Re: [asterisk-users] Difference between 1.4.x and 1.6.x?

[Jose Arias]

For changes between 1.4 and 1.6 you might find useful this one:
http://svn.digium.com/svn/asterisk/tags/1.6.0/CHANGES

For changes between 1.6 branches:
http://svn.digium.com/svn/asterisk/tags/1.6.1.0/CHANGES
http://svn.digium.com/svn/asterisk/tags/1.6.2.0-beta1/CHANGES

Regards
Jose
2009/8/4 Michael Cunningham 

> Thanks Leif,
>
> That cleared up the versioning.. Is there a list of new features in 1.6.x
> versus the 1.4.x version?
>
>   On Mon, Aug 3, 2009 at 4:34 PM, Leif Madsen <
> leif.mad...@asteriskdocs.org> wrote:
>
>>  Michael Cunningham wrote:
>> > Forgive me if this is a FAQ question but I didnt see anything on the
>> > website
>> > of forum spelling out the difference between 1.4.x and 1.6.x
>> >
>> > Obviously 1.6.x is in development. Is it stable enough for production
>>  use?
>> > What are the new features being implemented in 1.6.x?
>> >
>> > Will Cepstral work with 1.6.x?
>>
>> This may be a useful article from asterisk.org for you to read:
>>
>> http://www.asterisk.org/node/48602
>>
>> Leif Madsen.
>>
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>
>
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[asterisk-users] Matching Originate action with its NewChannel event

[Jose Arias]

An application commanding asterisk with AMI is going to launch lots of
concurrent calls in very few seconds using the Originate AMI command but
it's also going to need to be able to cancel very quickly any call of them
even before each OriginateResponse event comes in. All the calls will be
done by the same trunk (a trunking enabled channel). But there's a problem
for canceling any call: there's no way to know what channel to hangup to
because all channel prefixes in the NewChannel event are the same (the
trunking channel one) and although the Originate action has an ActionId
property, it isn't available in the NewChannel event but only in the
OriginateResponse event, being very late. I've read some of you are using
the CallerId property but in this case it's not an option because the
application needs to establish the same callerId for all of them. Is there
any solution using AMI? I'm planning to use asterisk 1.4.18
Thanks
Jose
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Re: [asterisk-users] Asterisk CSTA

[Jose Arias]

I just see how recurrent this subject is. I have some experience on
developing CTI systems and I know how hard to develop a CSTA link can be.
However I'm a newbie in Asterisk and this is the cause why I asked for some
CSTA gateways already developed. Unfortunately www.opencsta.org looks like
an empty site and sourceforge.net/projects/oscsta also looks like an empty
project. I think I don't have enough expertise in Asterisk to start a
project like this but I'm sure I would be able to colaborate if someone
decided to give a try.
Regards
Jose

2009/7/27 John Todd 

>
> Jose -
>There have been several discussions about an Asterisk/OSS CSTA
> gateway or API extension set, but it's a fairly complex undertaking
> which to my knowledge has not been completed or even started in
> earnest by anyone yet.  Personally, I think that an Asterisk-based
> CSTA API would be great - it would open a huge number of "enterprise"
> applications up for plug-and-play use (or at least, much closer to
> plug-and-play than they are today.)  If you're interested in
> developing such a beast, take the conversation to the asterisk-dev
> mailing list and start to put some code on the screen.  :-)
>
> Obvious, but relevant URLs:
>   http://www.google.com/search?hl=en&q=csta+asterisk&aq=f&oq=&aqi=
>   http://sourceforge.net/projects/oscsta/
>   http://www.opencsta.org/
>
>
> JT
>
>
> On Jul 25, 2009, at 7:21 AM, Jose Arias wrote:
>
> > Thanks Steve. But I couldn't find anything about a CSTA to AMI
> > gateway for asterisk at the quintum site. All I be able to find are
> > TDM and FXO/FXS to SIP gateways among others. I'm talking about (and
> > I think gergis.rasmy too) 3rd party call control gateways, not
> > interoperable gateways. By the way, what about an open source csta
> > gateway project?
> > Jose
> > 2009/7/24 Steve Totaro 
> > Without any research, I would check out the Quintum lineup.
> >
> > They have feature sets which are amazing (and confusing as all heck)
> > and work great once configured.
> >
> > Thanks,
> > Steve T
> >
> >
> > On Fri, Jul 24, 2009 at 1:03 PM, Jose Arias  wrote:
> > Do you know what names those gateways have?
> > Jose
> >
> > 2009/7/24 Olivier 
> >
> >
> >
> > 2009/7/22 gergis.rasmy 
> >
> > does Asterisk suppoet CSTA protocol for CTI applications?
> >
> > No it doesn't but I've heard some gateways exist (software
> > translating CST to AMI).
> >
> > Regards
> >
> >
> >
> > --
> > Thanks,
> > Steve Totaro
> > +18887771888 (Toll Free)
> > +12409381212 (Cell)
> > +12024369784 (Skype)
> >
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> ---
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> email:jt...@digium.com
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> 445 Jan Davis Drive NW -  Huntsville AL 35806  -   USA
> direct: +1-256-428-6083 http://www.digium.com/
>
>
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Re: [asterisk-users] Asterisk CSTA

