:
> >
> > Session.DtmfReceivedFunction = (d, t) =>
> > {
> > Log.WriteLine(LogLevel.Info, "Received {0} for
> > {1}.", d, t);
> > CollectedDigits = d.ToString().Trim();
> > return "";
> > };
> >
> >
> > Session.flushDigits();
> > Session.StreamFile(VoicemailPromptsDirectory +
> > "abigfile.wav", 0);
> >
> > Question is, it there a way to kill the streaming when
> > the a digit is pressed?
> >
> > I would use the Session.PlayAndGetDigits()
> >
> > but that does not help when want to string things
> > together like:
> >
> > Session.StreamFile(VoicemailPromptsDirectory +
> > "vm-to_delete_the_message.wav", 0);
> > Session.StreamFile(VoicemailPromptsDirectory +
> > "vm-press.wav", 0);
> > Session.Say("7", "en", "number", "pronounced");
> >
> > Any help would be appreciated.
> >
> >
> > ___
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> >
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> >
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> --
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> T: +44 20 3298 2000
> E: d...@3c.co.uk
> W: http://www.3c.co.uk
>
>
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Re: [Freeswitch-users] Barge on on prompts

[Phillip Jones]

Hi there,

Thanks for the reply. That information is extremely useful.

Given the code below though - when if I press '1' when the phrase is playing
- playing does not stop. It continues. I am looking for a method to barge in
and collect & react to digits immediately.


Session.DtmfReceivedFunction = (d, t) =>
{
Log.WriteLine(LogLevel.Info, "Received {0} for {1}.", d,
t);
Session.StreamFile("", 0);
CollectedDigits = d.ToString().Trim();
return "";

};

Session.SayPhrase("msgcount", "187346", "en");


Any ideas? I am sure I must be missing something simple.

Thanks a lot.


Phillip Jones





On Thu, Jul 23, 2009 at 2:40 PM, Michael Collins  wrote:

> I think you might want to check out phrase macros...
> http://wiki.freeswitch.org/wiki/Speech_Phrase_Management
> -MC
>
> On Thu, Jul 23, 2009 at 11:13 AM, Phillip Jones wrote:
>
>> Hi there,
>>
>> Very simple scenario:
>>
>> Session.DtmfReceivedFunction = (d, t) =>
>> {
>> Log.WriteLine(LogLevel.Info, "Received {0} for {1}.", d, t);
>> CollectedDigits = d.ToString().Trim();
>> return "";
>> };
>>
>>
>> Session.flushDigits();
>> Session.StreamFile(VoicemailPromptsDirectory + "abigfile.wav", 0);
>>
>> Question is, it there a way to kill the streaming when the a digit is
>> pressed?
>>
>> I would use the Session.PlayAndGetDigits()
>>
>> but that does not help when want to string things together like:
>>
>> Session.StreamFile(VoicemailPromptsDirectory +
>> "vm-to_delete_the_message.wav", 0);
>> Session.StreamFile(VoicemailPromptsDirectory + "vm-press.wav", 0);
>> Session.Say("7", "en", "number", "pronounced");
>>
>> Any help would be appreciated.
>>
>> ___
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>>
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[Freeswitch-users] Barge on on prompts

[Phillip Jones]

Hi there,

Very simple scenario:

Session.DtmfReceivedFunction = (d, t) =>
{
Log.WriteLine(LogLevel.Info, "Received {0} for {1}.", d, t);
CollectedDigits = d.ToString().Trim();
return "";
};


Session.flushDigits();
Session.StreamFile(VoicemailPromptsDirectory + "abigfile.wav", 0);

Question is, it there a way to kill the streaming when the a digit is
pressed?

I would use the Session.PlayAndGetDigits()

but that does not help when want to string things together like:

Session.StreamFile(VoicemailPromptsDirectory +
"vm-to_delete_the_message.wav", 0);
Session.StreamFile(VoicemailPromptsDirectory + "vm-press.wav", 0);
Session.Say("7", "en", "number", "pronounced");

Any help would be appreciated.
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Re: [Freeswitch-users] .NET demo / hangupHook()

[Phillip Jones]

Perfect - thank you very much!




On Thu, Jul 16, 2009 at 5:32 PM, Michael Giagnocavo wrote:

>  The debug log has this:
>
> 65.2009-07-16 16:48:41.432200 [DEBUG] switch_cpp.cpp:1124 AppFunction is in
> hangupCallback.
>
> 66.2009-07-16 16:48:41.432200 [WARNING] switch_cpp.cpp:1124 Thread will not
> be aborted because Hangup was called from the Run thread.
>
>
>
> The problem is the Demo doesn’t have code to actually set hangupHook as the
> handler. Adding something like this to the app demo code:
>
>
>
> Session.HangupFunction = hangupHook;
>
>
>
> Should fix it.
>
>
>
> -Michael
>
>
>
> *From:* freeswitch-users-boun...@lists.freeswitch.org [mailto:
> freeswitch-users-boun...@lists.freeswitch.org] *On Behalf Of *Phillip
> Jones
> *Sent:* Thursday, July 16, 2009 3:08 PM
> *To:* freeswitch-users@lists.freeswitch.org
> *Subject:* [Freeswitch-users] .NET demo / hangupHook()
>
>
>
> Hi there,
>
> I am looking at the FreeSWITCH.Demo.AppDemo class. Have it running nicely
> from a soft phone - all is well.
>
> However I am not seeing the hangupHook() method fired, when I hangup.
>
> Debug log for an example call is at: http://pastebin.freeswitch.org/9744
>
> Reminder of demo code is here: http://pastebin.freeswitch.org/9745
>
> dialplan.xml is simply:
>
> 
> 
>  
> 
>  
>
> Are there changes to the demo required to get this method firing?
>
>
> Any help much appreciated.
>
>
> Phillip Jones
>
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[Freeswitch-users] .NET demo / hangupHook()

[Phillip Jones]

Hi there,

I am looking at the FreeSWITCH.Demo.AppDemo class. Have it running nicely
from a soft phone - all is well.

However I am not seeing the hangupHook() method fired, when I hangup.

Debug log for an example call is at: http://pastebin.freeswitch.org/9744

Reminder of demo code is here: http://pastebin.freeswitch.org/9745

dialplan.xml is simply:



 

 

Are there changes to the demo required to get this method firing?


Any help much appreciated.


Phillip Jones
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Re: [Freeswitch-users] leg_timeout

[Phillip Jones]

Hey Guys,

I took a look at the source that Anthony updated. I see this:

} else if (!strcasecmp((char *) hi->name, "group_confirm_file")) {
ok = 1;
} else if (!strcasecmp((char *) hi->name, "group_confirm_*clear*_timeout"))
{
ok = 1;
 } else if (!strcasecmp((char *) hi->name, "forked_dial")) {

and:

if (switch_true(switch_event_get_header(var_event,
"group_confirm_*cancel*_timeout")))
{
oglobals.cancel_timeout = -1;
}

I updated the *group_confirm_clear_timeout *to *group_confirm_cancel_timeout
* and recompiled and this is now working just great.

