Re: [asterisk-users] Platform decision ...

[Mauro Sergio Ferreira Brasil]

Man... I need to be very frank with you... I don't know any more.

We started analysing what can be done to get Asterisk working on a way 
we want it to work, that is: totally dynamic dial plan generated by an 
external server (responsible for business logic and legacy interface), 
and retrieved through an new configuration driver (something like 
"res_config_legacy.c").
This point is clear to us now that is reachable without much effort.

We considered, at first, a infraestructure with a 
redirect-server/load-balancer (played by OpenSIPS) directing the voip 
calls to final Asterisk instances.
The problem is that after getting the first issue solved (about the 
driver acessing the legacy interface explained above), I started a 
research about Asterisk scalability and I didn't liked of what I found.

Consulting some friends of mine that work with Voip (but that 
unfortunatelly don't need the PBX features) the impression was worst.
One of them told me that on the only part of their infraestructure where 
Asterisk is used they want at all costs to remove it.

Making things short, I need to have sure that Asterisk can handle a 
considerable number of concurrent calls, or I need an indication of 
another PBX that is scalable to be placed on Asterisk's place and that 
can be changed to retrieve the dialplan (or what it uses on call 
routing) from another server.

Does anyone have any idea ?

Thanks and best regards,
Mauro.



C. Savinovich escreveu:
> It all depends what are you going to use Asterisk for.  Sounds like it is
> for conferencing.  Would you care to elaborate?
>
> CS
>
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mauro Sergio
> Ferreira Brasil
> Sent: Tuesday, August 18, 2009 10:23 PM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] Platform decision ...
>
> Hello there!
>
> During some research on Internet I found the following comparison on site
> Voip-Info (see, "http://www.voip-info.org/wiki/view/OpenPBX.org+FAQ";):
>
> The main points listed on Asterisk's "CONS" that concerned me were:
>
>* Conferencing on Asterisk depends on Zaptel hardware and/or kernel
> modules for timing;
>* Lack of built-in STUN support for SIP NAT traversal;
>* Asterisk doesn't use SpanDSP;
>* Use of no longer maintained Berkeley DB1 engine as its internal
> database;
>* Asterisk doesn't allow CSRC entries in RTP;
>* Asterisk doesn't have an universal jitterbuffer for use with any
> channel type;
>* Asterisk doesn't use POSIX realtime extensions (having dependency with
> Zaptel timing);
>
> We were considering Asterisk as the chosen platform, but after reading this
> I got a little worried.
> The comparison considers 1.4 old version of Asterisk.
>
> So, can someone give me an update on what have changed for this items
> considering new 1.6 version ?
> Maybe someone can point me a site with an updated comparison.
>
> As long as I could see by now SpanDSP is present on new version of Asterisk,
> so this item isn't a difference any more. Right ?
>
> Thanks and best regards,
>
>   

-- 
__At.,  
   
   _
 
*Technology and Quality on Information*
Mauro Sérgio Ferreira Brasil
Coordenador de Projetos e Analista de Sistemas
+ mauro.bra...@tqi.com.br <mailto:@tqi.com.br>
: www.tqi.com.br <http://www.tqi.com.br>
( + 55 (34)3291-1700
( + 55 (34)9971-2572


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Re: [asterisk-users] Platform decision ...

[Moises Silva]

>
> possibly very simple needs, but that doesn't make it very complete
> documentation.
>

Which nobody has stated it is. Just means for your needs (which you have
never stated nor pointed to particular missing chunks of documentation for
what you intended to do) was far from fulfilling your expectations.



> I'm not introducing FUD by stating my opinion about the lack of
> documentation, Moises. You're sounding incredibly defensive. Why?
>

I though I made it clear. Your comment sounded to me like FUD, since you
give an opinion without pointing to particular instances of your claims,
saying "wholly incomplete documentation" sounds extreme (which is typical
when trying to introduce FUD). So, is just you sounded like that to me, if
you are not trying to do so, then don't get it personal, isn't the point
here, for me anyways.