[Jose Arias]

Thanks Steve. But I couldn't find anything about a CSTA to AMI gateway for
asterisk at the quintum site. All I be able to find are TDM and FXO/FXS to
SIP gateways among others. I'm talking about (and I think gergis.rasmy too)
3rd party call control gateways, not interoperable gateways. By the way,
what about an open source csta gateway project?
Jose
2009/7/24 Steve Totaro 

> Without any research, I would check out the Quintum lineup.
>
> They have feature sets which are amazing (and confusing as all heck) and
> work great once configured.
>
> Thanks,
> Steve T
>
>
> On Fri, Jul 24, 2009 at 1:03 PM, Jose Arias  wrote:
>
>> Do you know what names those gateways have?
>> Jose
>>
>> 2009/7/24 Olivier 
>>
>>
>>>
>>> 2009/7/22 gergis.rasmy 
>>>
>>>>  does Asterisk suppoet CSTA protocol for CTI applications?
>>>>
>>>
>>> No it doesn't but I've heard some gateways exist (software translating
>>> CST to AMI).
>>>
>>> Regards
>>>
>>>>
>>>>
>>>>
> --
> Thanks,
> Steve Totaro
> +18887771888 (Toll Free)
> +12409381212 (Cell)
> +12024369784 (Skype)
>
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Re: [asterisk-users] Asterisk CSTA

[Jose Arias]

Do you know what names those gateways have?
Jose

2009/7/24 Olivier 

>
>
> 2009/7/22 gergis.rasmy 
>
>>  does Asterisk suppoet CSTA protocol for CTI applications?
>>
>
> No it doesn't but I've heard some gateways exist (software translating CST
> to AMI).
>
> Regards
>
>>
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>
>
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Re: [asterisk-users] Scalability and stability matters

[Jose Arias]

Many thanks Matt,

I heard asterisk had some problems with registering over 100 SIP endpoints
and I was worried about how much the transcoding load could be for over 100
concurrents calls too. I expect to be over these figures. Regarding the AMI
connection, yes, there will be only one, like any third-party cti-link but
my concern was about how many commands an events asterisk is able to handle
without becoming in a bottleneck.

You said you're using about 8 patches. Are all of them to make sure the
stability and scalability of the system? Well, one of them is the AsyncAGI
patch, isn't? Is there anyone to mach originate commands with new_channel
events?

I'm planning to use asterisk 1.4.18

Regards
Jose


2009/7/21 Matt Florell 

> On 7/21/09, Jose Arias  wrote:
> > Hi all,
> >
> > I'm planning to develop a custom autodialer application which will be
> > dealing with its own model for agents and queues, therefore it won't use
> > neither asterisk agents nor asterisk queues, nor asterisk cdr. The
> > application will supply the whole reporting and agent managing features
> by
> > itself.
> >
> > The application will command asterisk through an AMI telnet connection
> using
> > only the originate, redirect and hangup AMI commands plus the stream file
> > AGI command (AsyncAGI patch will be required).
> >
> > The application will make outbound calls, then they will be redirected on
> > the fly to dynamically defined meetme rooms, then the application will
> call
> > extensions (registered endpoints) where it will know there are available
> > agents in order to redirect them to the previous meetme rooms. If the
> > application launched more calls than available agents it would play
> prompts
> > while waiting for agents to become available.
> >
> > Since the planned features set from asterisk to be used by the
> application
> > will be very short, but the figures can be very large (in terms of
> > concurrent calls, registered endpoints, traffic on the AMI port, etc..)
>  I
> > would appreciate if anybody can help me to find out what's the more
> suitable
> > asterisk version to use in terms of scalability and stability:
> >
> > - concurrent registered endpoints (SIP and IAX)
> > - concurrent two and tree party meetme rooms (whatever codec can be used)
> > - concurrent mixmonitor recordings
> > - concurrent playings for prompts
> > - commands and events rate on the AMI port
> >
> > It's important to notice the advanced features from asterisk aren't a
> > priority.
> >
> > I already looked over some links like
> > http://www.voip-info.org/wiki/view/Asterisk+dimensioning
> > and others but I found more questions than answers there.
> >
> > Thanks in advance
> > Jose
> >
>
> This sounds a lot like ViciDial, which does use meetme instead of
> Asterisk Queues/Agents, is already engineered to be multi-server, is
> capable of placing 200,000+ outbound calls per server per day, has a
> web-based GUI for configuring the system and a web-based agent
> interface.
>
>
> - concurrent registered endpoints (SIP and IAX)
>
> Doesn't really matter, we've done 500+ on a single server before and
> it didn't really affect load much. As for number of agents, we are
> usually conservative on that front, usually we keep it under 50 agents
> per outbound server, but we have done 100 before.
>
> - concurrent two and tree party meetme rooms (whatever codec can be used)
>
> Everything is transcoded in a meetme room to slin. ViciDial does
> everything in Meetme, and while it does use slightly more resources
> than Asterisk Queues, it is more stable and offers more flexibility
>
> - concurrent mixmonitor recordings
>
> We do not recommend using mxmonitor. It is better to have a custom
> recording handling script. And if you are using Meetme for everything
> you don't have to bother mixing recordings anyway.
>
> - concurrent playings for prompts
>
> This depends on a lot of different things, if load or playback quality
> becomes an issue then you should put prompts on a RAM drive or tmpfs
>
> - commands and events rate on the AMI port
>
> Use a single point(or a few limited points) of entry to the AMI to
> keep it working well. You should not have an AMI connection for each
> agent.
>
>
> We currently use a version of 1.4.21.2 that has about 8 patches
> applied to it, and we have found it to be very stable in production.
>
> MATT---
>
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[asterisk-users] Scalability and stability matters

