Re: [asterisk-users] G729

[Doug Lytle]

>>> And I'm not at all against Digium making money. But this whole 729 thing is 
>>> a mystery without any official feedback...

Not an official statement from Digium, but at the top of that thread, it says 
it all:


"david551
Sep '17

It still need copyright licensing.

You will need to find an open source implementation that is not based on code 
with commercial use restrictions. The allegedly open source implementation that 
was around whilst the patent was live actually had a void licence, because the 
purported licence was GPL, but it contained code identified by its upstream 
supplier as not for commercial use, invalidating the GPL.

If you want a royalty free G.729, you need to write it yourself, form the 
specification, not from the existing sample code."

Doug

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Re: [asterisk-users] G729

[Jonathan H]

Hmmm, again, this conversation has just faded out. I wondered why no
response from Digium?

So I found this discussion -
https://community.asterisk.org/t/why-does-g729-still-require-licensing/71920/8
- seems very clear that G729 is patent free, but still no response from
Digium.

Also, the link to the "definitely royalty free" version was removed.

Don't get me wrong, I'm sure there must be a valid reason.

And I'm not at all against Digium making money. But this whole 729 thing is
a mystery without any official feedback...



On Mon, 23 Jul 2018 at 05:37, Dmitry Melekhov  wrote:

> 20.07.2018 23:35, John Kiniston пишет:
>
>
> On Fri, Jul 20, 2018 at 11:41 AM Saint Michael  wrote:
>
>> ​The community would benefit if a non/licensed version of G729 would be
>>> included with Asterisk​, since the license expired. The current codec
>>> source code posted still requires licensing.
>>>
>> ​I am sure Digium would not prefer to ​
>>
>> ​acknowledge this, but the phenomenal growth of Asterisk is due to the
>> a​availability of a free G729 codec compiled and distributed free by Arkadi
>> Shislov.
>>
>> That'd be a surprise to me with the 325 G.729 licenses I have from Digium.
>
> I'm not a software pirate, I doubt that most telephony providers are
> either.
>
>
> Once again- patent is expired, g729 algorithm is now free.
> You spent you money to wrong place :-)
>
>
>
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[asterisk-users] ​G729 (Dmitry Melekhov)

[Saint Michael]

Maybe Digium should include a G729 codec inside Asterisk. What is keeping
them from doing it?

>
> Today's Topics:
>
>1. Re:
> ​​
> G729 (Dmitry Melekhov)
>
>
> --
>
> Message: 1
> Date: Mon, 23 Jul 2018 08:36:19 +0400
> From: Dmitry Melekhov 
> To: asterisk-users@lists.digium.com
> Subject: Re: [asterisk-users] G729
> Message-ID: <24c4a934-5210-9df4-f9e0-e5681b22b...@belkam.com>
> Content-Type: text/plain; charset="utf-8"; Format="flowed"
>
> 20.07.2018 23:35, John Kiniston пишет:
> >
> > On Fri, Jul 20, 2018 at 11:41 AM Saint Michael  > <mailto:vene...@gmail.com>> wrote:
> >
> > ​The community would benefit if a non/licensed version of G729
> > would be included with Asterisk​, since the license expired.
> > The current codec source code posted still requires licensing.
> >
> > ​I am sure Digium would not prefer to ​
> > ​acknowledge this, but the phenomenal growth of Asterisk is due to
> > the a​availability of a free G729 codec compiled and distributed
> > free by Arkadi Shislov.
> >
> > That'd be a surprise to me with the 325 G.729 licenses I have from
> Digium.
> >
> > I'm not a software pirate, I doubt that most telephony providers are
> > either.
>
> Once again- patent is expired, g729 algorithm is now free.
> You spent you money to wrong place :-)
>
>
>
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Re: [asterisk-users] G729

[Dmitry Melekhov]

20.07.2018 23:35, John Kiniston пишет:


On Fri, Jul 20, 2018 at 11:41 AM Saint Michael > wrote:


​The community would benefit if a non/licensed version of G729
would be included with Asterisk​, since the license expired.
The current codec source code posted still requires licensing.

​I am sure Digium would not prefer to ​
​acknowledge this, but the phenomenal growth of Asterisk is due to
the a​availability of a free G729 codec compiled and distributed
free by Arkadi Shislov.

That'd be a surprise to me with the 325 G.729 licenses I have from Digium.

I'm not a software pirate, I doubt that most telephony providers are 
either.


Once again- patent is expired, g729 algorithm is now free.
You spent you money to wrong place :-)



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Re: [asterisk-users] G729

[John Kiniston]

On Fri, Jul 20, 2018 at 11:41 AM Saint Michael  wrote:

> ​The community would benefit if a non/licensed version of G729 would be
>> included with Asterisk​, since the license expired. The current codec
>> source code posted still requires licensing.
>>
> ​I am sure Digium would not prefer to ​
>
> ​acknowledge this, but the phenomenal growth of Asterisk is due to the
> a​availability of a free G729 codec compiled and distributed free by Arkadi
> Shislov.
>
> That'd be a surprise to me with the 325 G.729 licenses I have from Digium.

I'm not a software pirate, I doubt that most telephony providers are either.


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[asterisk-users] G729

[Saint Michael]

>
> ​The community would benefit if a non/licensed version of G729 would be
> included with Asterisk​, since the license expired. The current codec
> source code posted still requires licensing.
>
​I am sure Digium would not prefer to ​

​acknowledge this, but the phenomenal growth of Asterisk is due to the
a​availability of a free G729 codec compiled and distributed free by Arkadi
Shislov.
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Re: [asterisk-users] G729 Licensing Revisited - I'm Sorry!

[Jayson Devor]

Hello Everyone,

Thank your for your response. There are two critical questions I would
like clarified
kindly:

1) We do not perform any transcoding whatsoever. All recordings, and
voice mail are in G729,
and allow=g729 for all peers and in sip.conf. Is there anything else
we need to perform g729 passthrough. More importantly are we still
liable? Given that most vendors support G729, why do some still
require the need to transcode?

2) If we decide that we require to purchase licenses, can we purchase
23 licenses and continue to use the open source version?

 Darryl Said
 The real question is: is there really any choice other than Digium for the 
 licence? Due to
 the dual licensing of the asterisk code, even if you could license the codec 
 elsewhere, you  might be violating Digium's OSS license when you don't but 
 their commercial asterisk
 license.

This only applies to the commercial versions of the codec right? We
are still ok in respect to Digium's OSS license with the open source
should we decide to continue using that version?

I really appreciate some light on this gentlemen.

J.

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Re: [asterisk-users] G729 Licensing Revisited - I'm Sorry!

[Tahir Almas]

 1) We do not perform any transcoding whatsoever. All recordings, and
 voice mail are in G729,
 and allow=g729 for all peers and in sip.conf. Is there anything else
 we need to perform g729 passthrough. More importantly are we still
 liable? Given that most vendors support G729, why do some still
 require the need to transcode?


 As earlier referred following quote from their site

DISCLAIMER: You might have to pay royalty fees to the G.729/723.1
patent holders for using their algorithm

You have to pay royalty fee for using their algorithm and it does not
matter whether you are trans-coding or not however there is no restriction
to pay their royalty fee under testing / evaluation environment.


2) If we decide that we require to purchase licenses, can we purchase
 23 licenses and continue to use the open source version?


I do't think there is any restriction to use open source version when you
paid their roylity fee

 Darryl Said
  The real question is: is there really any choice other than Digium for
 the licence? Due to
  the dual licensing of the asterisk code, even if you could license the
 codec elsewhere, you  might be violating Digium's OSS license when you
 don't but their commercial asterisk
  license.

 This only applies to the commercial versions of the codec right? We
 are still ok in respect to Digium's OSS license with the open source
 should we decide to continue using that version?


Yes it is .   I do't think there is confilit between GPL license and g.729
patent fee

Regards
*Tahir Almas*

Managing Partner
ICT Innovations
http://www.ictinnovations.com
Leveraging open source in ICT
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Re: [asterisk-users] G729 Licensing Revisited - I'm Sorry!

[Johann Steinwendtner]

On 2014-02-28 14:04, Tahir Almas wrote:

1) We do not perform any transcoding whatsoever. All recordings, and
voice mail are in G729,
and allow=g729 for all peers and in sip.conf. Is there anything else
we need to perform g729 passthrough. More importantly are we still
liable? Given that most vendors support G729, why do some still
require the need to transcode?


  As earlier referred following quote from their site

DISCLAIMER: You might have to pay royalty fees to the G.729/723.1
patent holders for using their algorithm

You have to pay royalty fee for using their algorithm and it does not matter 
whether you are trans-coding or not however there is no restriction to pay 
their royalty fee under testing / evaluation
environment.


Hmm, wouldn't that mean, that every single ISP needs to pay the fee for passing 
G.729 data through their network ? They really do not transcode.
If you do not transcode, you do not use their algorithm. This is my opinion, 
but I 'm not a layer.

Regards

Hans


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Re: [asterisk-users] G729 Licensing Revisited - I'm Sorry!

[A J Stiles]

On Friday 28 Feb 2014, Tahir Almas wrote:
  As earlier referred following quote from their site
 
 DISCLAIMER: You might have to pay royalty fees to the G.729/723.1
 patent holders for using their algorithm
 
 You have to pay royalty fee for using their algorithm and it does not
 matter whether you are trans-coding or not however there is no restriction
 to pay their royalty fee under testing / evaluation environment.