IRC answers from people hanging around is not documentation. It's what
> open-source developers like to think of, sometimes, as a 'complete'
> solution, but it doesn't even come close.

Again, never stated IRC was documentation, and was rather introduced as way
of saying, what you cannot find on the wiki, you can find it on IRC, because
is for sure there are things that are not on the wiki, even for more used
technologies like Asterisk.

documentation. Which is lacking. This is not FUD. This is not me saying
> "Don't touch FreeSWITCH."   This is me saying that, if you're looking
> for a product with good docs to make an easy transition from traditional
> PBX tech to new, or even an easy transition from Asterisk to something
> else, you will not find them with FS.

That is at least a bit better than your first post, re-read your first post
again, you never even mentioned transition from Asterisk nor whether you
wanted a PBX. Even when Asterisk is sometimes seen as a PBX, can be used as
a pstn gw or something else, so, as I said, you never mentioned what you
wanted to do nor when did you try (as projects evolve, your statements may
no longer be true). In short, the only piece of relevant information I was
able to take from your initial post was "documentation is crap and no useful
applications exist", for me, that sounds a lot like FUD. But hey, that's
just me, I'm sure others will learn something from your post, I just felt
the need to give my opinion as you did.

 I'm sorry you're offended by my opinions, but in your words, 'I defy

> you' to show me some comprehensive FreeSWITCH docs.  Heck, even SER has
> more comprehensive documentation, and that's saying a LOT.
>

No offense taken ;-), hope is the same for you. Again, our perspective of
what "comprehensive" documentation is differs, needs improvement for sure
though.

-- 
Moises Silva
Software Developer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. m...@sangoma.com
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Re: [asterisk-users] Platform decision ...

[SIP]

Moises Silva wrote:
> Hi there,
>
> I though to chime in here just to share my opinion for what is worth. 
> As a developer who enjoys playing with telephony in general I try to 
> remain as objective as possible when talking about one or the other, 
> and I felt that "N" from arcdiv was a bit unfair with FreeSWITCH docs.
>  
>
> Then you'd have to add the con:   cryptic, difficult to find, and
> wholly
> incomplete documentation.
>
> Don't get me wrong. FreeSwitch is a very nice back-end product. 
>
>
> It's hard to not get you wrong when, in my opinion, you start by 
> writing as facts what is barely your particular poor user experience 
> with it. Others, including me, have found what they need in FreeSWITCH 
> wiki just as I have found what I need about Asterisk docs in voip-info.

Sorry, Moises, but I've been to the FreeSWITCH Wiki. It's sparse at 
best. You may have found what you were looking for based on your 
possibly very simple needs, but that doesn't make it very complete 
documentation.

>
> far as ease of putting it into deployment goes, it's a nightmare from
> its complete dearth of anything related to coherent docs. It still
> feels
> very  nuts and bolts. Like being handed a Porsche Boxter engine,
> frame, and a wrench and being told to sort of 'figure out' how it all
> goes together. And even when you do, it will function screamingly
> well.
> But it won't have doors, windows, AC, or creature comforts that we've
> all come to expect.
>
>
> You mean comforts which you have come to expect. Again, my needs have 
> been so far fulfilled for conferencing and SIP/PSTN gateway uses. 
> Pointing to particular missing applications instead of making your own 
> analogy would be useful, otherwise you are not really being of much 
> help, and just introducing FUD.

I'm not introducing FUD by stating my opinion about the lack of 
documentation, Moises. You're sounding incredibly defensive. Why?