Thanks very much for incorporating this. It is much appreciated.


Phillip Jones



On Tue, Jul 14, 2009 at 6:22 PM, Phillip Jones  wrote:

> Hi there,
>
> I downloaded the latest trunk, compiled and updated. Still no joy I am
> afraid.
>
> This is the log file in pastebin - http://pastebin.freeswitch.org/9712
>
> Code in my managed DLL is at: http://pastebin.freeswitch.org/9715
>
> Dialplan binds to above:
>
>   
>
>
> confirm.js is at: http://pastebin.freeswitch.org/9713
>
>
> Thanks again for your help on this.
>
>
>
>
>
> On Tue, Jul 14, 2009 at 2:35 PM, Phillip Jones wrote:
>
>> Ah - SVN Trunk - thought you meant DID trunk!!! My bad.
>>
>> Sorry - understand now! Will recompile and let you know.
>>
>> On Tue, Jul 14, 2009 at 2:27 PM, Anthony Minessale <
>> anthony.miness...@gmail.com> wrote:
>>
>>> You actually updated your code and recompiled it all too?
>>> This param was added about 30 seconds before I sent you the email.
>>>
>>>
>>>
>>> On Tue, Jul 14, 2009 at 1:14 PM, Phillip Jones wrote:
>>>
>>>> Thanks for your response.
>>>>
>>>> That does not seem to work. Here is my code:
>>>>
>>>> if(Session.Ready())
>>>> {
>>>>  Session.Execute("set", "ignore_early_media=true");
>>>>  Session.Execute("set", "hangup_after_bridge=true");
>>>>  Session.Execute("set", "ringback=${us-ring}");
>>>>
>>>> Session.Answer();
>>>> string Caller_ID_Number =
>>>> this.Session.GetVariable("caller_id_number");
>>>> Session.Execute("set", "group_confirm_key=exec");
>>>> *Session.Execute("set", "group_confirm_cancel_timeout=true");
>>>> *Session.Execute("set", "group_confirm_file=javascript
>>>> confirm.js");
>>>>  Session.Execute("bridge",
>>>> "[leg_timeout=20,leg_confirm=y]sofia/gateway/broadvox/6095553828");
>>>> }
>>>> Session.Hangup("USER_BUSY");
>>>>
>>>> I also tried *group_confirm_cancel_leg_timeout* just in case.
>>>>
>>>> Am I missing something?
>>>>
>>>>
>>>>
>>>>
>>>> On Tue, Jul 14, 2009 at 1:37 PM, Michael Collins 
>>>> wrote:
>>>>
>>>>> FYI,
>>>>> This has been added to the wiki:
>>>>>
>>>>> http://wiki.freeswitch.org/wiki/Channel_Variables#group_confirm_cancel_timeout
>>>>>
>>>>> -MC
>>>>>
>>>>>
>>>>> On Tue, Jul 14, 2009 at 10:20 AM, Anthony Minessale <
>>>>> anthony.miness...@gmail.com> wrote:
>>>>>
>>>>>> update to trunk and try setting
>>>>>> group_confirm_cancel_timeout=true
>>>>>>
>>>>>> let me know if it works
>>>>>>
>>>>>>
>>>>>> On Tue, Jul 14, 2009 at 12:02 PM, Phillip Jones >>>>> > wrote:
>>>>>>
>>>>>>> Hi,
>>>>>>>
>>>>>>> Thanks for the reply.
>>>>>>>
>>>>>>> >> Both "leg_timeout" and "leg_progress_timeout" are for controlling
>>>>>>> how long to wait prior to the B-leg answering.
>>>>>>>
>>>>>>> I think this is my point. leg_timeout seems to control how long to
>>>>>>> wait prior to the bridge completeing, not the B-leg answering.
>>>>>>>
>>>>>>>  In my situation I am using:
>>>>>>>
>>>>>>> Session.Execute("set", "group_confirm_key=exec");
>>>>>>>

Re: [Freeswitch-users] leg_timeout

[Phillip Jones]

Hi there,

I downloaded the latest trunk, compiled and updated. Still no joy I am
afraid.

This is the log file in pastebin - http://pastebin.freeswitch.org/9712

Code in my managed DLL is at: http://pastebin.freeswitch.org/9715

Dialplan binds to above:

  


confirm.js is at: http://pastebin.freeswitch.org/9713


Thanks again for your help on this.




On Tue, Jul 14, 2009 at 2:35 PM, Phillip Jones  wrote:

> Ah - SVN Trunk - thought you meant DID trunk!!! My bad.
>
> Sorry - understand now! Will recompile and let you know.
>
> On Tue, Jul 14, 2009 at 2:27 PM, Anthony Minessale <
> anthony.miness...@gmail.com> wrote:
>
>> You actually updated your code and recompiled it all too?
>> This param was added about 30 seconds before I sent you the email.
>>
>>
>>
>> On Tue, Jul 14, 2009 at 1:14 PM, Phillip Jones wrote:
>>
>>> Thanks for your response.
>>>
>>> That does not seem to work. Here is my code:
>>>
>>> if(Session.Ready())
>>> {
>>>  Session.Execute("set", "ignore_early_media=true");
>>>  Session.Execute("set", "hangup_after_bridge=true");
>>>  Session.Execute("set", "ringback=${us-ring}");
>>>
>>> Session.Answer();
>>> string Caller_ID_Number =
>>> this.Session.GetVariable("caller_id_number");
>>> Session.Execute("set", "group_confirm_key=exec");
>>> *Session.Execute("set", "group_confirm_cancel_timeout=true");
>>> *Session.Execute("set", "group_confirm_file=javascript confirm.js");
>>>  Session.Execute("bridge",
>>> "[leg_timeout=20,leg_confirm=y]sofia/gateway/broadvox/6095553828");
>>> }
>>> Session.Hangup("USER_BUSY");
>>>
>>> I also tried *group_confirm_cancel_leg_timeout* just in case.
>>>
>>> Am I missing something?
>>>
>>>
>>>
>>>
>>> On Tue, Jul 14, 2009 at 1:37 PM, Michael Collins wrote:
>>>
>>>> FYI,
>>>> This has been added to the wiki:
>>>>
>>>> http://wiki.freeswitch.org/wiki/Channel_Variables#group_confirm_cancel_timeout
>>>>
>>>> -MC
>>>>
>>>>
>>>> On Tue, Jul 14, 2009 at 10:20 AM, Anthony Minessale <
>>>> anthony.miness...@gmail.com> wrote:
>>>>
>>>>> update to trunk and try setting
>>>>> group_confirm_cancel_timeout=true
>>>>>
>>>>> let me know if it works
>>>>>
>>>>>
>>>>> On Tue, Jul 14, 2009 at 12:02 PM, Phillip Jones 
>>>>> wrote:
>>>>>
>>>>>> Hi,
>>>>>>
>>>>>> Thanks for the reply.
>>>>>>
>>>>>> >> Both "leg_timeout" and "leg_progress_timeout" are for controlling
>>>>>> how long to wait prior to the B-leg answering.
>>>>>>
>>>>>> I think this is my point. leg_timeout seems to control how long to
>>>>>> wait prior to the bridge completeing, not the B-leg answering.
>>>>>>
>>>>>>  In my situation I am using:
>>>>>>
>>>>>> Session.Execute("set", "group_confirm_key=exec");
>>>>>> Session.Execute("set", "group_confirm_file=javascript confirm.js");
>>>>>>
>>>>>> my confirm.js prompts leg_b for 1 (to take the call). If leg_timeout
>>>>>> is set to 10 you have 10 seconds to answer the call AND press 1.
>>>>>>
>>>>>> I just want call_timeout to be satisfied when the call is answered.
>>>>>> Not when the called party presses 1 and the bridge is complete.
>>>>>>
>>>>>> I am new all this so I will work out how to use the pastebin etc.
>>>>>>
>>>>>> Thanks for your help.
>>>>>>
>>>>>>
>>>>>> Phillip Jones
>>>>>>
>>>>>>
>>>>>>
>>>>>> On Tue, Jul 14, 2009 at 12:16 PM, Michael Collins >>>>> > wrote:
>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>  On Tue, Jul 14, 2009 at 8:35 AM, Phillip Jones <
>>>>>>> pjinthe...@gmail.com> wrote:
>>>>>>>
>>&