There is also no requirement to pay any royalty fee in jurisdictions where 
software is beyond the scope of patentability  (i.e., most of the world except 
the USA).  

-- 
AJS

Answers come *after* questions.

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Re: [asterisk-users] G729 Licensing Revisited - I'm Sorry!

[Paul Belanger]

On Thu, Feb 27, 2014 at 10:55 PM, Darryl Moore dar...@moores.ca wrote:

 On Feb 27, 2014 10:02 PM, Paul Belanger paul.belan...@polybeacon.com
 wrote:

 
 No such thing as 'free open source g729 license', if you actually read the
 site:


 There is regarding the copyright on the code. The fact it is also patent
 encumbered is a different issue.

 DISCLAIMER: You might have to pay royalty fees to the G.729/723.1
 patent holders for using their algorithm.

 So, basically you are illegal using them if you didn't pay for them.


 Not true. He said it was a lab setup. It is totally legit to use patented
 processes in an evaluation/lab environment.

Correct, I didn't mention this, since I was assuming OP was talking
about getting it into production.  Should have been more clear.

  3) Is there a performance/stability/security gain when using the
  commercial vs. open source version or vice versa.
 
 See above about about open source license.

 Your comment about open source is irrelevant to performance, stability, and
 security. WRT these criteria, I would be surprised if there is much of a
 difference. The free software isn't locked to a mother board, so that might
 count towards performance by some measures.

 Now having said that. I agree once you leave the lab environment and decide
 you need g.729, you will unfortunatly need a licence to keep using it.

 The real question is: is there really any choice other than Digium for the
 licence? Due to the dual licensing of the asterisk code, even if you could
 license the codec elsewhere, you might be violating Digium's OSS license
 when you don't but their commercial asterisk license.

 Cheers,
 Darryl


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Re: [asterisk-users] G729 Licensing Revisited - I'm Sorry!

[Jayson Devor]

On 2/28/14, Johann Steinwendtner steinwendt...@gmx.net wrote:
 On 2014-02-28 14:04, Tahir Almas wrote:
 1) We do not perform any transcoding whatsoever. All recordings, and
 voice mail are in G729,
 and allow=g729 for all peers and in sip.conf. Is there anything else
 we need to perform g729 passthrough. More importantly are we still
 liable? Given that most vendors support G729, why do some still
 require the need to transcode?


   As earlier referred following quote from their site

 DISCLAIMER: You might have to pay royalty fees to the G.729/723.1
 patent holders for using their algorithm

 You have to pay royalty fee for using their algorithm and it does not
 matter whether you are trans-coding or not however there is no restriction
 to pay their royalty fee under testing / evaluation
 environment.

 Hmm, wouldn't that mean, that every single ISP needs to pay the fee for
 passing G.729 data through their network ? They really do not transcode.
 If you do not transcode, you do not use their algorithm. This is my opinion,
 but I 'm not a layer.

 Regards

 Hans


 --

This is what I would expect? Not a laywer either however can someone please
share some light on the legality of straight passing through traffic
using the g729
codec without any transcoding. Is a license require for such cases?

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Re: [asterisk-users] G729 Licensing Revisited - I'm Sorry!

[Jayson Devor]

 Correct, I didn't mention this, since I was assuming OP was talking
 about getting it into production.  Should have been more clear.


Sorry I should clarify. We were incubated for the testing period
however now will be depolying
for commercial use. That being said, we do feel the need to
contribute, and staying legit at
the same time. That being said, will purchasing 23 licenses (one for
each channel that we use), and continue to use the open source g729
sorftware keep us legal? Or do we have to use the commercial software
to keep our licenses valid? Sorry, i'm a type of idiot savant
(probably more towards the idiot side), of need of some concrete
answers so I can sleep at night :).

J

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Re: [asterisk-users] G729 Licensing Revisited - I'm Sorry!

[Eric Wieling]

Asterisk transcodes at many other points.  Inband ringing, audio mixing for 
conferences, beep tones.  It is naive to think you can passthrough g729 and 
never transcode without spending significant amounts of time tracking down each 
instance Asterisk would have to transcode.

Over the years I've often seen people claiming I'm only using it for 
research/educational use so I don't need a license.I don't recall a single 
one of those people who could point to an actual law or case precedent to back 
up their claim, at least for the United States.  Personally I think it is 
simply wishful thinking.   

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jayson Devor
Sent: Friday, February 28, 2014 4:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] G729 Licensing Revisited - I'm Sorry!

On 2/28/14, Johann Steinwendtner steinwendt...@gmx.net wrote:
 On 2014-02-28 14:04, Tahir Almas wrote:
 1) We do not perform any transcoding whatsoever. All recordings, and
 voice mail are in G729,
 and allow=g729 for all peers and in sip.conf. Is there anything else
 we need to perform g729 passthrough. More importantly are we still
 liable? Given that most vendors support G729, why do some still
 require the need to transcode?


   As earlier referred following quote from their site

 DISCLAIMER: You might have to pay royalty fees to the G.729/723.1 
 patent holders for using their algorithm

 You have to pay royalty fee for using their algorithm and it does not 
 matter whether you are trans-coding or not however there is no 
 restriction to pay their royalty fee under testing / evaluation 
 environment.

 Hmm, wouldn't that mean, that every single ISP needs to pay the fee 
 for passing G.729 data through their network ? They really do not transcode.
 If you do not transcode, you do not use their algorithm. This is my 
 opinion, but I 'm not a layer.

 Regards

 Hans


 --

This is what I would expect? Not a laywer either however can someone please 
share some light on the legality of straight passing through traffic using the 
g729 codec without any transcoding. Is a license require for such cases?

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Re: [asterisk-users] G729 Licensing Revisited - I'm Sorry!

[Eric Wieling]

Why would you use anything other than Digium's fully licensed and fully 
compatable with Asterisk modules?

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jayson Devor
Sent: Friday, February 28, 2014 4:04 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] G729 Licensing Revisited - I'm Sorry!

 Correct, I didn't mention this, since I was assuming OP was talking 
 about getting it into production.  Should have been more clear.


Sorry I should clarify. We were incubated for the testing period however now 
will be depolying for commercial use. That being said, we do feel the need to 
contribute, and staying legit at the same time. That being said, will 
purchasing 23 licenses (one for each channel that we use), and continue to use 
the open source g729 sorftware keep us legal? Or do we have to use the 
commercial software to keep our licenses valid? Sorry, i'm a type of idiot 
savant (probably more towards the idiot side), of need of some concrete answers 
so I can sleep at night :).

J

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Re: [asterisk-users] G729 Licensing Revisited - I'm Sorry!

[Chris Bagnall]

On 28/2/14 9:04 pm, Jayson Devor wrote:

That being said, will purchasing 23 licenses (one for
each channel that we use), and continue to use the open source g729
sorftware keep us legal?


I know at least half a dozen people who do this so that they can more 
effectively balance their licence commitment over a number of services, 
rather than locking licences down to MAC addresses of specific NICs in 
specific servers. But I'm based in the EU where (as others have said) 
patentability laws are quite different.


If you're worried about whether it's legal in your jurisdiction, you 
really should speak to a qualified legal professional to allay your 
concerns. This list has such an international audience that what's 
perfectly acceptable in one jurisdiction might land you in hot water in 
another.


Kind regards,

Chris
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Re: [asterisk-users] G729 Licensing Revisited - I'm Sorry!

[Eric Wieling]

For 23 channels I recommend a hardware transcoding card.   

We use http://www.sangoma.com/products/d100-30-400-sessions/   I think Digium 
also has a transcoding card also.

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chris Bagnall
Sent: Friday, February 28, 2014 4:28 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] G729 Licensing Revisited - I'm Sorry!

On 28/2/14 9:04 pm, Jayson Devor wrote:
 That being said, will purchasing 23 licenses (one for each channel 
 that we use), and continue to use the open source g729 sorftware keep 
 us legal?

I know at least half a dozen people who do this so that they can more 
effectively balance their licence commitment over a number of services, rather 
than locking licences down to MAC addresses of specific NICs in specific 
servers. But I'm based in the EU where (as others have said) patentability laws 
are quite different.

If you're worried about whether it's legal in your jurisdiction, you really 
should speak to a qualified legal professional to allay your concerns. This 
list has such an international audience that what's perfectly acceptable in one 
jurisdiction might land you in hot water in another.

Kind regards,

Chris
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[asterisk-users] G729 Licensing Revisited - I'm Sorry!

[Jayson Devor]

Hello Everyone,

We are looking to transition our 23 channels from testing/lab into
production. During testing we used the free open source g729 license
using the instructions found here:

http://blog.manhag.org/2010/05/installing-the-free-g729-codec-for-asterisk/

A little more about our setup. All recordings have been converted to
G729, all voicemail messages are also in G729, finally allow=g729,
disallow=all is specified in our config.

Questions:

1) Is there anything we overlooked in our attempt to implement g729
passthough, and stop all transcoding efforts?
2) do we still need to purchase 23 G729 licenses? If so, is asterisk
10$ license recognized by the patent holders (ie, is Digium authorized
to sell the license on behalf of the patent holders)?
3) Is there a performance/stability/security gain when using the
commercial vs. open source version or vice versa.