[Jose Arias]

Hi all,

I'm planning to develop a custom autodialer application which will be
dealing with its own model for agents and queues, therefore it won't use
neither asterisk agents nor asterisk queues, nor asterisk cdr. The
application will supply the whole reporting and agent managing features by
itself.

The application will command asterisk through an AMI telnet connection using
only the originate, redirect and hangup AMI commands plus the stream file
AGI command (AsyncAGI patch will be required).

The application will make outbound calls, then they will be redirected on
the fly to dynamically defined meetme rooms, then the application will call
extensions (registered endpoints) where it will know there are available
agents in order to redirect them to the previous meetme rooms. If the
application launched more calls than available agents it would play prompts
while waiting for agents to become available.

Since the planned features set from asterisk to be used by the application
will be very short, but the figures can be very large (in terms of
concurrent calls, registered endpoints, traffic on the AMI port, etc..)  I
would appreciate if anybody can help me to find out what's the more suitable
asterisk version to use in terms of scalability and stability:

- concurrent registered endpoints (SIP and IAX)
- concurrent two and tree party meetme rooms (whatever codec can be used)
- concurrent mixmonitor recordings
- concurrent playings for prompts
- commands and events rate on the AMI port
It's important to notice the advanced features from asterisk aren't a
priority.

I already looked over some links like
http://www.voip-info.org/wiki/view/Asterisk+dimensioning and others but I
found more questions than answers there.

Thanks in advance
Jose
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Re: [asterisk-users] How run AsyncAGI commands in background

[Jose Arias]

Hi Moy,
many thanks for clarifying. I'll do some further investigations about it 
and I'll post the result here.

Regards
Jose


Moises Silva escribió:

On Fri, Jun 19, 2009 at 5:32 AM, Jose Arias wrote:
  

Hi Moy,

I'm using an asterisk 1.4.18 from scratch patched with the last AsyncAGI
patch, which fixes a bug about stopping AsyncAGI applications, as may be you
can recall from the thread [asterisk-users] async agi question in
http://lists.digium.com/pipermail/asterisk-users/2009-April/230488.html.

This patched asterisk works fine and it stops the async agi applications
launched from the AsyncAGI loop before the Redirect as it's expected. It's
for that I don't think stopping the mixmonitor application launched from the
AsyncAGI loop would be a bug if I redirect the call. I would be only getting
the same behavior than I got with the stream file application as you
explained it should be at
http://www.moythreads.com/wordpress/2007/12/24/asterisk-asynchronous-agi/#comment-365

I'm only asking if there's any way to prevent stopping applications launched
on a channel from the AsyncAGI loop if this channel is redirected afterward,
with something like a continue_running_in_background flag in the previous
AGI invocation from AMI. Of course, it bring us the problem we'll need some
kind of identifier and some stop action to be able to stop those
applications running in background launched from the AsyncAGI loop

Anyway, as you asked me some days ago, I have published at
http://docs.google.com/View?id=ahfnfrcrh3rr_4dkcx9dgw a simple configuration
and a simple scenario in order you can try to reproduce what I'm saying.

I don't need anyone to do anything for me. I'm willing to do the work, I
like programming and trying new things as well, but I'll need some
guidelines to go straight ahead.




Jose, the thing is that MixMonitor IS a background application in
nature, that's why I say is unexpected that after a redirect the
recording no longer works. In fact, that's why StopMixMonitor
application is needed, because all MixMonitor does is to launch a
background thread that hooks into the channel audio, then the channel
continues to execute other applications in the dial plan while this
background thread monitors its audio, on a redirect StopMixMonitor
thread should continue saving audio until StopMixMonitor is called.