Re: [Freeswitch-users] Pastebin.freeswitch.org

[Phillip Jones]

Thank you!

Not that obvious in IE actually - I should stick to FF.


On Tue, Jul 14, 2009 at 5:43 PM, Michael Collins  wrote:

>
>
> On Tue, Jul 14, 2009 at 2:27 PM, Phillip Jones wrote:
>
>> Hi there,
>>
>> I am sure I am missing something. Can someone point out where to signup
>> for a username / password to pastebin.freeswitch.org.
>>
>> I am pulling my hair out and feel kinda stupid asking this.
>
>
> That's okay, you are neither the first nor the last person to get caught by
> this one. :)
> -MC
>
> ___
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[Freeswitch-users] Pastebin.freeswitch.org

[Phillip Jones]

Hi there,

I am sure I am missing something. Can someone point out where to signup for
a username / password to pastebin.freeswitch.org.

I am pulling my hair out and feel kinda stupid asking this.

Thanks


Phillip Jones
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Re: [Freeswitch-users] leg_timeout

[Phillip Jones]

Ah - SVN Trunk - thought you meant DID trunk!!! My bad.

Sorry - understand now! Will recompile and let you know.

On Tue, Jul 14, 2009 at 2:27 PM, Anthony Minessale <
anthony.miness...@gmail.com> wrote:

> You actually updated your code and recompiled it all too?
> This param was added about 30 seconds before I sent you the email.
>
>
>
> On Tue, Jul 14, 2009 at 1:14 PM, Phillip Jones wrote:
>
>> Thanks for your response.
>>
>> That does not seem to work. Here is my code:
>>
>> if(Session.Ready())
>> {
>>  Session.Execute("set", "ignore_early_media=true");
>>  Session.Execute("set", "hangup_after_bridge=true");
>>  Session.Execute("set", "ringback=${us-ring}");
>>
>> Session.Answer();
>> string Caller_ID_Number =
>> this.Session.GetVariable("caller_id_number");
>> Session.Execute("set", "group_confirm_key=exec");
>> *Session.Execute("set", "group_confirm_cancel_timeout=true");
>> *Session.Execute("set", "group_confirm_file=javascript confirm.js");
>>  Session.Execute("bridge",
>> "[leg_timeout=20,leg_confirm=y]sofia/gateway/broadvox/6095553828");
>> }
>> Session.Hangup("USER_BUSY");
>>
>> I also tried *group_confirm_cancel_leg_timeout* just in case.
>>
>> Am I missing something?
>>
>>
>>
>>
>> On Tue, Jul 14, 2009 at 1:37 PM, Michael Collins wrote:
>>
>>> FYI,
>>> This has been added to the wiki:
>>>
>>> http://wiki.freeswitch.org/wiki/Channel_Variables#group_confirm_cancel_timeout
>>>
>>> -MC
>>>
>>>
>>> On Tue, Jul 14, 2009 at 10:20 AM, Anthony Minessale <
>>> anthony.miness...@gmail.com> wrote:
>>>
>>>> update to trunk and try setting
>>>> group_confirm_cancel_timeout=true
>>>>
>>>> let me know if it works
>>>>
>>>>
>>>> On Tue, Jul 14, 2009 at 12:02 PM, Phillip Jones 
>>>> wrote:
>>>>
>>>>> Hi,
>>>>>
>>>>> Thanks for the reply.
>>>>>
>>>>> >> Both "leg_timeout" and "leg_progress_timeout" are for controlling
>>>>> how long to wait prior to the B-leg answering.
>>>>>
>>>>> I think this is my point. leg_timeout seems to control how long to wait
>>>>> prior to the bridge completeing, not the B-leg answering.
>>>>>
>>>>>  In my situation I am using:
>>>>>
>>>>> Session.Execute("set", "group_confirm_key=exec");
>>>>> Session.Execute("set", "group_confirm_file=javascript confirm.js");
>>>>>
>>>>> my confirm.js prompts leg_b for 1 (to take the call). If leg_timeout is
>>>>> set to 10 you have 10 seconds to answer the call AND press 1.
>>>>>
>>>>> I just want call_timeout to be satisfied when the call is answered. Not
>>>>> when the called party presses 1 and the bridge is complete.
>>>>>
>>>>> I am new all this so I will work out how to use the pastebin etc.
>>>>>
>>>>> Thanks for your help.
>>>>>
>>>>>
>>>>> Phillip Jones
>>>>>
>>>>>
>>>>>
>>>>> On Tue, Jul 14, 2009 at 12:16 PM, Michael Collins 
>>>>> wrote:
>>>>>
>>>>>>
>>>>>>
>>>>>>  On Tue, Jul 14, 2009 at 8:35 AM, Phillip Jones >>>>> > wrote:
>>>>>>
>>>>>>> Hi there,
>>>>>>>
>>>>>>> Here is my call flow:
>>>>>>>
>>>>>>> 1) leg A is bridged to leg B
>>>>>>> 2) when leg B is answered I play a confirm script - "please 1 to
>>>>>>> accept this call"
>>>>>>>
>>>>>>> I only want leg B to ring 20 seconds. BUT when the caller party
>>>>>>> answers, he should have as long as he needs to press 1.
>>>>>>>
>>>>>>> "leg_timeout" seems to be in play until the bridge is completed. I
>>>>>>> need it to reset when leg b is answered.
>>>>>>>
>>>>>>
>>>>>> Correct. Both "leg

Re: [Freeswitch-users] leg_timeout

[Phillip Jones]

Thanks for your response.