I was reluctant to bring this topic up yet again , and yes I did
google around and read the different material on the subject however,
I am still in need of some definitive answers.

Kind Regards,

Jayson Devor.

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Re: [asterisk-users] G729 Licensing Revisited - I'm Sorry!

[Paul Belanger]

On Thu, Feb 27, 2014 at 5:24 PM, Jayson Devor jayson.de...@gmail.com wrote:
 Hello Everyone,

 We are looking to transition our 23 channels from testing/lab into
 production. During testing we used the free open source g729 license
 using the instructions found here:

 http://blog.manhag.org/2010/05/installing-the-free-g729-codec-for-asterisk/

No such thing as 'free open source g729 license', if you actually read the site:

DISCLAIMER: You might have to pay royalty fees to the G.729/723.1
patent holders for using their algorithm.

So, basically you are illegal using them if you didn't pay for them.

 A little more about our setup. All recordings have been converted to
 G729, all voicemail messages are also in G729, finally allow=g729,
 disallow=all is specified in our config.

 Questions:

 1) Is there anything we overlooked in our attempt to implement g729
 passthough, and stop all transcoding efforts?
 2) do we still need to purchase 23 G729 licenses? If so, is asterisk
 10$ license recognized by the patent holders (ie, is Digium authorized
 to sell the license on behalf of the patent holders)?
Yes, getting a license from digium should be sufficient to cover your
usage.  Plus you'll be supporting the project.

 3) Is there a performance/stability/security gain when using the
 commercial vs. open source version or vice versa.

See above about about open source license.

 I was reluctant to bring this topic up yet again , and yes I did
 google around and read the different material on the subject however,
 I am still in need of some definitive answers.


-- 
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Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger

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Re: [asterisk-users] G729 Licensing Revisited - I'm Sorry!

[Darryl Moore]

On Feb 27, 2014 10:02 PM, Paul Belanger paul.belan...@polybeacon.com
wrote:

 
 No such thing as 'free open source g729 license', if you actually read
the site:


There is regarding the copyright on the code. The fact it is also patent
encumbered is a different issue.

 DISCLAIMER: You might have to pay royalty fees to the G.729/723.1
 patent holders for using their algorithm.

 So, basically you are illegal using them if you didn't pay for them.


Not true. He said it was a lab setup. It is totally legit to use patented
processes in an evaluation/lab environment.

  3) Is there a performance/stability/security gain when using the
  commercial vs. open source version or vice versa.
 
 See above about about open source license.

Your comment about open source is irrelevant to performance, stability, and
security. WRT these criteria, I would be surprised if there is much of a
difference. The free software isn't locked to a mother board, so that might
count towards performance by some measures.

Now having said that. I agree once you leave the lab environment and decide
you need g.729, you will unfortunatly need a licence to keep using it.

The real question is: is there really any choice other than Digium for the
licence? Due to the dual licensing of the asterisk code, even if you could
license the codec elsewhere, you might be violating Digium's OSS license
when you don't but their commercial asterisk license.

Cheers,
Darryl
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[asterisk-users] G729 - what happens if licences used up?

[Tony Mountifield]

I haven't been able to find the answer online, and am not currently
able to conduct an experiment to find the answer...

I understand that in a SIP call where G729 has been negotiated as the
preferred codec, a G.729 licence is not consumed until there is a need
to perform transcoding, e.g. play a non-g729 sound, or do voicemail,
or enter a Meetme, etc.

What happens when a SIP call in progress needs a G.729 licence and
they are all in use already? Does the call fail, or go silent, or do a
re-INVITE to negotiate another codec?

I'm interested in what happens on Asterisk 1.2 (for a legacy system),
and also whether it is any different on later versions.

Thanks,
Tony
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Play: t...@mountifield.org - http://tony.mountifield.org

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Re: [asterisk-users] G729 - what happens if licences used up?

[Paul Belanger]

On Thu, Feb 20, 2014 at 10:40 AM, Tony Mountifield t...@softins.co.uk wrote:
 I haven't been able to find the answer online, and am not currently
 able to conduct an experiment to find the answer...

 I understand that in a SIP call where G729 has been negotiated as the
 preferred codec, a G.729 licence is not consumed until there is a need
 to perform transcoding, e.g. play a non-g729 sound, or do voicemail,
 or enter a Meetme, etc.

 What happens when a SIP call in progress needs a G.729 licence and
 they are all in use already? Does the call fail, or go silent, or do a
 re-INVITE to negotiate another codec?

 I'm interested in what happens on Asterisk 1.2 (for a legacy system),
 and also whether it is any different on later versions.

The question depends if you are offering up other codecs or not.  If
you only using g729, the call will fail to establish because lack of
codecs.  If you offer a both g729 and ulaw, then ulaw will be used.

-- 
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Re: [asterisk-users] G729 - what happens if licences used up?

[Eric Wieling]

In my experience when you run out of g729 licenses additional calls will fail.  
 Simple as that.   Make sure you run out of licenses.

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tony Mountifield
Sent: Thursday, February 20, 2014 10:40 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] G729 - what happens if licences used up?

I haven't been able to find the answer online, and am not currently able to 
conduct an experiment to find the answer...

I understand that in a SIP call where G729 has been negotiated as the preferred 
codec, a G.729 licence is not consumed until there is a need to perform 
transcoding, e.g. play a non-g729 sound, or do voicemail, or enter a Meetme, 
etc.

What happens when a SIP call in progress needs a G.729 licence and they are all 
in use already? Does the call fail, or go silent, or do a re-INVITE to 
negotiate another codec?

I'm interested in what happens on Asterisk 1.2 (for a legacy system), and also 
whether it is any different on later versions.

Thanks,
Tony
--
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Work: t...@softins.co.uk - http://www.softins.co.uk
Play: t...@mountifield.org - http://tony.mountifield.org

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Re: [asterisk-users] G729 - what happens if licences used up?

[Tony Mountifield]

In article CALLKq0RpimD05jz=osbgjydx-41uebohxmft_skwfjt51ko...@mail.gmail.com,
Paul Belanger paul.belan...@polybeacon.com wrote:
 On Thu, Feb 20, 2014 at 10:40 AM, Tony Mountifield t...@softins.co.uk wrote:
  I haven't been able to find the answer online, and am not currently
  able to conduct an experiment to find the answer...
 
  I understand that in a SIP call where G729 has been negotiated as the
  preferred codec, a G.729 licence is not consumed until there is a need
  to perform transcoding, e.g. play a non-g729 sound, or do voicemail,
  or enter a Meetme, etc.
 
  What happens when a SIP call in progress needs a G.729 licence and
  they are all in use already? Does the call fail, or go silent, or do a
  re-INVITE to negotiate another codec?
 
  I'm interested in what happens on Asterisk 1.2 (for a legacy system),
  and also whether it is any different on later versions.
 
 The question depends if you are offering up other codecs or not.  If
 you only using g729, the call will fail to establish because lack of
 codecs.  If you offer a both g729 and ulaw, then ulaw will be used.

The codecs offered by each end would be g729, alaw and ulaw.

I guess my point is that the licence is NOT required to negotiate codecs
and establish the call, e.g. if g.729 sounds are installed and calls are
pass-through, then no transcoding is required.

So the call will negotiate g729 and get established, and then if later
the dialplan calls something that requires transcoding, the licence is
requested at that time. What happens if there is not one available?
Can/will it do a re-INVITE to change codec, or does the call fail,
or does it continue but go silent?

Cheers
Tony
-- 
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Work: t...@softins.co.uk - http://www.softins.co.uk
Play: t...@mountifield.org - http://tony.mountifield.org

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Re: [asterisk-users] G729 - what happens if licences used up?

[Gareth Blades]

On 20/02/14 17:16, Paul Belanger wrote:

On Thu, Feb 20, 2014 at 10:40 AM, Tony Mountifieldt...@softins.co.uk  wrote:

I haven't been able to find the answer online, and am not currently
able to conduct an experiment to find the answer...

I understand that in a SIP call where G729 has been negotiated as the
preferred codec, a G.729 licence is not consumed until there is a need
to perform transcoding, e.g. play a non-g729 sound, or do voicemail,
or enter a Meetme, etc.

What happens when a SIP call in progress needs a G.729 licence and
they are all in use already? Does the call fail, or go silent, or do a
re-INVITE to negotiate another codec?

I'm interested in what happens on Asterisk 1.2 (for a legacy system),
and also whether it is any different on later versions.


The question depends if you are offering up other codecs or not.  If
you only using g729, the call will fail to establish because lack of
codecs.  If you offer a both g729 and ulaw, then ulaw will be used.

That would only apply for new calls. Even new calls would still 
typically accept g729 even if there are no licenses remaining as there 
might not be transcoding required.
What I would expect to happen if there were no licenses is for you to 
see an error on the console (possibly repeated multiple times) and for 
there to be no audio. This is certainly what happens if you have a g729 
call with no license and then try to play a sound file which does not 
have a native g729 format.


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Re: [asterisk-users] G729 CPU Utilization

[james . zhu]

hello:
it really depends on number of calls transcded. it the number is less 20 
calls, the CPU should be ok.
but it is very sure that software based take take much more than 
hardware based transcoder.