>
> Many users are confused because they try to do things the same way 
> they are used to with Asterisk and some concepts just don't fit or are 
> differently applied. From what I've seen the users that get annoyed 
> the most are those who keep trying to do things in the Asterisk-way 
> and get overwhelmed by the configuration differences, instead of 
> learning the FreeSWITCH-way to accomplish the same goals. Users just 
> get impatient because they're already familiar with something and this 
> new engine is not managed as the old one. The recent announcement of 
> FreePBX running over FreeSWITCH 
> (http://www.freepbx.org/news/2009-08-04/freepbx-v3-come-help-us-shape-the-future)
>  
> should help to close the gap in user configuration and ease of management.
>
> Of course, there is some truth in your statements. FreeSWITCH needs to 
> catch up with documentation, but I would defy anyone to say they've 
> come to hang around on IRC and did not get their question answered.

IRC answers from people hanging around is not documentation. It's what 
open-source developers like to think of, sometimes, as a 'complete' 
solution, but it doesn't even come close. My comment was about 
documentation. Which is lacking. This is not FUD. This is not me saying 
"Don't touch FreeSWITCH."   This is me saying that, if you're looking 
for a product with good docs to make an easy transition from traditional 
PBX tech to new, or even an easy transition from Asterisk to something 
else, you will not find them with FS.

I'm sure that's changing as time goes on, but it's not there yet, and 
the focus doesn't seem to be on ensuring it gets there. The focus seems 
to be on the band-aid of asking questions on fora and IRC to try and get 
an answer. That may work for some things, but for overall deployment, 
it's lacking.

I'm sorry you're offended by my opinions, but in your words, 'I defy 
you' to show me some comprehensive FreeSWITCH docs.  Heck, even SER has 
more comprehensive documentation, and that's saying a LOT.

N.

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Re: [asterisk-users] Platform decision ...

[Moises Silva]

Hi there,

I though to chime in here just to share my opinion for what is worth. As a
developer who enjoys playing with telephony in general I try to remain as
objective as possible when talking about one or the other, and I felt that
"N" from arcdiv was a bit unfair with FreeSWITCH docs.


> Then you'd have to add the con:   cryptic, difficult to find, and wholly
> incomplete documentation.
>
> Don't get me wrong. FreeSwitch is a very nice back-end product.


It's hard to not get you wrong when, in my opinion, you start by writing as
facts what is barely your particular poor user experience with it. Others,
including me, have found what they need in FreeSWITCH wiki just as I have
found what I need about Asterisk docs in voip-info.

far as ease of putting it into deployment goes, it's a nightmare from
> its complete dearth of anything related to coherent docs. It still feels
> very  nuts and bolts. Like being handed a Porsche Boxter engine,
> frame, and a wrench and being told to sort of 'figure out' how it all
> goes together. And even when you do, it will function screamingly well.
> But it won't have doors, windows, AC, or creature comforts that we've
> all come to expect.
>

You mean comforts which you have come to expect. Again, my needs have been
so far fulfilled for conferencing and SIP/PSTN gateway uses. Pointing to
particular missing applications instead of making your own analogy would be
useful, otherwise you are not really being of much help, and just
introducing FUD.

Many users are confused because they try to do things the same way they are
used to with Asterisk and some concepts just don't fit or are differently
applied. From what I've seen the users that get annoyed the most are those
who keep trying to do things in the Asterisk-way and get overwhelmed by the
configuration differences, instead of learning the FreeSWITCH-way to
accomplish the same goals. Users just get impatient because they're already
familiar with something and this new engine is not managed as the old one.
The recent announcement of FreePBX running over FreeSWITCH (
http://www.freepbx.org/news/2009-08-04/freepbx-v3-come-help-us-shape-the-future)
should help to close the gap in user configuration and ease of management.

Of course, there is some truth in your statements. FreeSWITCH needs to catch
up with documentation, but I would defy anyone to say they've come to hang
around on IRC and did not get their question answered.

Both Asterisk and FreeSWITCH share features, pro's, cont's and for some
people one is better than the other. I am looking forward for people to make
informed opinions about their experience with both engines.

-- 
Moises Silva
Software Developer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. m...@sangoma.com
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Re: [asterisk-users] Platform decision ...

[C. Savinovich]

It all depends what are you going to use Asterisk for.  Sounds like it is
for conferencing.  Would you care to elaborate?