  


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Re: [asterisk-users] How run AsyncAGI commands in background

[Jose Arias]

Hi Moy,

I'm using an asterisk 1.4.18 from scratch patched with the last AsyncAGI 
patch, which fixes a bug about stopping AsyncAGI applications, as may be 
you can recall from the thread [asterisk-users] async agi question in 
http://lists.digium.com/pipermail/asterisk-users/2009-April/230488.html.


This patched asterisk works fine and it stops the async agi applications 
launched from the AsyncAGI loop before the Redirect as it's expected. 
It's for that I don't think stopping the mixmonitor application launched 
from the AsyncAGI loop would be a bug if I redirect the call. I would be 
only getting the same behavior than I got with the stream file 
application as you explained it should be at 
http://www.moythreads.com/wordpress/2007/12/24/asterisk-asynchronous-agi/#comment-365 



I'm only asking if there's any way to prevent stopping applications 
launched on a channel from the AsyncAGI loop if this channel is 
redirected afterward, with something like a 
continue_running_in_background flag in the previous AGI invocation from 
AMI. Of course, it bring us the problem we'll need some kind of 
identifier and some stop action to be able to stop those applications 
running in background launched from the AsyncAGI loop


Anyway, as you asked me some days ago, I have published at 
http://docs.google.com/View?id=ahfnfrcrh3rr_4dkcx9dgw a simple 
configuration and a simple scenario in order you can try to reproduce 
what I'm saying.


I don't need anyone to do anything for me. I'm willing to do the work, I 
like programming and trying new things as well, but I'll need some 
guidelines to go straight ahead.


Thanks all
Jose

Moises Silva escribió:

On Sun, Jun 7, 2009 at 4:37 PM, Jose Arias wrote:
  

Hi Moy,

I'll do it so, but for your answer, it seems you are thinking about it as it
could be a bug. I don't think so. I mean: the redirect action on a channel
in AsyncAGI stops the current agi execution. It's the normal behavior. It's
the way to stop a playfile on a channel if it was previously launched from
AsyncAGI: making a redirect out of the AsyncAGI loop.

Therefore, when I realized the previously launched EXE MixMonitor AsyncAGI
execution was stopping after doing a redirect to meetme, I didn't think it
was a bug. I though what I was needing it was a way to tell AsyncAGI, "hey,
don't stop this agi execution on the channel, even it will be redirected out
of AGI" on an individual basis for each AsyncAGI EXEC command launched.

Thanks
Jose



The way I see it if you make EXEC MixMonitor inside AsyncAGI loop and
then redirect to MeetMe and you don't get the audio recorded, then
it's not a normal behavior, MixMonitor is an application that should
passively monitor the channel audio independently of where the channel
is (regardless of whether the command was executed in Async AGI or
dial plan or whatever). However you are also using an old asterisk
version and is not likely you can report a bug unless you upgrade to
the latest Asterisk and reproduce without a patched Asterisk (for
example executing EXEC MixMonitor inside a regular AGI script and then
redirect to MeetMe).

  


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Re: [asterisk-users] How run AsyncAGI commands in background

[Jose Arias]

Hi Moy, 

I'll do it so, but for your answer, it seems you are thinking about it 
as it could be a bug. I don't think so. I mean: the redirect action on a 
channel in AsyncAGI stops the current agi execution. It's the normal 
behavior. It's the way to stop a playfile on a channel if it was 
previously launched from AsyncAGI: making a redirect out of the AsyncAGI 
loop.


Therefore, when I realized the previously launched EXE MixMonitor 
AsyncAGI execution was stopping after doing a redirect to meetme, I 
didn't think it was a bug. I though what I was needing it was a way to 
tell AsyncAGI, "hey, don't stop this agi execution on the channel, even 
it will be redirected out of AGI" on an individual basis for each 
AsyncAGI EXEC command launched.


Thanks
Jose

Moises Silva escribió:

> then it should work, create a *simple* extensions.conf and pastebin it
> along with instructions so I can try to reproduce.
>
> > On Sat, Jun 6, 2009 at 5:02 PM, Jose Arias<http://lists.digium.com/mailman/listinfo/asterisk-users>> wrote:

> >/ Hi,
/> >/ Asterisk 1.4.18
/> >/ AsyncAGI patch from http://moythreads.com/testasync2.diff
/> >/ Regards
/> >/ Jose

/
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Re: [asterisk-users] How run AsyncAGI commands in background

[Jose Arias]

Never mind, it was my mistake. I had some problems with my email client.
Regards
Jose
2009/6/7 Philipp Kempgen 