? 2013-9-9 15:53, Gopalakrishnan N ??:

Hi,

How much CPU utilization will it take when I use G729 transcoding via 
hardware based transcoder.


Regards


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Best regards!
Connect with Sangoma in APAC
www.hiastar.com

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[asterisk-users] G729 CPU Utilization

[Gopalakrishnan N]

Hi,

How much CPU utilization will it take when I use G729 transcoding via
hardware based transcoder.

Regards
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Re: [asterisk-users] G729 Passthrough How To

[Nick Cameo]

You ok sir? Are you going to make it?

N.

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Re: [asterisk-users] G729 Passthrough How To

[John Rodgers]

Sent from my Verizon Wireless 4G LTE DROID

Eric Wieling ewiel...@nyigc.com wrote:

If you want to do pass-thu you need to make sure Asterisk NEVER EVER needs to 
transcode.   This means all calls must use only g729, sound files must be in 
g729 format and no early audio, inband ringing or anything else which might 
cause Asterisk to require a temp transcoding path.

In my experience it never works right.The most you should expect to be 
able to do is reduce the need for transcoding by doing the above steps.

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Khamis
Sent: Wednesday, August 14, 2013 10:20 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] G729 Passthrough How To

Anyone? :)

N.

On 8/13/13, Nick Khamis sym...@gmail.com wrote:
 Hello Everyone,

 We are currently experiencing some higher load on our servers, and 
 since signaling comes into our servers on G729, we would like to 
 implement G729 pass-through. A few questions arise, do we need to 
 convert all the recording to the codec, and what about voicemail?

 We are also using A2Billing (hope I am not violating any thread 
 rules), and would like to convert all that recording to G729 as well.

 Any help is greatly appreciated.

 Kind Regards,

 Nick from Toronto.


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Re: [asterisk-users] G729 Passthrough How To

[John Rodgers]

Sent from my Verizon Wireless 4G LTE DROID

Eric Wieling ewiel...@nyigc.com wrote:

If you want to do pass-thu you need to make sure Asterisk NEVER EVER needs to 
transcode.   This means all calls must use only g729, sound files must be in 
g729 format and no early audio, inband ringing or anything else which might 
cause Asterisk to require a temp transcoding path.

In my experience it never works right.The most you should expect to be 
able to do is reduce the need for transcoding by doing the above steps.

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Khamis
Sent: Wednesday, August 14, 2013 10:20 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] G729 Passthrough How To

Anyone? :)

N.

On 8/13/13, Nick Khamis sym...@gmail.com wrote:
 Hello Everyone,

 We are currently experiencing some higher load on our servers, and 
 since signaling comes into our servers on G729, we would like to 
 implement G729 pass-through. A few questions arise, do we need to 
 convert all the recording to the codec, and what about voicemail?

 We are also using A2Billing (hope I am not violating any thread 
 rules), and would like to convert all that recording to G729 as well.

 Any help is greatly appreciated.

 Kind Regards,

 Nick from Toronto.


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Re: [asterisk-users] G729 Passthrough How To

[Nick Khamis]

Anyone? :)

N.

On 8/13/13, Nick Khamis sym...@gmail.com wrote:
 Hello Everyone,

 We are currently experiencing some higher load on our servers, and
 since signaling comes into our servers on G729, we would like to
 implement G729 pass-through. A few questions arise, do we need to
 convert all the recording to the codec, and what about voicemail?

 We are also using A2Billing (hope I am not violating any thread
 rules), and would like to convert all that recording to G729 as well.

 Any help is greatly appreciated.

 Kind Regards,

 Nick from Toronto.


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Re: [asterisk-users] G729 Passthrough How To

[Eric Wieling]

If you want to do pass-thu you need to make sure Asterisk NEVER EVER needs to 
transcode.   This means all calls must use only g729, sound files must be in 
g729 format and no early audio, inband ringing or anything else which might 
cause Asterisk to require a temp transcoding path.

In my experience it never works right.The most you should expect to be able 
to do is reduce the need for transcoding by doing the above steps.

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Khamis
Sent: Wednesday, August 14, 2013 10:20 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] G729 Passthrough How To

Anyone? :)

N.

On 8/13/13, Nick Khamis sym...@gmail.com wrote:
 Hello Everyone,

 We are currently experiencing some higher load on our servers, and 
 since signaling comes into our servers on G729, we would like to 
 implement G729 pass-through. A few questions arise, do we need to 
 convert all the recording to the codec, and what about voicemail?

 We are also using A2Billing (hope I am not violating any thread 
 rules), and would like to convert all that recording to G729 as well.

 Any help is greatly appreciated.

 Kind Regards,

 Nick from Toronto.


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Re: [asterisk-users] G729 Passthrough How To

[Nick Khamis]

Hey!!! Eric thank you so much for your response. Could you guys please
direct us in achieving as much as possible. For example:
* What linux command can we use to convert all recording to G729
* Which files do we need to convert and there locations
* For *testing* how do we make sure Asterisk NEVER EVER transcodes.

Do we still need the G729 codec installed on the asterisk machine if
we manage to implement pass-through that would suffice our needs.

Kind Regards,

Nick.

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Re: [asterisk-users] G729 Passthrough How To

[Nick Khamis]

I forgot to mention that all our equipment (phones etc..) are using
G729, and this is for internal use over the net. The problem,
concurrent calls, and bad bandwidth at some locations...

N.

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Re: [asterisk-users] G729 Passthrough How To

[Asghar Mohammad]

As my understanding Asterisk always pass-thu g729 if both ends have this
codec.
But if you answer the call or play some audio before dialing to end point
then asterisk stay between both legs.
In case of VM. you should install g729 if your prompts are in g729 format.
As a2billing play voice prompts you cannot pass-thu transparently.
I think the load on you server is not for transcoding  but PHP scripts.
I was in this situation and reduce the upto 80% by removing A2B.


On Wed, Aug 14, 2013 at 4:44 PM, Nick Khamis sym...@gmail.com wrote:

 Hey!!! Eric thank you so much for your response. Could you guys please
 direct us in achieving as much as possible. For example:
 * What linux command can we use to convert all recording to G729
 * Which files do we need to convert and there locations
 * For *testing* how do we make sure Asterisk NEVER EVER transcodes.

 Do we still need the G729 codec installed on the asterisk machine if
 we manage to implement pass-through that would suffice our needs.

 Kind Regards,

 Nick.

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Re: [asterisk-users] G729 Passthrough How To

[Nick Khamis]

Hello Ashgar,

Thank you so much for your response. As removing A2B is not an option
we would first like to begin by converting all audio files (Asterisk,
VM, A2B prompts etc...) to G729 to minimize unneeded trascoding. Linux
commands and the list of recording would be a great help. Sorry, not
new to VoIP but new to Asterisk :).

N.

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Re: [asterisk-users] G729 Passthrough How To

[Eric Wieling]

file convert in the Asterisk CLI, IF you have the g729 codec installed.
You need to convert every single file you may play to a caller
You can't force Asterisk to never attempt transcoding, the most you can do is 
force all sip.conf entries to use g729.  It will still transcode to play 
ringback to the caller.

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Khamis
Sent: Wednesday, August 14, 2013 10:45 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] G729 Passthrough How To

Hey!!! Eric thank you so much for your response. Could you guys please direct 
us in achieving as much as possible. For example:
* What linux command can we use to convert all recording to G729
* Which files do we need to convert and there locations
* For *testing* how do we make sure Asterisk NEVER EVER transcodes.

Do we still need the G729 codec installed on the asterisk machine if we manage 
to implement pass-through that would suffice our needs.

Kind Regards,

Nick.

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Re: [asterisk-users] G729 Passthrough How To

[Nick Khamis]

Hey Eric, I do have the codec installed, and I remember hearing about
the CLI command to convert. Is there a recent how-to of blog already
discussing this somewhere?

N.

On 8/14/13, Nick Khamis sym...@gmail.com wrote:
 I wanted to mention that I do not mind posting the converted files on
 this list for future individuals, given that I am not doing anything
 illegal...

 N.


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Re: [asterisk-users] G729 Passthrough How To

[Eric Wieling]

I have no idea, though Google might.

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Khamis
Sent: Wednesday, August 14, 2013 11:16 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] G729 Passthrough How To

Hey Eric, I do have the codec installed, and I remember hearing about the CLI 
command to convert. Is there a recent how-to of blog already discussing this 
somewhere?

N.

On 8/14/13, Nick Khamis sym...@gmail.com wrote:
 I wanted to mention that I do not mind posting the converted files on 
 this list for future individuals, given that I am not doing anything 
 illegal...

 N.


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Re: [asterisk-users] G729 Passthrough How To

[Nick Khamis]

I wanted to mention that I do not mind posting the converted files on
this list for future individuals, given that I am not doing anything
illegal...

N.

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Re: [asterisk-users] G729 Passthrough How To

[Nick Khamis]

Not really no... And how do I make sure Asterisk always generates
prompts and VM recordings in G729 from now on. This is also hard to
find information..


N.

On 8/14/13, Eric Wieling ewiel...@nyigc.com wrote:
 I have no idea, though Google might.

 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Khamis
 Sent: Wednesday, August 14, 2013 11:16 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] G729 Passthrough How To

 Hey Eric, I do have the codec installed, and I remember hearing about the
 CLI command to convert. Is there a recent how-to of blog already discussing
 this somewhere?