That does not seem to work. Here is my code:

if(Session.Ready())
{
 Session.Execute("set", "ignore_early_media=true");
 Session.Execute("set", "hangup_after_bridge=true");
 Session.Execute("set", "ringback=${us-ring}");

Session.Answer();
string Caller_ID_Number = this.Session.GetVariable("caller_id_number");
Session.Execute("set", "group_confirm_key=exec");
*Session.Execute("set", "group_confirm_cancel_timeout=true");
*Session.Execute("set", "group_confirm_file=javascript confirm.js");
Session.Execute("bridge",
"[leg_timeout=20,leg_confirm=y]sofia/gateway/broadvox/6095553828");
}
Session.Hangup("USER_BUSY");

I also tried *group_confirm_cancel_leg_timeout* just in case.

Am I missing something?




On Tue, Jul 14, 2009 at 1:37 PM, Michael Collins  wrote:

> FYI,
> This has been added to the wiki:
>
> http://wiki.freeswitch.org/wiki/Channel_Variables#group_confirm_cancel_timeout
>
> -MC
>
>
> On Tue, Jul 14, 2009 at 10:20 AM, Anthony Minessale <
> anthony.miness...@gmail.com> wrote:
>
>> update to trunk and try setting
>> group_confirm_cancel_timeout=true
>>
>> let me know if it works
>>
>>
>> On Tue, Jul 14, 2009 at 12:02 PM, Phillip Jones wrote:
>>
>>> Hi,
>>>
>>> Thanks for the reply.
>>>
>>> >> Both "leg_timeout" and "leg_progress_timeout" are for controlling how
>>> long to wait prior to the B-leg answering.
>>>
>>> I think this is my point. leg_timeout seems to control how long to wait
>>> prior to the bridge completeing, not the B-leg answering.
>>>
>>>  In my situation I am using:
>>>
>>> Session.Execute("set", "group_confirm_key=exec");
>>> Session.Execute("set", "group_confirm_file=javascript confirm.js");
>>>
>>> my confirm.js prompts leg_b for 1 (to take the call). If leg_timeout is
>>> set to 10 you have 10 seconds to answer the call AND press 1.
>>>
>>> I just want call_timeout to be satisfied when the call is answered. Not
>>> when the called party presses 1 and the bridge is complete.
>>>
>>> I am new all this so I will work out how to use the pastebin etc.
>>>
>>> Thanks for your help.
>>>
>>>
>>> Phillip Jones
>>>
>>>
>>>
>>> On Tue, Jul 14, 2009 at 12:16 PM, Michael Collins 
>>> wrote:
>>>
>>>>
>>>>
>>>>  On Tue, Jul 14, 2009 at 8:35 AM, Phillip Jones 
>>>> wrote:
>>>>
>>>>> Hi there,
>>>>>
>>>>> Here is my call flow:
>>>>>
>>>>> 1) leg A is bridged to leg B
>>>>> 2) when leg B is answered I play a confirm script - "please 1 to accept
>>>>> this call"
>>>>>
>>>>> I only want leg B to ring 20 seconds. BUT when the caller party
>>>>> answers, he should have as long as he needs to press 1.
>>>>>
>>>>> "leg_timeout" seems to be in play until the bridge is completed. I need
>>>>> it to reset when leg b is answered.
>>>>>
>>>>
>>>> Correct. Both "leg_timeout" and "leg_progress_timeout" are for
>>>> controlling how long to wait prior to the B-leg answering.
>>>> (leg_progress_timeout specifies how long to wait for any kind of progress,
>>>> be it early media of some sort, ringing, or an answer.)
>>>>
>>>>>
>>>>> I tried resetting the leg_timeout in the confirm script after leg b is
>>>>> answered. I also tried using leg_progress_timeout. Neither seemed to work.
>>>>>
>>>>
>>>> What exactly are you trying to do? The two variables you've mentioned
>>>> shouldn't have any effect on the call after it has been established.
>>>>
>>>>>
>>>>> Any help or suggestions would be welcome.
>>>>>
>>>>
>>>> Could you pastebin your dialplan and a debug log of a call that does not
>>>> work? See this page for some handy tips on using pastebin and collecting
>>>> information for debugging purposes:
>>>> http://wiki.freeswitch.org/wiki/Reporting_Bugs#Debugging_Steps
>>>>
>>>> -MC
>>>>
>>>>>
>>

Re: [Freeswitch-users] leg_timeout

[Phillip Jones]

Hi,

Thanks for the reply.

>> Both "leg_timeout" and "leg_progress_timeout" are for controlling how
long to wait prior to the B-leg answering.

I think this is my point. leg_timeout seems to control how long to wait
prior to the bridge completeing, not the B-leg answering.

 In my situation I am using:

Session.Execute("set", "group_confirm_key=exec");
Session.Execute("set", "group_confirm_file=javascript confirm.js");

my confirm.js prompts leg_b for 1 (to take the call). If leg_timeout is set
to 10 you have 10 seconds to answer the call AND press 1.

I just want call_timeout to be satisfied when the call is answered. Not when
the called party presses 1 and the bridge is complete.

I am new all this so I will work out how to use the pastebin etc.

Thanks for your help.


Phillip Jones



On Tue, Jul 14, 2009 at 12:16 PM, Michael Collins wrote:

>
>
>  On Tue, Jul 14, 2009 at 8:35 AM, Phillip Jones wrote:
>
>> Hi there,
>>
>> Here is my call flow:
>>
>> 1) leg A is bridged to leg B
>> 2) when leg B is answered I play a confirm script - "please 1 to accept
>> this call"
>>
>> I only want leg B to ring 20 seconds. BUT when the caller party answers,
>> he should have as long as he needs to press 1.
>>
>> "leg_timeout" seems to be in play until the bridge is completed. I need it
>> to reset when leg b is answered.
>>
>
> Correct. Both "leg_timeout" and "leg_progress_timeout" are for controlling
> how long to wait prior to the B-leg answering. (leg_progress_timeout
> specifies how long to wait for any kind of progress, be it early media of
> some sort, ringing, or an answer.)
>
>>
>> I tried resetting the leg_timeout in the confirm script after leg b is
>> answered. I also tried using leg_progress_timeout. Neither seemed to work.
>>
>
> What exactly are you trying to do? The two variables you've mentioned
> shouldn't have any effect on the call after it has been established.
>
>>
>> Any help or suggestions would be welcome.
>>
>
> Could you pastebin your dialplan and a debug log of a call that does not
> work? See this page for some handy tips on using pastebin and collecting
> information for debugging purposes:
> http://wiki.freeswitch.org/wiki/Reporting_Bugs#Debugging_Steps
>
> -MC
>
>>
>> Phillip Jones
>>
>> ___
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>>
>
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[Freeswitch-users] leg_timeout

[Phillip Jones]

Hi there,

Here is my call flow:

1) leg A is bridged to leg B
2) when leg B is answered I play a confirm script - "please 1 to accept this
call"

I only want leg B to ring 20 seconds. BUT when the caller party answers, he
should have as long as he needs to press 1.