> Moises Silva schrieb:
> > On Sat, Jun 6, 2009 at 7:18 PM, Philipp
> > Kempgen wrote:
> >> Jose Arias schrieb:
>  >>> Hi,
> >>> Asterisk 1.4.18
> >>> AsyncAGI patch from //http://moythreads.com/testasync2.diff
> >>> <http://moythreads.com/testasync2.diff>//
> >>> Regards
> >>
> >> So what?
> >>
> > What do you mean with "so what?", if you have not been involved in the
> > conversation you would not understand.
> >
> > http://lists.digium.com/pipermail/asterisk-users/2009-June/232995.html
>
> Sorry for the noise. I didn't realize this was a discussion. The
> message didn't quote anything and the subject didn't start with
> "Re: " so it appeared as if Jose was just posting his version of
> Asterisk without any context.
>
>
>Philipp Kempgen
> --
> AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied  ->  http://www.amooma.de
> Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998
> Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de
> Videos of the AMOOCON VoIP conference 2009 ->  http://www.amoocon.de
> --
>
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[asterisk-users] How run AsyncAGI commands in background

[Jose Arias]

Hi,
Asterisk 1.4.18
AsyncAGI patch from //http://moythreads.com/testasync2.diff 
//

Regards
Jose

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[asterisk-users] How run AsyncAGI commands in background

[Jose Arias]

Hi all,
I have an external application commanding asterisk by AMI and AsyncAGI. I
also have a dialplan like this:

; AsyncAGI extensions
exten => _8.,1,Noop(entering in AGI loop at 8 ${EXTEN});
exten => _8.,n,AGI(agi:async);
exten => _8.,n,Hangup();

; Meetme extensions
exten => _1.,1,Noop(Conference ${EXTEN} ${CONTEXT});
exten => _1.,n,Set(MEETME_RECORDINGFILE=${EXTEN:}-${UNIQUEID:})
exten => _1.,n,MixMonitor(${MEETME_RECORDINGFILE}.wav)
exten => _1.,n,Meetme(${EXTEN},qdx);
exten => _1.,n,Hangup();

It works fine:

Incoming channels are sent to meetme by an external application, which
receives events by AMI and decides what meetme to use, making a redirect
action to it by AMI. Every channel falling in a meetme (dynamically
created) is recorded by the MixMonitor application.

But there's a little problem:

I don't need to record all calls but only those ones are switable of be
recorded (by some kind of external rules). As it's a waste of cpu and space
to record everything and then to discard almost all of them but some few
ones, I tought to use AsyncAGI to recording only some calls by sending an
AsyncAGI EXE MixMonitor command instead of the dial plan approach.

To do that, the external application, instead of making the redial to
meetme, it must make the redial to an AsyncAGI extension, then it must make
the AGI EXE MixMonitor action, and finally it must make the original
redirect to meetme.

But it doesn't work :-(

When the application reachs the third step (redial to meetme while the
channel is still into the AGI loop, after having sent it the AGI EXE
MixMonitor action) the MixMonitor AGI action is stopped automatically and
the recording ends.

Therefore, does anyone know how to manage that an AsyncAGI action to remain
running in background even if the channel is redirected out of AGI?

Thanks in advanced
Jose
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Re: [asterisk-users] How run AsyncAGI commands in background

[Jose Arias]

I'm sorry,
in my last email, where I said redial, I mean redirect.
Thanks
Jose
2009/6/5 Jose Arias 

> Hi all,
> I have an external application commanding asterisk by AMI and AsyncAGI. I
> also have a dialplan like this:
>
> ; AsyncAGI extensions
> exten => _8.,1,Noop(entering in AGI loop at 8 ${EXTEN});
> exten => _8.,n,AGI(agi:async);
> exten => _8.,n,Hangup();
>
> ; Meetme extensions
> exten => _1.,1,Noop(Conference ${EXTEN} ${CONTEXT});
> exten => _1.,n,Set(MEETME_RECORDINGFILE=${EXTEN:}-${UNIQUEID:})
> exten => _1.,n,MixMonitor(${MEETME_RECORDINGFILE}.wav)
> exten => _1.,n,Meetme(${EXTEN},qdx);
> exten => _1.,n,Hangup();
>
> It works fine:
>
> Incoming channels are sent to meetme by an external application, which
> receives events by AMI and decides what meetme to use, making a redirect
> action to it by AMI. Every channel falling in a meetme (dynamically
> created) is recorded by the MixMonitor application.
>
> But there's a little problem:
>
> I don't need to record all calls but only those ones are switable of be
> recorded (by some kind of external rules). As it's a waste of cpu and space
> to record everything and then to discard almost all of them but some few
> ones, I tought to use AsyncAGI to recording only some calls by sending an
> AsyncAGI EXE MixMonitor command instead of the dial plan approach.
>
> To do that, the external application, instead of making the redial to
> meetme, it must make the redial to an AsyncAGI extension, then it must make
> the AGI EXE MixMonitor action, and finally it must make the original
> redirect to meetme.
>
> But it doesn't work :-(
>
> When the application reachs the third step (redial to meetme while the
> channel is still into the AGI loop, after having sent it the AGI EXE
> MixMonitor action) the MixMonitor AGI action is stopped automatically and
> the recording ends.
>
> Therefore, does anyone know how to manage that an AsyncAGI action to remain
> running in background even if the channel is redirected out of AGI?
>
> Thanks in advanced
> Jose
>
>
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Re: [asterisk-users] async agi question

[Jose Arias]

Hi Moy,

thank you for your answer and this new testasync2.diff. It matches better my
mind about the subject.