 N.

 On 8/14/13, Nick Khamis sym...@gmail.com wrote:
 I wanted to mention that I do not mind posting the converted files on
 this list for future individuals, given that I am not doing anything
 illegal...

 N.


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Re: [asterisk-users] G729 Passthrough How To

[Eric Wieling]

Asterisk does not generate prompts.   You force G729 in VM by only allowing 
g729 in voicemail.conf.Try reading the Asterisk book.  

Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at 
http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is 
released under a Creative Commons License 
(http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is 
available for reading online at http://www.asteriskdocs.org/

https://www.google.com/search?q=asterisk+%22file+convert%22

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Khamis
Sent: Wednesday, August 14, 2013 11:25 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] G729 Passthrough How To

Not really no... And how do I make sure Asterisk always generates prompts and 
VM recordings in G729 from now on. This is also hard to find information..


N.

On 8/14/13, Eric Wieling ewiel...@nyigc.com wrote:
 I have no idea, though Google might.

 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick 
 Khamis
 Sent: Wednesday, August 14, 2013 11:16 AM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: Re: [asterisk-users] G729 Passthrough How To

 Hey Eric, I do have the codec installed, and I remember hearing about 
 the CLI command to convert. Is there a recent how-to of blog already 
 discussing this somewhere?

 N.

 On 8/14/13, Nick Khamis sym...@gmail.com wrote:
 I wanted to mention that I do not mind posting the converted files on 
 this list for future individuals, given that I am not doing anything 
 illegal...

 N.


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[asterisk-users] G729 Passthrough How To

[Nick Khamis]

Hello Everyone,

We are currently experiencing some higher load on our servers, and
since signaling comes into our servers on G729, we would like to
implement G729 pass-through. A few questions arise, do we need to
convert all the recording to the codec, and what about voicemail?

We are also using A2Billing (hope I am not violating any thread
rules), and would like to convert all that recording to G729 as well.

Any help is greatly appreciated.

Kind Regards,

Nick from Toronto.

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[asterisk-users] g729 codec over SIP Trunk between CCM and Asterisk

[Onur Cem Çelebi]

Hello,

My problem is, outgoing calls (from asterisk to CCM) work fine but incoming
(from CCM to Asterisk) does not work because of CCM is trying to use g729
over SIP trunk. I have found that link after a quick search. Problem is the
same as in link below (However my Asterisk version is 1.8.13) and solution
seems to have H323 trunk between CCM and Asterisk for using g729 codec. The
post was written in 2006. Is there any better solution since that time ?
Thanks for reading.

link : g279 codec over SIP Trunk between CCM and
Asteriskhttps://supportforums.cisco.com/message/1072037
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Re: [asterisk-users] g729 codec over SIP Trunk between CCM and Asterisk

[Leandro Dardini]

2013/1/17 Onur Cem Çelebi occel...@gmail.com

 Hello,

 My problem is, outgoing calls (from asterisk to CCM) work fine but
 incoming (from CCM to Asterisk) does not work because of CCM is trying to
 use g729 over SIP trunk. I have found that link after a quick search.
 Problem is the same as in link below (However my Asterisk version is
 1.8.13) and solution seems to have H323 trunk between CCM and Asterisk for
 using g729 codec. The post was written in 2006. Is there any better
 solution since that time ? Thanks for reading.

 link : g279 codec over SIP Trunk between CCM and 
 Asteriskhttps://supportforums.cisco.com/message/1072037



Have you checked if the problem is the license? Asterisk doesn't have a
free encoder/decoder for g729, only pass through is available. Try to debug
the SIP call to see if the capabilities don't match or just buy a $10
license from Digium (1 concurrent call).

Leandro
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Re: [asterisk-users] g729 codec over SIP Trunk between CCM and Asterisk

[Onur Cem Çelebi]

Thanks for reply Leandro.

We have installed g279 codec in Asterisk box.Even if not so, there is no
problem outgoing (from Asterisk to CCM) calls. But after i searched  the
issue, i figured out that CCM 4.x does not let g729 codec to pass through
over SIP trunk. This is limited only in CCM. If we changed codec g729 into
g711u (ulaw) then communication over SIP trunk go on perfectly.

Because of CCM does not inject any packets encoded g729 over SIP trunk, i
am not able to debug it. But i have tried that i am able to force my SIP
phone suscribed Asterisk box to use g729 codec and get work successfully.

2013/1/17 Leandro Dardini ldard...@gmail.com

 2013/1/17 Onur Cem Çelebi occel...@gmail.com

 Hello,

 My problem is, outgoing calls (from asterisk to CCM) work fine but
 incoming (from CCM to Asterisk) does not work because of CCM is trying to
 use g729 over SIP trunk. I have found that link after a quick search.
 Problem is the same as in link below (However my Asterisk version is
 1.8.13) and solution seems to have H323 trunk between CCM and Asterisk for
 using g729 codec. The post was written in 2006. Is there any better
 solution since that time ? Thanks for reading.

 link : g279 codec over SIP Trunk between CCM and 
 Asteriskhttps://supportforums.cisco.com/message/1072037



 Have you checked if the problem is the license? Asterisk doesn't have a
 free encoder/decoder for g729, only pass through is available. Try to debug
 the SIP call to see if the capabilities don't match or just buy a $10
 license from Digium (1 concurrent call).

 Leandro

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[asterisk-users] G729 and voice mail

[Tim King]

I am trying to figure out the best way to deal with this. I want all of the
calls in the network to be G729 and this is working. I do have hardware
that provides me 30 g729 licenses. I am setting each extensions to
disallow=all and allow=g729. However when I have this setup, I get no voice
mail prompts. I tried setting to disallow=all and allow=g729,alaw and I
still have no audio when calling voice mail. If I remove the disallow=all I
do have voice mail prompts, but the calls do not seem to be always using
g729 when possible.
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Re: [asterisk-users] G729 and voice mail

[Eric Wieling]

What does the output of g729 show licenses show?  If it doesn't show licenses 
then Asterisk is not licensed for G729 codec.

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim King
Sent: Tuesday, June 05, 2012 2:32 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] G729 and voice mail

I am trying to figure out the best way to deal with this. I want all of the 
calls in the network to be G729 and this is working. I do have hardware that 
provides me 30 g729 licenses. I am setting each extensions to disallow=all and 
allow=g729. However when I have this setup, I get no voice mail prompts. I 
tried setting to disallow=all and allow=g729,alaw and I still have no audio 
when calling voice mail. If I remove the disallow=all I do have voice mail 
prompts, but the calls do not seem to be always using g729 when possible.


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Re: [asterisk-users] G729 and voice mail

[Tim King]

The G729 is coming from a Sangoma D100-030 card and the G729 transcoding is
working, the only issue I have is not hearing prompts from the system.

On Tue, Jun 5, 2012 at 2:42 PM, Eric Wieling ewiel...@nyigc.com wrote:

 What does the output of g729 show licenses show?  If it doesn't show
 licenses then Asterisk is not licensed for G729 codec.

 -Original Message-
 From: asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] On Behalf Of Tim King
 Sent: Tuesday, June 05, 2012 2:32 PM
 To: Asterisk Users Mailing List - Non-Commercial Discussion
 Subject: [asterisk-users] G729 and voice mail

 I am trying to figure out the best way to deal with this. I want all of
 the calls in the network to be G729 and this is working. I do have hardware
 that provides me 30 g729 licenses. I am setting each extensions to
 disallow=all and allow=g729. However when I have this setup, I get no voice
 mail prompts. I tried setting to disallow=all and allow=g729,alaw and I
 still have no audio when calling voice mail. If I remove the disallow=all I
 do have voice mail prompts, but the calls do not seem to be always using
 g729 when possible.


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Re: [asterisk-users] G729 and voice mail

[John Knight]

When you installed asterisk, did you also enable the installation of the 
g729 asterisk-(foo)-sounds options in 'make menuconfig'?


On 6/5/2012 2:46 PM, Tim King wrote:
The G729 is coming from a Sangoma D100-030 card and the G729 
transcoding is working, the only issue I have is not hearing prompts 
from the system.


On Tue, Jun 5, 2012 at 2:42 PM, Eric Wieling ewiel...@nyigc.com 
mailto:ewiel...@nyigc.com wrote:


What does the output of g729 show licenses show?  If it doesn't
show licenses then Asterisk is not licensed for G729 codec.

-Original Message-
From: asterisk-users-boun...@lists.digium.com
mailto:asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com
mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim
King
Sent: Tuesday, June 05, 2012 2:32 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] G729 and voice mail

I am trying to figure out the best way to deal with this. I want
all of the calls in the network to be G729 and this is working. I
do have hardware that provides me 30 g729 licenses. I am setting
each extensions to disallow=all and allow=g729. However when I
have this setup, I get no voice mail prompts. I tried setting to
disallow=all and allow=g729,alaw and I still have no audio when
calling voice mail. If I remove the disallow=all I do have voice
mail prompts, but the calls do not seem to be always using g729
when possible.


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Re: [asterisk-users] g729 freezes 1.8

[samuel]

I've just submitted a case in the official support web interface.

Thank you to all,

Samuel.