"leg_timeout" seems to be in play until the bridge is completed. I need it
to reset when leg b is answered.

I tried resetting the leg_timeout in the confirm script after leg b is
answered. I also tried using leg_progress_timeout. Neither seemed to work.

Any help or suggestions would be welcome.

Phillip Jones
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Re: [Freeswitch-users] managed_mod directories

[Phillip Jones]

Got it! Thanks very much for that clarification.

Phil

On Sat, Jul 11, 2009 at 1:43 PM, Jeff Lenk  wrote:

>
> Hi,
>
> The base dll (FreeSWITCH.Managed.dll) is loaded from /mod - additional
> managed dlls are loaded from /mod/managed. This is designed to allow your
> dll's to be built and maintained independant of the FS build files. You can
> simply just drop your dlls into mod/managed and they will be loaded and
> available for use(this happens at FS startup).
>
> The base managed dll (FreeSWITCH.Managed.dll) is only really supposed to be
> used for loader support and the demo classes - you should place your code
> in
> your own dll.
>
> - Jeff
>
>
> Phillip Jones-2 wrote:
> >
> > Hi,
> >
> > If I place the DLL in mod\managed I get the following error:
> >
> > [err] mod_managed.cpp:287 Assembly::LoadFrom failed:
> > system.IO.FileNotFoundException: Could not load file or assembly
> > 'file:///c:program files (x86)\Freeswitch\mod\freeSWITCH.Managed.dll' or
> > one
> > of its dependencies. The system could not find the file specified.
> >
> > As I said. When I place freeSWITCH.Managed.dll straight into \mod then
> > everything works fine.
> >
> > Thanks
> >
> >
> > Phil
> >
> >
> > On Sat, Jul 11, 2009 at 9:15 AM, Diego Toro  wrote:
> >
> >> Hello,
> >>
> >> What error do you get when dll is put on mod/managed ?, I work with
> dll's
> >> on mod/managed although I changed loadfile by loadfrom on loader.cs.
> >>
> >> Diego
> >>
> >>
> >> --- On *Fri, 7/10/09, Phillip Jones * wrote:
> >>
> >>
> >> From: Phillip Jones 
> >> Subject: Re: [Freeswitch-users] managed_mod directories
> >> To: freeswitch-users@lists.freeswitch.org
> >> Date: Friday, July 10, 2009, 6:22 PM
> >>
> >>
> >> It is looking in mod.
> >>
> >> It required the mod\managed directory, but if I place my dll in
> >> mod\managed
> >> it fails. DLL must be in mod - mod\managed is empty.
> >>
> >> My app works fine though
> >>
> >>
> >> Phil
> >>
> >>
> >> On Fri, Jul 10, 2009 at 4:48 PM, Michael Giagnocavo
> >>  http://us.mc335.mail.yahoo.com/mc/compose?to=...@giagnocavo.net>
> >> > wrote:
> >>
> >>>  You’re saying that it requires the managed DLL to be in both the mod
> >>> and
> >>> mod\managed directory?  What error do you get if it’s only in mod? It’s
> >>> been
> >>> months, but I just checked loader.cs and it looks explicitly in the
> >>> managed
> >>> directory to resolve assemblies as well as to scan to load them.
> >>>
> >>> -Michael
> >>>
> >>>  *From:*
> >>> freeswitch-users-boun...@lists.freeswitch.org<
> http://us.mc335.mail.yahoo.com/mc/compose?to=freeswitch-users-boun...@lists.freeswitch.org
> >[mailto:
> >>> freeswitch-users-boun...@lists.freeswitch.org<
> http://us.mc335.mail.yahoo.com/mc/compose?to=freeswitch-users-boun...@lists.freeswitch.org
> >]
> >>> *On Behalf Of *Phillip Jones
> >>> *Sent:* Friday, July 10, 2009 11:46 AM
> >>> *To:*
> >>> freeswitch-users@lists.freeswitch.org<
> http://us.mc335.mail.yahoo.com/mc/compose?to=freeswitch-us...@lists.freeswitch.org
> >
> >>> *Subject:* [Freeswitch-users] managed_mod directories
> >>>
> >>> Hi there,
> >>>
> >>>
> >>> Using windows with the pre-compiled binary / msi found via the WIKI
> >>>
> >>> Using mod_managed with no problems however:
> >>>
> >>> mod_managed appears to require I create a directory 'managed' under
> >>> C:\Program Files (x86)\FreeSWITCH\mod
> >>>
> >>> BUT also requires that I place my .dll in C:\Program Files
> >>> (x86)\FreeSWITCH\mod and NOT C:\Program Files
> >>> (x86)\FreeSWITCH\mod\managed
> >>>
> >>> thus making C:\Program Files (x86)\FreeSWITCH\mod\managed redundant
> >>>
> >>> Anyone else seen this behavior?
> >>>
> >>> Thanks!
> >>>
> >>>
> >>> Phillip Jones
> >>>
> >>> ___
> >>> Freeswitch-users mailing list
> >>> Freeswitch-users@lists.freeswitch.org<
> http://us.mc335.mail.yahoo.com/mc/compose?to=freeswitch-us...@lists

Re: [Freeswitch-users] managed_mod directories

[Phillip Jones]

Hi,

If I place the DLL in mod\managed I get the following error:

[err] mod_managed.cpp:287 Assembly::LoadFrom failed:
system.IO.FileNotFoundException: Could not load file or assembly
'file:///c:program files (x86)\Freeswitch\mod\freeSWITCH.Managed.dll' or one
of its dependencies. The system could not find the file specified.

As I said. When I place freeSWITCH.Managed.dll straight into \mod then
everything works fine.