I tried it out and it worked fine. The execution went right this time, as
you can see at bottom this mail.

 A last question: would you apply this fix to
asterisk-async-AGI-rev92324.patch for asterisk 1.6? do you think it would be
useful too? I think so.

Anyway, thanks a lot for your help. Tell me if you need to test anymore and,
as I said before, don't hesitate contact me for whatever help can need.

Best regards.
Jose


[Apr 17 09:40:39] DEBUG[1363]: manager.c:2108 process_message: Manager
received command 'AGI'
[Apr 17 09:40:39] -- Playing 'es/demo-congrats' (escape_digits=1)
(sample_offset 0)
[Apr 17 09:40:39] DEBUG[1402]: rtp.c:2753 ast_rtp_write: Ooh, format changed
from unknown to alaw
[Apr 17 09:40:39] DEBUG[1402]: rtp.c:2770 ast_rtp_write: Created smoother:
format: 8 ms: 20 len: 160
[Apr 17 09:40:39] DEBUG[1402]: channel.c:1793 ast_settimeout: Scheduling
timer at 160 sample intervals
[Apr 17 09:40:45] DEBUG[1363]: manager.c:2108 process_message: Manager
received command 'Redirect'
[Apr 17 09:40:45] DEBUG[1363]: channel.c:1378 ast_softhangup_nolock:
Soft-Hanging up channel 'SIP/502-08288f98'
[Apr 17 09:40:45] DEBUG[1402]: channel.c:1793 ast_settimeout: Scheduling
timer at 0 sample intervals
[Apr 17 09:40:45] DEBUG[1402]: res_agi.c:439 launch_asyncagi:
ast_check_hangup returned true after handling command on chan
SIP/502-08288f98 (softhangup = 0x2)
[Apr 17 09:40:45] DEBUG[1402]: res_agi.c:497 launch_asyncagi:
launch_asyncagi returned (0x4) for chan SIP/502-08288f98
[Apr 17 09:40:45] DEBUG[1402]: pbx.c:2427 __ast_pbx_run: Spawn extension
(sip_sercom,801,0) exited non-zero on 'SIP/502-08288f98'
[Apr 17 09:40:45]   == Spawn extension (sip_sercom, 801, 0) exited non-zero
on 'SIP/502-08288f98'
[Apr 17 09:40:45] DEBUG[1402]: pbx.c:1831 pbx_extension_helper: Launching
'NoOp'
[Apr 17 09:40:45] -- Executing [...@sip_sercom:1]
NoOp("SIP/502-08288f98", "entrada numeracion del 8 801") in new stack
[Apr 17 09:40:45] DEBUG[1402]: pbx.c:1831 pbx_extension_helper: Launching
'AGI'
[Apr 17 09:40:45] -- Executing [...@sip_sercom:2]
AGI("SIP/502-08288f98", "agi:async") in new stack



2009/4/16 Moises Silva moises.si...@gmail.com

> This can be tricky, I did not spend much time looking at different
> return codes. My reasoning was that returnstatus |=
> agi_handle_command() should preserve the command return status and
> return it to the caller (just as I saw was done for regular AGI), but
> also preserve the AGI status. However, taking a second look at the
> code and definitions of AST_PBX_KEEPALIVE, I think my fix is wrong,
> probably I just broke the "break asyncagi" AGI command and it seems
> the AST_PBX_KEEPALIVE code is not even used by the caller, just
> ignored. Can you help me to test something else?
>
> http://moythreads.com/testasync2.diff
>
> Let me know if it works,
>
> Moy
>
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Re: [asterisk-users] async agi question

[Jose Arias]

Hi Moy,

¡Great, it works! Thanks ever so much.