On 18 April 2012 14:52, Kevin P. Fleming kpflem...@digium.com wrote:

 On 04/18/2012 06:13 AM, A J Stiles wrote:

 On Wednesday 18 April 2012, samuel wrote:

 On 18 April 2012 10:33, A J 
 Stilesasterisk_list@**earthshod.co.ukasterisk_l...@earthshod.co.uk
  wrote:

 Are you sure your g729 module, your Asterisk and your kernel are of the
 same
 bittedness?

 I'm pretty sure it's not a problem of 32-64 bits:

 Asterisk 1.8.11.0 built by root   on a x86_64 running Linux on 2012-04-18
 07:45:43 UTC

 and I downladed the binaries from
 http://downloads.digium.com/**pub/telephony/codec_g729/**
 asterisk-1.8.0/x86-64/http://downloads.digium.com/pub/telephony/codec_g729/asterisk-1.8.0/x86-64/

 And asterisk loads the module, as you can see in the log files I sent.

 So it doesn't look like a problem with 32-64 bits


 Ah, well.  It's always worth a shot, though.

 It could still be a missing library; run `ldd` on the .so file(s), and
 make
 sure all needed libraries are installed.


 The simplest route to solving this problem is to contact Digium's support
 department; this is a Digium commercial product and you are entitled to
 technical support.

 The simple answer to your question is no, there are no known
 incompatibilities between Asterisk 1.8 and Digium's G.729 codec modules (if
 there were, we'd fix them).

 --
 Kevin P. Fleming
 Digium, Inc. | Director of Software Technologies
 Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
 Check us out at www.digium.com  www.asterisk.org


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Re: [asterisk-users] g729 freezes 1.8

[samuel]

Just in case it helps:

It turned out that from asterisk version 1.8.4 on, the g729 binaries are
different from the previous versions so it was a version mismatch between
the g729 (1.8.0_3.1.5) and asterisk (1.8.8 and higher).

Thanks to the Digium support department that found out the issue.

Samuel.

On 19 April 2012 08:18, samuel sam...@gmail.com wrote:

 I've just submitted a case in the official support web interface.

 Thank you to all,

 Samuel.


 On 18 April 2012 14:52, Kevin P. Fleming kpflem...@digium.com wrote:

 On 04/18/2012 06:13 AM, A J Stiles wrote:

 On Wednesday 18 April 2012, samuel wrote:

 On 18 April 2012 10:33, A J 
 Stilesasterisk_list@**earthshod.co.ukasterisk_l...@earthshod.co.uk
  wrote:

 Are you sure your g729 module, your Asterisk and your kernel are of the
 same
 bittedness?

 I'm pretty sure it's not a problem of 32-64 bits:

 Asterisk 1.8.11.0 built by root   on a x86_64 running Linux on
 2012-04-18
 07:45:43 UTC

 and I downladed the binaries from
 http://downloads.digium.com/**pub/telephony/codec_g729/**
 asterisk-1.8.0/x86-64/http://downloads.digium.com/pub/telephony/codec_g729/asterisk-1.8.0/x86-64/

 And asterisk loads the module, as you can see in the log files I sent.

 So it doesn't look like a problem with 32-64 bits


 Ah, well.  It's always worth a shot, though.

 It could still be a missing library; run `ldd` on the .so file(s), and
 make
 sure all needed libraries are installed.


 The simplest route to solving this problem is to contact Digium's support
 department; this is a Digium commercial product and you are entitled to
 technical support.

 The simple answer to your question is no, there are no known
 incompatibilities between Asterisk 1.8 and Digium's G.729 codec modules (if
 there were, we'd fix them).

 --
 Kevin P. Fleming
 Digium, Inc. | Director of Software Technologies
 Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype:
 kpfleming
 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
 Check us out at www.digium.com  www.asterisk.org


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Re: [asterisk-users] g729 freezes 1.8

[Kevin P. Fleming]

On 04/19/2012 10:57 AM, samuel wrote:

Just in case it helps:

It turned out that from asterisk version 1.8.4 on, the g729 binaries are
different from the previous versions so it was a version mismatch
between the g729 (1.8.0_3.1.5) and asterisk (1.8.8 and higher).


Ahh, and that is 'documented' on the G.729 download selector page here:

http://www.digium.com/en/docs/G729/g729-download.php

Did you use that download selector, or go directly to the 
downloads.digium.com site to grab the files?


--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com  www.asterisk.org

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Re: [asterisk-users] g729 freezes 1.8

[A J Stiles]

On Thursday 19 April 2012, samuel wrote:
 Just in case it helps:
 
 It turned out that from asterisk version 1.8.4 on, the g729 binaries are
 different from the previous versions so it was a version mismatch between
 the g729 (1.8.0_3.1.5) and asterisk (1.8.8 and higher).
 
 Thanks to the Digium support department that found out the issue.

Someone really needs to get the mPlayer folks  (based on the Continent, where 
mathematics is not patentable)  to create an Open Source g729 codec 
implementation .

-- 
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Answers come *after* questions.

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Re: [asterisk-users] g729 freezes 1.8

[Kevin P. Fleming]

On 04/19/2012 11:52 AM, A J Stiles wrote:

On Thursday 19 April 2012, samuel wrote:

Just in case it helps:

It turned out that from asterisk version 1.8.4 on, the g729 binaries are
different from the previous versions so it was a version mismatch between
the g729 (1.8.0_3.1.5) and asterisk (1.8.8 and higher).

Thanks to the Digium support department that found out the issue.


Someone really needs to get the mPlayer folks  (based on the Continent, where
mathematics is not patentable)  to create an Open Source g729 codec
implementation .


Source code availability is not the issue; the reference source code is 
easily obtained from the ITU-T. Many of the G.729 patent holders are 
companies based in Europe, so I suspect they would have a different 
opinion than you do about the legitimacy of their patent claims on G.729 :-)


In any case (and of course IANAL), it is my understanding that the 
patents that cover the base G.729 recommendation, along with Appendices 
A and B, will all expire in the next year or so. We'll have to see what 
that means for the market, especially with new, more freely licensed, 
codecs coming out that provide substantially better performance.


--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com  www.asterisk.org

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Re: [asterisk-users] g729 freezes 1.8

[Jeff Brower]

AJ-

 On Thursday 19 April 2012, samuel wrote:
 Just in case it helps:

 It turned out that from asterisk version 1.8.4 on, the g729 binaries are
 different from the previous versions so it was a version mismatch between
 the g729 (1.8.0_3.1.5) and asterisk (1.8.8 and higher).

 Thanks to the Digium support department that found out the issue.

 Someone really needs to get the mPlayer folks  (based on the Continent, where
 mathematics is not patentable)  to create an Open Source g729 codec
 implementation .

IMO

Transformations are patented, not mathematics.  This is true in the US also, if 
you want to create solid patents. 
Algorithms are certainly patented in both US and Europe -- Fraunhofer and MP3 
is a good example.  Although many people
disagree whether this should be allowed, there has yet to be high level court 
cases to decide the issue.

As for G729 and other codecs, they can also be implemented in hardware, in 
which case there would be novel circuit
apparatus that does the job.  And if someone used software to get the same 
results -- violation.  People too often
think that just because something can conveniently be done in software, 
traditional patent law no longer matters... 
that doesn't mean hardware approaches somehow disappeared.  People who write 
algorithm patents know this and make them
more solid using hardware techniques as additional methods.

You're dreaming if you think you can use G729 in ways other than what the 
patent holders grant.  The only reason you
don't get bothered (yet) is if you're not making money.

/IMO

-Jeff


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[asterisk-users] g729 freezes 1.8

[samuel]

Hi folks,

I was recently installing g729 support as usual (register, cp codec_g729 to
modules) and found out that the resulting asterisk instances freezes
whenever g729 commands are executed in the command line.

The registration process went OK and I can see the module being loaded by
asterisk:

[Apr 18 09:54:29] NOTICE[23775] codec_g729a.c: G.729A transcoding module
version 1.8.0_3.1.5, Copyright (C) 1999-2009 Digium, Inc.
[Apr 18 09:54:29] NOTICE[23775] codec_g729a.c: This module is supplied
under a commercial license granted by Digium, Inc.
[Apr 18 09:54:29] NOTICE[23775] codec_g729a.c: Please see the full license
text supplied by the accompanying
[Apr 18 09:54:29] NOTICE[23775] codec_g729a.c: register utility, or ask
for a copy from Digium.
[Apr 18 09:54:29] NOTICE[23775] codec_g729a.c: This product includes
software developed by the OpenSSL Project
[Apr 18 09:54:29] NOTICE[23775] codec_g729a.c: for use in the OpenSSL
Toolkit. (http://www.openssl.org/)
[Apr 18 09:54:29] NOTICE[23775] codec_g729a.c: Copyright (C) 1998-2006 The
OpenSSL Project

[Apr 18 09:54:29] VERBOSE[23775] manager.c:   == Manager registered action
G729LicenseStatus
[Apr 18 09:54:29] VERBOSE[23775] manager.c:   == Manager registered action
G729LicenseList
[Apr 18 09:54:29] VERBOSE[23775] codec_g729a.c:   == Host-ID:
7e:f8:73:46:bf:58:23:8f:82:10:6d:e5:59:6f:90:04:a8:30:74:fe
[Apr 18 09:54:29] VERBOSE[23775] codec_g729a.c:   == Found license
'G729-ZNSEND59XW4H' providing 25 channels
[Apr 18 09:54:29] VERBOSE[23775] codec_g729a.c:   == Found total of 25
G.729 licenses