CS


-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mauro Sergio
Ferreira Brasil
Sent: Tuesday, August 18, 2009 10:23 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Platform decision ...

Hello there!

During some research on Internet I found the following comparison on site
Voip-Info (see, "http://www.voip-info.org/wiki/view/OpenPBX.org+FAQ";):

The main points listed on Asterisk's "CONS" that concerned me were:

   * Conferencing on Asterisk depends on Zaptel hardware and/or kernel
modules for timing;
   * Lack of built-in STUN support for SIP NAT traversal;
   * Asterisk doesn't use SpanDSP;
   * Use of no longer maintained Berkeley DB1 engine as its internal
database;
   * Asterisk doesn't allow CSRC entries in RTP;
   * Asterisk doesn't have an universal jitterbuffer for use with any
channel type;
   * Asterisk doesn't use POSIX realtime extensions (having dependency with
Zaptel timing);

We were considering Asterisk as the chosen platform, but after reading this
I got a little worried.
The comparison considers 1.4 old version of Asterisk.

So, can someone give me an update on what have changed for this items
considering new 1.6 version ?
Maybe someone can point me a site with an updated comparison.

As long as I could see by now SpanDSP is present on new version of Asterisk,
so this item isn't a difference any more. Right ?

Thanks and best regards,

-- 
__At.,

   _
 
*Technology and Quality on Information*
Mauro Sérgio Ferreira Brasil
Coordenador de Projetos e Analista de Sistemas
+ mauro.bra...@tqi.com.br <mailto:@tqi.com.br>
: www.tqi.com.br <http://www.tqi.com.br> ( + 55 (34)3291-1700 ( + 55
(34)9971-2572


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Re: [asterisk-users] Platform decision ...

[SIP]

Steve Totaro wrote:
>
>
> On Tue, Aug 18, 2009 at 1:52 PM, Mauro Sergio Ferreira Brasil
> mailto:mauro.bra...@tqi.com.br>> wrote:
>
> Hello there!
>
> During some research on Internet I found the following comparison on
> site Voip-Info (see,
> "http://www.voip-info.org/wiki/view/OpenPBX.org+FAQ";):
>
> The main points listed on Asterisk's "CONS" that concerned me were:
>
>   * Conferencing on Asterisk depends on Zaptel hardware and/or kernel
> modules for timing;
>   * Lack of built-in STUN support for SIP NAT traversal;
>   * Asterisk doesn't use SpanDSP;
>   * Use of no longer maintained Berkeley DB1 engine as its internal
> database;
>   * Asterisk doesn't allow CSRC entries in RTP;
>   * Asterisk doesn't have an universal jitterbuffer for use with any
> channel type;
>   * Asterisk doesn't use POSIX realtime extensions (having dependency
> with Zaptel timing);
>
> We were considering Asterisk as the chosen platform, but after reading
> this I got a little worried.
> The comparison considers 1.4 old version of Asterisk.
>
> So, can someone give me an update on what have changed for this items
> considering new 1.6 version ?
> Maybe someone can point me a site with an updated comparison.
>
> As long as I could see by now SpanDSP is present on new version of
> Asterisk, so this item isn't a difference any more. Right ?
>
> Thanks and best regards,
>
> --
> __At.,
>   _
>
> *Technology and Quality on Information*
> Mauro Sérgio Ferreira Brasil
> Coordenador de Projetos e Analista de Sistemas
> + mauro.bra...@tqi.com.br 
> >
> : www.tqi.com.br  
> ( + 55 (34)3291-1700
> ( + 55 (34)9971-2572
>
>
> Don't forget to add FreeSwitch to your comparison chart too.
>

Then you'd have to add the con:   cryptic, difficult to find, and wholly
incomplete documentation.