But, I don't understand anything. I mean:
The only difference I can see between this testasync.diff and your last
asterisk-1.4.18-async-agi.patch is at the lines:

returnstatus |= agi_handle_command(chan, &async_agi, cmd->cmd_buffer);
if ((returnstatus < 0) || (returnstatus == AST_PBX_KEEPALIVE)) {
 free_agi_cmd(cmd);
 ast_log(LOG_DEBUG, "agi_handle_command returned error or AST_PBX_KEEPALIVE
on chan %s (0x%X)\n", chan->name, returnstatus);
 break;
}
where you have inserted an ast_log output among other things, however, the
code isn't exiting by that point, as you can see:

[Apr 16 15:35:41] DEBUG[22621]: manager.c:2108 process_message: Manager
received command 'AGI'
[Apr 16 15:35:41] -- Playing 'es/demo-congrats' (escape_digits=1)
(sample_offset 0)
[Apr 16 15:35:41] DEBUG[22662]: rtp.c:2753 ast_rtp_write: Ooh, format
changed from unknown to alaw
[Apr 16 15:35:41] DEBUG[22662]: rtp.c:2770 ast_rtp_write: Created smoother:
format: 8 ms: 20 len: 160
[Apr 16 15:35:41] DEBUG[22662]: channel.c:1793 ast_settimeout: Scheduling
timer at 160 sample intervals
[Apr 16 15:35:50] DEBUG[22621]: manager.c:2108 process_message: Manager
received command 'Redirect'
[Apr 16 15:35:50] DEBUG[22621]: channel.c:1378 ast_softhangup_nolock:
Soft-Hanging up channel 'SIP/502-082901f0'
[Apr 16 15:35:50] DEBUG[22662]: channel.c:1793 ast_settimeout: Scheduling
timer at 0 sample intervals
[Apr 16 15:35:50] DEBUG[22662]: res_agi.c:430 launch_asyncagi:
ast_check_hangup returned true on chan SIP/502-082901f0 (0x2)
[Apr 16 15:35:50] DEBUG[22662]: res_agi.c:489 launch_asyncagi:
launch_asyncagi returned (0x4) for chan SIP/502-082901f0
[Apr 16 15:35:50] DEBUG[22662]: pbx.c:2427 __ast_pbx_run: Spawn extension
(sip_sercom,801,0) exited non-zero on 'SIP/502-082901f0'
[Apr 16 15:35:50]   == Spawn extension (sip_sercom, 801, 0) exited non-zero
on 'SIP/502-082901f0'
[Apr 16 15:35:50] DEBUG[22662]: pbx.c:1831 pbx_extension_helper: Launching
'NoOp'
[Apr 16 15:35:50] -- Executing [...@sip_sercom:1]
NoOp("SIP/502-082901f0", "entrada numeracion del 8 801") in new stack
[Apr 16 15:35:50] DEBUG[22662]: pbx.c:1831 pbx_extension_helper: Launching
'AGI'
[Apr 16 15:35:50] -- Executing [...@sip_sercom:2]
AGI("SIP/502-082901f0", "agi:async") in new stack

It's exiting by the if-else, above this last modification, already fixed in
asterisk-1.4.18-async-agi.patch. If I go back to
asterisk-1.4.18-async-agi.patch, then the execution doesn't exit overthere.
As I already said, I don't understand anything :-(

I would appreciate some kind of clarification about it if you can. I'm a
senior programmer, both c and c++, and I have developed asyncronous voice
applications for years, so don't hesitate to use all technical terms you
want.
Nevertheless, thanks again for all and count with me for any help you could
need

Regards
Jose

2009/4/15 Moises Silva 

> Ok, that makes more sense. Try this new patch and let me know how it
> goes, once you confirm it works I will post it in my blog with a
> better name.
>
> http://moythreads.com/testasync.diff
>
> Moy
>
> On Wed, Apr 15, 2009 at 11:52 AM,   wrote:
> > Hi Moy,
> > You are right. I failed applying the patch. In fact, I applied it but I
> didn't "make install" so I started a wrong asterisk. I apologize, it was my
> mistake. This time I made sure twice before getting the logs and this time
> the log message you said appears, but it doesn't work either as you can see:
> > I'm copying the whole log from the originate action to the hangup:
> >
> > =
> > [Apr 15 13:01:22] DEBUG[25752]: manager.c:2108 process_message: Manager
> received command 'originate'
> > [Apr 15 13:01:22] DEBUG[26933]: chan_sip.c:15874 sip_request_call: Asked
> to create a SIP channel with formats: 0x40 (slin)
> > [Apr 15 13:01:22] DEBUG[26933]: chan_sip.c:4508 sip_alloc: Allocating new
> SIP dialog for (No Call-ID) - INVITE (With RTP)
> > [Apr 15 13:01:22] DEBUG[26933]: chan_sip.c:2740 do_setnat: Setting NAT on
> RTP to Off
> > [Apr 15 13:01:22] DEBUG[26933]: chan_sip.c:2745 do_setnat: Setting NAT on
> VRTP to Off
> > [Apr 15 13:01:22] DEBUG[26933]: chan_sip.c:2998 sip_call: Outgoing Call
> for 501
> > [Apr 15 13:01:22] DEBUG[26933]: chan_sip.c:3013 sip_call: Our T38
> capability (0), joint T38 capability (0)
> > [Apr 15 13:01:22] DEBUG[26933]: chan_sip.c:6423 add_sdp: ** Our
> capability: 0x38000e (gsm|ulaw|alaw|h263|h263p|h264) Video flag: False
> > [Apr 15 13:01:22] DEBUG[26933]: chan_sip.c:6424 add_sdp: ** Our
> prefcodec: 0x40 (slin)
> > [Apr 15 13:01:22] DEBUG[26933]: chan_sip.c:6439 add_sdp: This call needs
> video offers!
> > [Apr 15 13:01:22] DEBUG[30207]: chan_sip.c:2215 __sip_semi_ack:
> (Provisional) Stopping retransmission (but retaining packet) on '
> 5c2607ce7cda26537726b6a4323a3...@10.0.5.20' Request 102: Found
> > [Apr 15 13:01:22] DEBUG[30207]: chan_sip.c:2215 __sip_semi_ack:
> (Provisional) Stopping retransmissio