After I can, sometimes, execute g729 CLI commands:
*CLI g729 show hostid
Host-ID: 7e:f8:73:46:bf:58:23:8f:82:10:6d:e5:59:6f:90:04:a8:30:74:fe

But no g729 seems to be loaded:
*CLI core show translation
 Translation times between formats (in microseconds) for one second
of data
  Source Format (Rows) Destination Format (Columns)

   g723   gsm  ulaw  alaw g726aal2 adpcm  slin lpc10  g729 speex
ilbc  g726  g722 siren7 siren14 slin16  g719 speex16 testlaw
 g723 - - - -- - - - -
- - - -  -   -  - -   -   -
  gsm - - 2 2- 2 1 2 - -
4001 - 2  -   -   4003 -   -   2
 ulaw - 2 - 2- 2 1 2 - -
4001 - 2  -   -   4003 -   -   2
 alaw - 2 2 -- 2 1 2 - -
4001 - 2  -   -   4003 -   -   2
 g726aal2 - - - -- - - - -
- - - -  -   -  - -   -   -
adpcm - 2 2 2- - 1 2 - -
4001 - 2  -   -   4003 -   -   2
 slin - 1 1 1- 1 - 1 - -
4000 - 1  -   -   4002 -   -   1
lpc10 - 2 2 2- 2 1 - - -
4001 - 2  -   -   4003 -   -   2
 g729 - - - -- - - - -
- - - -  -   -  - -   -   -
speex - - - -- - - - -
- - - -  -   -  - -   -   -
 ilbc - 2 2 2- 2 1 2 -
- - - 2  -   -   4003 -   -   2
 g726 - - - -- - - - -
- - - -  -   -  - -   -   -
 g722 - 2 2 2- 2 1 2 - -
4001 - -  -   -   4001 -   -   2
   siren7 - - - -- - - - -
- - - -  -   -  - -   -   -
  siren14 - - - -- - - - -
- - - -  -   -  - -   -   -
   slin16 - 3 3 3- 3 2 3 - -
4002 - 1  -   -  - -   -   3
 g719 - - - -- - - - -
- - - -  -   -  - -   -   -
  speex16 - - - -- - - - -
- - - -  -   -  - -   -   -
  testlaw - 2 2 2- 2 1 2 - -
4001 - 2  -   -   4003 -   -   -

Whenever I try to execute g729 show licencese, the CLI freezes and, in a
few, asterisk itself freeezes:
*CLI g729 show licenses
*CLI core show translation
*CLI core show translation

I've tried both 1.8.8.1 and 1.8.11.0 asterisk versions, and generic and
barcelona g729 binaries from
http://downloads.digium.com/pub/telephony/codec_g729/asterisk-1.8.0/x86-64/

Whenever I delete the codec_g729a.so module everything runs smoothly.

Re: [asterisk-users] g729 freezes 1.8

[A J Stiles]

On Wednesday 18 April 2012, samuel wrote:
 Hi folks,
 
 I was recently installing g729 support as usual (register, cp codec_g729 to
 modules) and found out that the resulting asterisk instances freezes
 whenever g729 commands are executed in the command line.
 . stuff deleted .

Are you sure your g729 module, your Asterisk and your kernel are of the same 
bittedness?

You cannot load 32-bit modules into an application which was compiled as 64-
bit.  This is not a problem if you built everything yourself from Source Code; 
but if anything was supplied pre-compiled and binary-only, you need to compile 
your Asterisk to match it.  (And next time, insist on the Source Code; after 
all, you're paying money for it.  Your right to know trumps other people's 
rights to keep secrets from you.)

-- 
AJS

Answers come *after* questions.

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Re: [asterisk-users] g729 freezes 1.8

[samuel]

I'm pretty sure it's not a problem of 32-64 bits:

Asterisk 1.8.11.0 built by root   on a x86_64 running Linux on 2012-04-18
07:45:43 UTC

and I downladed the binaries from
http://downloads.digium.com/pub/telephony/codec_g729/asterisk-1.8.0/x86-64/

And asterisk loads the module, as you can see in the log files I sent.

So it doesn't look like a problem with 32-64 bits

Thanks for the answer,
Samuel.

On 18 April 2012 10:33, A J Stiles asterisk_l...@earthshod.co.uk wrote:

 On Wednesday 18 April 2012, samuel wrote:
  Hi folks,
 
  I was recently installing g729 support as usual (register, cp codec_g729
 to
  modules) and found out that the resulting asterisk instances freezes
  whenever g729 commands are executed in the command line.
  . stuff deleted .

 Are you sure your g729 module, your Asterisk and your kernel are of the
 same
 bittedness?

 You cannot load 32-bit modules into an application which was compiled as
 64-
 bit.  This is not a problem if you built everything yourself from Source
 Code;
 but if anything was supplied pre-compiled and binary-only, you need to
 compile
 your Asterisk to match it.  (And next time, insist on the Source Code;
 after
 all, you're paying money for it.  Your right to know trumps other people's
 rights to keep secrets from you.)

 --
 AJS

 Answers come *after* questions.

 --
 _
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Re: [asterisk-users] g729 freezes 1.8

[samuel]

Just to confirm, I try to load a 32 bits g729 binary and asterisk doesn't
load it and complain about it:

[Apr 18 12:38:50] WARNING[2033] loader.c: Error loading module
'codec_g729a.so':
 /usr/lib/asterisk/modules/codec_g729a.so: wrong ELF class: ELFCLASS32
[Apr 18 12:38:50] WARNING[2033] loader.c: Module 'codec_g729a.so' could not
be l
oaded.

On 18 April 2012 12:36, samuel sam...@gmail.com wrote:

 I'm pretty sure it's not a problem of 32-64 bits:

 Asterisk 1.8.11.0 built by root   on a x86_64 running Linux on 2012-04-18
 07:45:43 UTC

 and I downladed the binaries from
 http://downloads.digium.com/pub/telephony/codec_g729/asterisk-1.8.0/x86-64/

 And asterisk loads the module, as you can see in the log files I sent.

 So it doesn't look like a problem with 32-64 bits

 Thanks for the answer,
 Samuel.


 On 18 April 2012 10:33, A J Stiles asterisk_l...@earthshod.co.uk wrote:

 On Wednesday 18 April 2012, samuel wrote:
  Hi folks,
 
  I was recently installing g729 support as usual (register, cp
 codec_g729 to
  modules) and found out that the resulting asterisk instances freezes
  whenever g729 commands are executed in the command line.
  . stuff deleted .

 Are you sure your g729 module, your Asterisk and your kernel are of the
 same
 bittedness?

 You cannot load 32-bit modules into an application which was compiled as
 64-
 bit.  This is not a problem if you built everything yourself from Source
 Code;
 but if anything was supplied pre-compiled and binary-only, you need to
 compile
 your Asterisk to match it.  (And next time, insist on the Source Code;
 after
 all, you're paying money for it.  Your right to know trumps other people's
 rights to keep secrets from you.)

 --
 AJS

 Answers come *after* questions.

 --
 _
 -- Bandwidth and Colocation Provided by http://www.api-digital.com --
 New to Asterisk? Join us for a live introductory webinar every Thurs:
   http://www.asterisk.org/hello

 asterisk-users mailing list
 To UNSUBSCRIBE or update options visit:
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Re: [asterisk-users] g729 freezes 1.8

[A J Stiles]

On Wednesday 18 April 2012, samuel wrote:
 On 18 April 2012 10:33, A J Stiles asterisk_l...@earthshod.co.uk wrote:
  Are you sure your g729 module, your Asterisk and your kernel are of the
  same
  bittedness?
 I'm pretty sure it's not a problem of 32-64 bits:
 
 Asterisk 1.8.11.0 built by root   on a x86_64 running Linux on 2012-04-18
 07:45:43 UTC
 
 and I downladed the binaries from
 http://downloads.digium.com/pub/telephony/codec_g729/asterisk-1.8.0/x86-64/
 
 And asterisk loads the module, as you can see in the log files I sent.
 
 So it doesn't look like a problem with 32-64 bits

Ah, well.  It's always worth a shot, though.

It could still be a missing library; run `ldd` on the .so file(s), and make 
sure all needed libraries are installed.

-- 
AJS

Answers come *after* questions.

--
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Re: [asterisk-users] g729 freezes 1.8

[Kevin P. Fleming]

On 04/18/2012 06:13 AM, A J Stiles wrote:

On Wednesday 18 April 2012, samuel wrote:

On 18 April 2012 10:33, A J Stilesasterisk_l...@earthshod.co.uk  wrote:

Are you sure your g729 module, your Asterisk and your kernel are of the
same
bittedness?

I'm pretty sure it's not a problem of 32-64 bits:

Asterisk 1.8.11.0 built by root   on a x86_64 running Linux on 2012-04-18
07:45:43 UTC

and I downladed the binaries from
http://downloads.digium.com/pub/telephony/codec_g729/asterisk-1.8.0/x86-64/

And asterisk loads the module, as you can see in the log files I sent.

So it doesn't look like a problem with 32-64 bits


Ah, well.  It's always worth a shot, though.