Thanks


Phil


On Sat, Jul 11, 2009 at 9:15 AM, Diego Toro  wrote:

> Hello,
>
> What error do you get when dll is put on mod/managed ?, I work with dll's
> on mod/managed although I changed loadfile by loadfrom on loader.cs.
>
> Diego
>
>
> --- On *Fri, 7/10/09, Phillip Jones * wrote:
>
>
> From: Phillip Jones 
> Subject: Re: [Freeswitch-users] managed_mod directories
> To: freeswitch-users@lists.freeswitch.org
> Date: Friday, July 10, 2009, 6:22 PM
>
>
> It is looking in mod.
>
> It required the mod\managed directory, but if I place my dll in mod\managed
> it fails. DLL must be in mod - mod\managed is empty.
>
> My app works fine though
>
>
> Phil
>
>
> On Fri, Jul 10, 2009 at 4:48 PM, Michael Giagnocavo 
> http://us.mc335.mail.yahoo.com/mc/compose?to=...@giagnocavo.net>
> > wrote:
>
>>  You’re saying that it requires the managed DLL to be in both the mod and
>> mod\managed directory?  What error do you get if it’s only in mod? It’s been
>> months, but I just checked loader.cs and it looks explicitly in the managed
>> directory to resolve assemblies as well as to scan to load them.
>>
>> -Michael
>>
>>  *From:* 
>> freeswitch-users-boun...@lists.freeswitch.org<http://us.mc335.mail.yahoo.com/mc/compose?to=freeswitch-users-boun...@lists.freeswitch.org>[mailto:
>> freeswitch-users-boun...@lists.freeswitch.org<http://us.mc335.mail.yahoo.com/mc/compose?to=freeswitch-users-boun...@lists.freeswitch.org>]
>> *On Behalf Of *Phillip Jones
>> *Sent:* Friday, July 10, 2009 11:46 AM
>> *To:* 
>> freeswitch-users@lists.freeswitch.org<http://us.mc335.mail.yahoo.com/mc/compose?to=freeswitch-us...@lists.freeswitch.org>
>> *Subject:* [Freeswitch-users] managed_mod directories
>>
>> Hi there,
>>
>>
>> Using windows with the pre-compiled binary / msi found via the WIKI
>>
>> Using mod_managed with no problems however:
>>
>> mod_managed appears to require I create a directory 'managed' under
>> C:\Program Files (x86)\FreeSWITCH\mod
>>
>> BUT also requires that I place my .dll in C:\Program Files
>> (x86)\FreeSWITCH\mod and NOT C:\Program Files (x86)\FreeSWITCH\mod\managed
>>
>> thus making C:\Program Files (x86)\FreeSWITCH\mod\managed redundant
>>
>> Anyone else seen this behavior?
>>
>> Thanks!
>>
>>
>> Phillip Jones
>>
>> ___
>> Freeswitch-users mailing list
>> Freeswitch-users@lists.freeswitch.org<http://us.mc335.mail.yahoo.com/mc/compose?to=freeswitch-us...@lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> -Inline Attachment Follows-
>
> ___
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Re: [Freeswitch-users] managed_mod directories

[Phillip Jones]

It is looking in mod.

It required the mod\managed directory, but if I place my dll in mod\managed
it fails. DLL must be in mod - mod\managed is empty.

My app works fine though


Phil


On Fri, Jul 10, 2009 at 4:48 PM, Michael Giagnocavo wrote:

>  You’re saying that it requires the managed DLL to be in both the mod and
> mod\managed directory?  What error do you get if it’s only in mod? It’s been
> months, but I just checked loader.cs and it looks explicitly in the managed
> directory to resolve assemblies as well as to scan to load them.
>
>
>
> -Michael
>
>
>
> *From:* freeswitch-users-boun...@lists.freeswitch.org [mailto:
> freeswitch-users-boun...@lists.freeswitch.org] *On Behalf Of *Phillip
> Jones
> *Sent:* Friday, July 10, 2009 11:46 AM
> *To:* freeswitch-users@lists.freeswitch.org
> *Subject:* [Freeswitch-users] managed_mod directories
>
>
>
> Hi there,
>
>
> Using windows with the pre-compiled binary / msi found via the WIKI
>
> Using mod_managed with no problems however:
>
> mod_managed appears to require I create a directory 'managed' under
> C:\Program Files (x86)\FreeSWITCH\mod
>
> BUT also requires that I place my .dll in C:\Program Files
> (x86)\FreeSWITCH\mod and NOT C:\Program Files (x86)\FreeSWITCH\mod\managed
>
> thus making C:\Program Files (x86)\FreeSWITCH\mod\managed redundant
>
> Anyone else seen this behavior?
>
> Thanks!
>
>
> Phillip Jones
>
> ___
> 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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Re: [Freeswitch-users] managed_mod directories

[Phillip Jones]

Hi,

Thanks for the reply.

My DDL is working fine. Just not in the mod\managed directory.

What is the mod\managed directory for? It is required but not used?

Phil


On Fri, Jul 10, 2009 at 3:04 PM,  wrote:

> Hi, check whether the dll references to other dll's, in that case you
> should put the references in managed directory.
>
> Check this link
> http://lists.freeswitch.org/pipermail/freeswitch-dev/2009-April/002232.html ,
> may be your case
>
> Diego
>
>
>
> --- On *Fri, 7/10/09, Phillip Jones * wrote:
>
>
> From: Phillip Jones 
> Subject: [Freeswitch-users] managed_mod directories
> To: freeswitch-users@lists.freeswitch.org
> Date: Friday, July 10, 2009, 12:46 PM
>
>
> Hi there,
>
> Using windows with the pre-compiled binary / msi found via the WIKI
>
> Using mod_managed with no problems however:
>
> mod_managed appears to require I create a directory 'managed' under
> C:\Program Files (x86)\FreeSWITCH\mod
>
> BUT also requires that I place my .dll in C:\Program Files
> (x86)\FreeSWITCH\mod and NOT C:\Program Files (x86)\FreeSWITCH\mod\managed
>
> thus making C:\Program Files (x86)\FreeSWITCH\mod\managed redundant
>
> Anyone else seen this behavior?
>
> Thanks!
>
>
> Phillip Jones
>
> -Inline Attachment Follows-
>
> ___
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>
>
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[Freeswitch-users] managed_mod directories

[Phillip Jones]

Hi there,

Using windows with the pre-compiled binary / msi found via the WIKI

Using mod_managed with no problems however:

mod_managed appears to require I create a directory 'managed' under
C:\Program Files (x86)\FreeSWITCH\mod

BUT also requires that I place my .dll in C:\Program Files
(x86)\FreeSWITCH\mod and NOT C:\Program Files (x86)\FreeSWITCH\mod\managed

thus making C:\Program Files (x86)\FreeSWITCH\mod\managed redundant

Anyone else seen this behavior?

Thanks!


Phillip Jones
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Re: [Freeswitch-users] mod_say_en directory location

[Phillip Jones]

Ok - forget this one - I did a fresh install from the pre-compiled windows
binary/msi - referenced on the wiki - and every thing is working as it
should be.