Don't get me wrong. FreeSwitch is a very nice back-end product. But as
far as ease of putting it into deployment goes, it's a nightmare from
its complete dearth of anything related to coherent docs. It still feels
very  nuts and bolts. Like being handed a Porsche Boxter engine,
frame, and a wrench and being told to sort of 'figure out' how it all
goes together. And even when you do, it will function screamingly well.
But it won't have doors, windows, AC, or creature comforts that we've
all come to expect.

N.

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Re: [asterisk-users] Platform decision ...

[Steve Totaro]

On Tue, Aug 18, 2009 at 1:52 PM, Mauro Sergio Ferreira Brasil <
mauro.bra...@tqi.com.br> wrote:

> Hello there!
>
> During some research on Internet I found the following comparison on
> site Voip-Info (see, "http://www.voip-info.org/wiki/view/OpenPBX.org+FAQ
> "):
>
> The main points listed on Asterisk's "CONS" that concerned me were:
>
>   * Conferencing on Asterisk depends on Zaptel hardware and/or kernel
> modules for timing;
>   * Lack of built-in STUN support for SIP NAT traversal;
>   * Asterisk doesn't use SpanDSP;
>   * Use of no longer maintained Berkeley DB1 engine as its internal
> database;
>   * Asterisk doesn't allow CSRC entries in RTP;
>   * Asterisk doesn't have an universal jitterbuffer for use with any
> channel type;
>   * Asterisk doesn't use POSIX realtime extensions (having dependency
> with Zaptel timing);
>
> We were considering Asterisk as the chosen platform, but after reading
> this I got a little worried.
> The comparison considers 1.4 old version of Asterisk.
>
> So, can someone give me an update on what have changed for this items
> considering new 1.6 version ?
> Maybe someone can point me a site with an updated comparison.
>
> As long as I could see by now SpanDSP is present on new version of
> Asterisk, so this item isn't a difference any more. Right ?
>
> Thanks and best regards,
>
> --
> __At.,
>   _
>
> *Technology and Quality on Information*
> Mauro Sérgio Ferreira Brasil
> Coordenador de Projetos e Analista de Sistemas
> + mauro.bra...@tqi.com.br 
> : www.tqi.com.br 
> ( + 55 (34)3291-1700
> ( + 55 (34)9971-2572
>
>
Don't forget to add FreeSwitch to your comparison chart too.

-- 
Thanks,
Steve Totaro
+18887771888 (Toll Free)
+12409381212 (Cell)
+12024369784 (Skype)
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[asterisk-users] Platform decision ...

[Mauro Sergio Ferreira Brasil]

Hello there!

During some research on Internet I found the following comparison on 
site Voip-Info (see, "http://www.voip-info.org/wiki/view/OpenPBX.org+FAQ";):

The main points listed on Asterisk's "CONS" that concerned me were:

   * Conferencing on Asterisk depends on Zaptel hardware and/or kernel 
modules for timing;
   * Lack of built-in STUN support for SIP NAT traversal;
   * Asterisk doesn't use SpanDSP;
   * Use of no longer maintained Berkeley DB1 engine as its internal 
database;
   * Asterisk doesn't allow CSRC entries in RTP;
   * Asterisk doesn't have an universal jitterbuffer for use with any 
channel type;
   * Asterisk doesn't use POSIX realtime extensions (having dependency 
with Zaptel timing);

We were considering Asterisk as the chosen platform, but after reading 
this I got a little worried.
The comparison considers 1.4 old version of Asterisk.

So, can someone give me an update on what have changed for this items 
considering new 1.6 version ?
Maybe someone can point me a site with an updated comparison.

As long as I could see by now SpanDSP is present on new version of 
Asterisk, so this item isn't a difference any more. Right ?

Thanks and best regards,

-- 
__At.,  
   
   _
 
*Technology and Quality on Information*
Mauro Sérgio Ferreira Brasil
Coordenador de Projetos e Analista de Sistemas
+ mauro.bra...@tqi.com.br 
: www.tqi.com.br 
( + 55 (34)3291-1700
( + 55 (34)9971-2572


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