Re: [asterisk-users] async agi question

[Jose Arias]

Hi,
I was asked for the patch and I sent it. Does anybody have any news about
this subject?
I'm willing to try a fix for 1.4 but I'd need any guidelines to do it.
Thanks in advanced
Jose
2009/4/2 Moises Silva 

> Async AGI was never released for Asterisk 1.4.X, so probably the patch
> you used has a bug or something, do you still have the patch around?
>
> Moy
>
> On Thu, Apr 2, 2009 at 5:44 AM,   wrote:
> > Hi Henrik,
> >
> > I would like to do the same thing you are doing here. I want to implement
> an external queue functionality so I need to stop a play file launched
> previously with an async agi command on caller's channel, sending the call
> to agent's extension.
> >
> > I'm redirecting caller's channel with REDIRECT while playing is taking
> place but I'm always getting a hang up on caller's channel.
> >
> > I'm using:
> >
> > asterisk-1.4.18
> > asterisk-addons-1.4.7
> > async agi patch 2007-12-11 10:34:12 (the one back-ported to 1.4)
> >
> > Both caller and agent are using 501 and 500 extensions and the async agi
> loop is waiting on 800, for example. The caller is dialing 800 where a play
> file is commanded through and async agi stream file command by the
> application.
> >
> > The relevant part of extensions.conf follows:
> >
> > exten => _5.,1,Noop(SIP call on 'sip_sercom' a ${EXTEN});
> > exten => _5.,n,Wait(1);
> > exten => _5.,n,Dial(SIP/${EXTEN},${TIMEOUTDIAL},Tto);
> > exten => _5.,n,Hangup();
> >
> > exten => _8.,1,Noop(every thing starting 8 ${EXTEN});
> > exten => _8.,n,AGI(agi:async);
> > exten => _8.,n,Hangup();
> >
> > And the redirect command the application is sending to is:
> >
> > Action: Redirect
> > Channel: SIP/501-081f0730
> > Exten: 500
> > Context: sip_sercom
> > Priority: 1
> >
> > Therefore, Henrik, could you show me your related dial plan and the
> redirect command you are sending? I wasn't able to see what I'm getting
> wrong.
> >
> > thanks in advanced
> > Jose M Arias
> >
> > --
> > This message was sent on behalf of cyr2...@gmail.com at
> openSubscriber.com
> >
> http://www.opensubscriber.com/message/asterisk-users@lists.digium.com/10933120.html
> >
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>
>
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Re: [asterisk-users] async agi question

[Jose Arias]

Yes, I have the patch around here. I think it's the one you said at
http://www.moythreads.com/wordpress/2007/12/24/asterisk-asynchronous-agi/

Due to the res_agi patch excedes the size limit for this mailing list,
(40Kb) I wasn't able to attach it on this post, so you can find it at
http://docs.google.com/Doc?id=ddb4rkts_0fd9z5qcr

Thanks
Jose


2009/4/2 Moises Silva

> Async AGI was never released for Asterisk 1.4.X, so probably the patch
> you used has a bug or something, do you still have the patch around?
>
> Moy
>
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[asterisk-users] The Redirect hangups the call while playing a file

[Jose Arias]

Hi,
I'm bringing this discussion here from
http://www.moythreads.com/wordpress/2007/12/24/asterisk-asynchronous-agi/
about how to manage stopping a playback on a extension previously launched
with AsyncAGI and redirecting the call to another exension.

If I make the Redirect without a playback, the Redirect works:
http://docs.google.com/Doc?id=ahfnfrcrh3rr_30f7fzq4hd

But if I make the Redirect while a playback, the Redirect fails
disconnecting the call:
http://docs.google.com/Doc?id=ahfnfrcrh3rr_31ghh84bkd

Regards
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