Thanks

On Thu, Jul 9, 2009 at 2:46 PM, Phillip Jones  wrote:

> Thanks for the response.
>
> I don't think so - the trace states:
>
> [ERR] mod_sndfile.c: 192 Error Opening File [C:\Program Files
> (x86)\Freeswitch\sounds\en\digits/1.wav] [System error : The system cannot
> find the path specified.]
>
> I created a 16000 directory to see whether that would help, and it did not.
>
> My C:\Program Files (x86)\Freeswitch\conf\lang\en\en.xml contains:
>
> 
>tts-engine="cepstral" tts-voice="callie">
>  
> 
>   
>   
> 
>
> Am I correct in thinking this is where the sound file dir for digits would
> be specified?
>
>
>
> On Thu, Jul 9, 2009 at 12:38 PM, Michael Collins wrote:
>
>> Is the call perhaps at 16kHz and it's looking for non-installed 16kHz
>> sound files?
>> -MC
>>
>> On Thu, Jul 9, 2009 at 9:06 AM, Phillip Jones wrote:
>>
>>> Hi there,
>>>
>>> I have a very simply script that speaks back some digits, as so:
>>>
>>> session:execute("say", "en number iterated 1234");
>>>
>>> However, to get this to work successfully I have had to move the 'digits'
>>> directory to:
>>>
>>> C:\Program Files (x86)\Freeswitch\sounds\en
>>>
>>> from the default:
>>>
>>> C:\Program Files (x86)\Freeswitch\sounds\en\us\callie\digits\8000
>>>
>>>
>>> This is a clean install of FreeSWITCH - so I am wondering why I needed to
>>> do this, what have not configured correctly?
>>>
>>> As you can see I am using windows with a resent build (3 days) from svn.
>>>
>>> Any help appreciated.
>>>
>>> Thanks
>>>
>>>
>>> Phillip Jones
>>>
>>>
>>>
>>>
>>>
>>> ___
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>>>
>>>
>>
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>>
>
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Re: [Freeswitch-users] mod_say_en directory location

[Phillip Jones]

Thanks for the response.

I don't think so - the trace states:

[ERR] mod_sndfile.c: 192 Error Opening File [C:\Program Files
(x86)\Freeswitch\sounds\en\digits/1.wav] [System error : The system cannot
find the path specified.]

I created a 16000 directory to see whether that would help, and it did not.

My C:\Program Files (x86)\Freeswitch\conf\lang\en\en.xml contains:


  
 

  
  


Am I correct in thinking this is where the sound file dir for digits would
be specified?


On Thu, Jul 9, 2009 at 12:38 PM, Michael Collins  wrote:

> Is the call perhaps at 16kHz and it's looking for non-installed 16kHz sound
> files?
> -MC
>
> On Thu, Jul 9, 2009 at 9:06 AM, Phillip Jones wrote:
>
>> Hi there,
>>
>> I have a very simply script that speaks back some digits, as so:
>>
>> session:execute("say", "en number iterated 1234");
>>
>> However, to get this to work successfully I have had to move the 'digits'
>> directory to:
>>
>> C:\Program Files (x86)\Freeswitch\sounds\en
>>
>> from the default:
>>
>> C:\Program Files (x86)\Freeswitch\sounds\en\us\callie\digits\8000
>>
>>
>> This is a clean install of FreeSWITCH - so I am wondering why I needed to
>> do this, what have not configured correctly?
>>
>> As you can see I am using windows with a resent build (3 days) from svn.
>>
>> Any help appreciated.
>>
>> Thanks
>>
>>
>> Phillip Jones
>>
>>
>>
>>
>>
>> ___
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>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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>>
>
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[Freeswitch-users] mod_say_en directory location

[Phillip Jones]

Hi there,

I have a very simply script that speaks back some digits, as so:

session:execute("say", "en number iterated 1234");

However, to get this to work successfully I have had to move the 'digits'
directory to:

C:\Program Files (x86)\Freeswitch\sounds\en

from the default:

C:\Program Files (x86)\Freeswitch\sounds\en\us\callie\digits\8000


This is a clean install of FreeSWITCH - so I am wondering why I needed to do
this, what have not configured correctly?

As you can see I am using windows with a resent build (3 days) from svn.

Any help appreciated.

Thanks


Phillip Jones
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Re: [Freeswitch-users] SIP re-invite / bypass_media

[Phillip Jones]

Used:

session.execute("set","bypass_media_after_bridge=true");
in the confirm.js script and that works perfectly!

Thank you for you help!
On Thu, Jul 2, 2009 at 1:05 PM, Anthony Minessale <
anthony.miness...@gmail.com> wrote:

> try setting bypass_media_after_bridge=true on the session in your confirm
> script
>
>
>
> On Thu, Jul 2, 2009 at 11:53 AM, Phillip Jones wrote:
>
>> Thanks for responding and for your help.
>>
>> The xml and confirm.js are attached below. Basically trying to
>> bypass_media after the leg B presses 1 to accept the call. I tried,
>> using bypass_media_after_bridge=true, but the re-invite appears to be done
>> before the confirm.js, So the media is successfully rerouted, but BEFORE the
>> leg b never gets hear a prompt or gets the opportunity to press 1.
>>
>> To get round this I am trying to manually bypass_media in the confirm.js
>> script with apiExecute("uuid_media", "off " + session.uuid);. However only
>> the B leg is reinvited (and media is routed correctly). I don't see the A
>> leg reinvite, and then a BYE is issueed on both legs.
>>
>> 
>> 
>>
>> 
>> 
>> 
>> <
>> 
>>
>> 
>> 
>> 
>>
>> This is the confirm.js:
>>
>> // confirm.js - FreeSwitch call confirmation script
>> // (c) 2009 - St‚phane Alnet
>> // License: GPL2 or above
>> console_log("info", "Destination: "+ session.destination + "\n");
>> if(!session.getVariable('leg_confirm'))
>> {
>> console_log("info", "No need to confirm, connect the call!\n");
>> exit();
>> }
>> var confirmed = false;
>> var confirmation_digit = "1";
>> var try_count = 6;
>> var prompt_file = "prompts/ToAcceptThisCallPress1.wav";
>> function onInput( session, type, data, arg ) {
>> if ( type == "dtmf" ) {
>> console_log( "info", "Got digit " + data.digit + "\n" );
>> if ( data.digit == confirmation_digit ) {
>> confirmed = true;
>> console_log( "info", "Confirming session..\n" );
>> return(false);
>> }
>> }
>> return(true);
>> }
>> if ( session.ready() )
>> {
>> session.answer();
>> session.flushDigits();
>> console_log("info", "Starting confirmation\n");
>> var count = try_count;
>> while( session.ready() && ! confirmed && count-- > 0 )
>> {
>> session.execute("sleep","200");
>> session.streamFile( prompt_file, onInput );
>> }
>>
>> if( ! confirmed )
>> {
>> console_log("info", "Not confirmed\n");
>> session.hangup();
>> }
>> else
>> {
>>  *apiExecute("uuid_media", "off " + session.uuid);*
>> console_log("info", "Confirmed\n");
>> }
>> }
>> else
>> {
>> console_log("info", "Session is not ready.\n");
>> }
>>
>>
>>
>>
>> On Thu, Jul 2, 2009 at 12:19 PM, Anthony Minessale <
>> anthony.miness...@gmail.com> wrote:
>>
>>> I would need to know more details about what you are doing.
>>>
>>> you could set the variable bypass_media_after_bridge=true on the a leg
>>> before you call the b leg and use the group_confirm feature to get the
>>> caller
>>> to press the key.
>>>
>>>
>>>
>>> On Thu, Jul 2, 2009 at 10:41 AM, Phillip Jones wrote:
>>>
>>>> Thanks for that.
>>>>
>>>> That seems to successfully re-invite and re-route the the B leg - but
>>>> does not reinvite the A leg and then immediately issues a "bye" on both
>>>> legs.
>>>>
>>>> Do I have to do something to reinvite that A leg?
>>>>
>>>>   On Wed, Jul 1, 2009 at 7:06 PM, Anthony Minessale <
>>>> anthony.miness...@gmail.com> wrote:
>>>>
>>>>> try
>>>>> apiExecute("uuid_media", "off " + session.uuid);
>>>>>
>>>>>
>>>>>
>>>>>   On Wed, Jul 1, 2009 at 3:22 PM, Phillip Jones 
>>>>> wrote:
>>>>>
>>>>>>   Hi there,
>>>>>>
>>>>>> I was wondering whether it is possible to have FreeSwitch go into
>>>>>> bypass_media mode on demand?
>>>>>>
>>

Re: [Freeswitch-users] SIP re-invite / bypass_media

[Phillip Jones]

 Thanks for responding and for your help.

The xml and confirm.js are attached below. Basically trying to bypass_media
after the leg B presses 1 to accept the call. I tried,
using bypass_media_after_bridge=true, but the re-invite appears to be done
before the confirm.js, So the media is successfully rerouted, but BEFORE the
leg b never gets hear a prompt or gets the opportunity to press 1.

To get round this I am trying to manually bypass_media in the confirm.js
script with apiExecute("uuid_media", "off " + session.uuid);. However only
the B leg is reinvited (and media is routed correctly). I don't see the A
leg reinvite, and then a BYE is issueed on both legs.







<






This is the confirm.js:

// confirm.js - FreeSwitch call confirmation script
// (c) 2009 - St‚phane Alnet
// License: GPL2 or above
console_log("info", "Destination: "+ session.destination + "\n");
if(!session.getVariable('leg_confirm'))
{
console_log("info", "No need to confirm, connect the call!\n");
exit();
}
var confirmed = false;
var confirmation_digit = "1";
var try_count = 6;
var prompt_file = "prompts/ToAcceptThisCallPress1.wav";
function onInput( session, type, data, arg ) {
if ( type == "dtmf" ) {
console_log( "info", "Got digit " + data.digit + "\n" );
if ( data.digit == confirmation_digit ) {
confirmed = true;
console_log( "info", "Confirming session..\n" );
return(false);
}
}
return(true);
}
if ( session.ready() )
{
session.answer();
session.flushDigits();
console_log("info", "Starting confirmation\n");
var count = try_count;
while( session.ready() && ! confirmed && count-- > 0 )
{
session.execute("sleep","200");
session.streamFile( prompt_file, onInput );
}

if( ! confirmed )
{
console_log("info", "Not confirmed\n");
session.hangup();
}
else
{
*apiExecute("uuid_media", "off " + session.uuid);*
console_log("info", "Confirmed\n");
}
}
else
{
console_log("info", "Session is not ready.\n");
}




On Thu, Jul 2, 2009 at 12:19 PM, Anthony Minessale <
anthony.miness...@gmail.com> wrote:

> I would need to know more details about what you are doing.
>
> you could set the variable bypass_media_after_bridge=true on the a leg
> before you call the b leg and use the group_confirm feature to get the
> caller
> to press the key.
>
>
>
> On Thu, Jul 2, 2009 at 10:41 AM, Phillip Jones wrote:
>
>> Thanks for that.
>>
>> That seems to successfully re-invite and re-route the the B leg - but does
>> not reinvite the A leg and then immediately issues a "bye" on both legs.
>>
>> Do I have to do something to reinvite that A leg?
>>
>>   On Wed, Jul 1, 2009 at 7:06 PM, Anthony Minessale <
>> anthony.miness...@gmail.com> wrote:
>>
>>> try
>>> apiExecute("uuid_media", "off " + session.uuid);
>>>
>>>
>>>
>>>   On Wed, Jul 1, 2009 at 3:22 PM, Phillip Jones wrote:
>>>
>>>>   Hi there,
>>>>
>>>> I was wondering whether it is possible to have FreeSwitch go into
>>>> bypass_media mode on demand?
>>>>
>>>> For instance, leg a bridges to leg b - leg b is invited to accept the
>>>> call by pressing 1. I want to go to bypass_media (do a SIP reinvite to
>>>> reroute the media) after the one is pressed.
>>>>
>>>> Currently I am issuing the following from my js script that prompts for
>>>> the 1:
>>>>
>>>> session.apiExecute("uuid_media",session.uuid);
>>>>
>>>> Not working however.
>>>>
>>>> Any help to get me going would be appreciated.
>>>>
>>>> Thanks
>>>>
>>>> Phillip Jones.
>>>>
>>>> ___
>>>> 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
>>>
&

Re: [Freeswitch-users] SIP re-invite / bypass_media

[Phillip Jones]

Thanks for that.

That seems to successfully re-invite and re-route the the B leg - but does
not reinvite the A leg and then immediately issues a "bye" on both legs.

Do I have to do something to reinvite that A leg?

On Wed, Jul 1, 2009 at 7:06 PM, Anthony Minessale <
anthony.miness...@gmail.com> wrote:

> try
> apiExecute("uuid_media", "off " + session.uuid);
>
>
>
>   On Wed, Jul 1, 2009 at 3:22 PM, Phillip Jones wrote:
>
>>   Hi there,
>>
>> I was wondering whether it is possible to have FreeSwitch go into
>> bypass_media mode on demand?
>>
>> For instance, leg a bridges to leg b - leg b is invited to accept the call
>> by pressing 1. I want to go to bypass_media (do a SIP reinvite to reroute
>> the media) after the one is pressed.
>>
>> Currently I am issuing the following from my js script that prompts for
>> the 1:
>>
>> session.apiExecute("uuid_media",session.uuid);
>>
>> Not working however.
>>
>> Any help to get me going would be appreciated.
>>
>> Thanks
>>
>> Phillip Jones.
>>
>> ___
>> 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
>
>
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[Freeswitch-users] SIP re-invite / bypass_media

[Phillip Jones]

Hi there,

I was wondering whether it is possible to have FreeSwitch go into
bypass_media mode on demand?

For instance, leg a bridges to leg b - leg b is invited to accept the call
by pressing 1. I want to go to bypass_media (do a SIP reinvite to reroute
the media) after the one is pressed.

Currently I am issuing the following from my js script that prompts for the
1:

session.apiExecute("uuid_media",session.uuid);

Not working however.

Any help to get me going would be appreciated.

Thanks

Phillip Jones.
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