It could still be a missing library; run `ldd` on the .so file(s), and make
sure all needed libraries are installed.


The simplest route to solving this problem is to contact Digium's 
support department; this is a Digium commercial product and you are 
entitled to technical support.


The simple answer to your question is no, there are no known 
incompatibilities between Asterisk 1.8 and Digium's G.729 codec modules 
(if there were, we'd fix them).


--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com  www.asterisk.org

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[asterisk-users] G729 and Dahdi: Inbound forcing ulaw!

[JT]

Happy day to ya list!

I've recently deployed an update on our Asterisk server, taking it from 1.2
up to 1.8.5 - going from zaptel to dahdi.  Excitement levels are high and
performance so far is wonderful.

The issue I have is I've purchased several G729 licenses, registered them,
installed the module in asterisk (per instructions) and for most calls they
work wonderfully.

Inbound calls through the PRI however are forcing ulaw as the codec!  I
noticed this as all of my phones are currently set to Disallow:All,
Allow:G729,gsm.  (This was the same configuration with Asterisk 1.2/Zaptel).
 When an inbound call arrives my logs fill up with:
chan_sip.c: Asked to transmit frame type ulaw, while native formats is 0x100
(g729) read/write = 0x100 (g729)/0x100 (g729)

These did not appear with Asterisk 1.2/Zaptel.  I have reviewed the Dahdi
settings and have found nothing relating to the codec being passed or how to
force it to use the codecs I've purchased.

Any suggestions on where this might be set for inbound calls?  (Outbound
calls, SIP to SIP and SIP to IAX2 to SIP are all using G729 - this only
relates to an inbound call from an external land line).
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Re: [asterisk-users] G729 and Dahdi: Inbound forcing ulaw!

[Eric Wieling]

Upgrade to 1.8.7.1  There was a bug fixed recently (I think in 1.8.6, but might 
have been 1.8.7) which caused Asterisk to sometimes not transcode when it 
should.  

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of JT
Sent: Wednesday, October 19, 2011 3:45 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] G729 and Dahdi: Inbound forcing ulaw!

Happy day to ya list!

I've recently deployed an update on our Asterisk server, taking it from 1.2 up 
to 1.8.5 - going from zaptel to dahdi.  Excitement levels are high and 
performance so far is wonderful.

The issue I have is I've purchased several G729 licenses, registered them, 
installed the module in asterisk (per instructions) and for most calls they 
work wonderfully.

Inbound calls through the PRI however are forcing ulaw as the codec!  I noticed 
this as all of my phones are currently set to Disallow:All, Allow:G729,gsm.  
(This was the same configuration with Asterisk 1.2/Zaptel).  When an inbound 
call arrives my logs fill up with:
chan_sip.c: Asked to transmit frame type ulaw, while native formats is 0x100 
(g729) read/write = 0x100 (g729)/0x100 (g729)

These did not appear with Asterisk 1.2/Zaptel.  I have reviewed the Dahdi 
settings and have found nothing relating to the codec being passed or how to 
force it to use the codecs I've purchased.

Any suggestions on where this might be set for inbound calls?  (Outbound calls, 
SIP to SIP and SIP to IAX2 to SIP are all using G729 - this only relates to an 
inbound call from an external land line).



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Re: [asterisk-users] G729 and Dahdi: Inbound forcing ulaw!

[Richard Mudgett]

 Upgrade to 1.8.7.1 There was a bug fixed recently (I think in 1.8.6,
 but might have been 1.8.7) which caused Asterisk to sometimes not
 transcode when it should.

A regression introduced in v1.8.7 broke the ability of the ./configure
script to generate the HAVE_PRI_xxx defines for ISDN. Fix committed to
v1.8 branch with -r339719. It is fixed in v1.8.8-rc2.

You could simply use the ./configure script from v1.8.6.

Richard

 
 -Original Message-
 From: asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of JT
 Sent: Wednesday, October 19, 2011 3:45 PM
 To: asterisk-users@lists.digium.com
 Subject: [asterisk-users] G729 and Dahdi: Inbound forcing ulaw!
 
 Happy day to ya list!
 
 I've recently deployed an update on our Asterisk server, taking it
 from 1.2 up to 1.8.5 - going from zaptel to dahdi. Excitement levels
 are high and performance so far is wonderful.
 
 The issue I have is I've purchased several G729 licenses, registered
 them, installed the module in asterisk (per instructions) and for most
 calls they work wonderfully.
 
 Inbound calls through the PRI however are forcing ulaw as the codec! I
 noticed this as all of my phones are currently set to Disallow:All,
 Allow:G729,gsm. (This was the same configuration with Asterisk
 1.2/Zaptel). When an inbound call arrives my logs fill up with:
 chan_sip.c: Asked to transmit frame type ulaw, while native formats is
 0x100 (g729) read/write = 0x100 (g729)/0x100 (g729)
 
 These did not appear with Asterisk 1.2/Zaptel. I have reviewed the
 Dahdi settings and have found nothing relating to the codec being
 passed or how to force it to use the codecs I've purchased.
 
 Any suggestions on where this might be set for inbound calls?
 (Outbound calls, SIP to SIP and SIP to IAX2 to SIP are all using G729
 - this only relates to an inbound call from an external land line).

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[asterisk-users] g729 codec loading

[Adolphe Cher-Aime]

Hello  Everyone,
I've successfully registered my g729a  licenses.
When i try to load the module from asterisk Cli  i  got the following error

 *Error loading module 'codec_g729a.so':
/usr/lib/asterisk/modules/codec_g729a.so: cannot restore segment prot after
reloc: Permission denied*
* loader.c:795 load_resource: Module 'codec_g729a.so' could not be loaded.*
*
*
*I'm running asterisk 1.6.2.9 on CentOs 5.4 *
*
*
*Best regards*
*
*
-- 
Adolphe CHER-AIME
Network Integrator
CCNA, CCNA VOICE, Global VSAT Forum Certified
(509) 3748-3875 / (509) 3449-4280
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Re: [asterisk-users] g729 codec loading

[Moises Silva]

Try disabling SELinux if you have it enabled (unless of course you need it).
I seem to remember there is certain compilation flags required (position
independent code, -fPIC?) to run with SELinux enabled, may be the
codec_g729a.so is not compiled properly to run under such circumstances?

Moises Silva
Senior Software Engineer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. m...@sangoma.com


On Fri, Jul 16, 2010 at 5:57 PM, Adolphe Cher-Aime achera...@gmail.comwrote:

 Hello  Everyone,
 I've successfully registered my g729a  licenses.
 When i try to load the module from asterisk Cli  i  got the following error

  *Error loading module 'codec_g729a.so':
 /usr/lib/asterisk/modules/codec_g729a.so: cannot restore segment prot after
 reloc: Permission denied*
 * loader.c:795 load_resource: Module 'codec_g729a.so' could not be loaded.
 *
 *
 *
 *I'm running asterisk 1.6.2.9 on CentOs 5.4 *
 *
 *
 *Best regards*
 *
 *
 --
 Adolphe CHER-AIME
 Network Integrator
 CCNA, CCNA VOICE, Global VSAT Forum Certified
 (509) 3748-3875 / (509) 3449-4280

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Re: [asterisk-users] g729 codec loading

[Adolphe Cher-Aime]

Thank  you Moises it works  perfectly.



On Fri, Jul 16, 2010 at 5:07 PM, Moises Silva moises.si...@gmail.comwrote:

 Try disabling SELinux if you have it enabled (unless of course you need
 it). I seem to remember there is certain compilation flags required
 (position independent code, -fPIC?) to run with SELinux enabled, may be the
 codec_g729a.so is not compiled properly to run under such circumstances?

 Moises Silva
 Senior Software Engineer
 Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R
 9T3 Canada
 t. 1 905 474 1990 x 128 | e. m...@sangoma.com


 On Fri, Jul 16, 2010 at 5:57 PM, Adolphe Cher-Aime achera...@gmail.comwrote:

 Hello  Everyone,
 I've successfully registered my g729a  licenses.
 When i try to load the module from asterisk Cli  i  got the following error

  *Error loading module 'codec_g729a.so':
 /usr/lib/asterisk/modules/codec_g729a.so: cannot restore segment prot after
 reloc: Permission denied*
 * loader.c:795 load_resource: Module 'codec_g729a.so' could not be
 loaded.*
 *
 *
 *I'm running asterisk 1.6.2.9 on CentOs 5.4 *
 *
 *
 *Best regards*
 *
 *
 --
 Adolphe CHER-AIME
 Network Integrator
 CCNA, CCNA VOICE, Global VSAT Forum Certified
 (509) 3748-3875 / (509) 3449-4280

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-- 
Adolphe CHER-AIME
Network Integrator
CCNA, CCNA VOICE, Global VSAT Forum Certified
(509) 3748-3875 / (509) 3449-4280
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Re: [asterisk-users] G729 license key registration

[Kiss András]

2010/6/25 Remco Bressers rbress...@signet.nl:
 On 06/25/2010 09:48 AM, Kiss András wrote:
 You selected 5, G.729 Codec
 Please enter your Key-ID: G729-10D2X----X
 This product key cannot be registered!  Please verify you entered the
 correct product key.
 Server response: 404 - Key not found.

 Any suggestions?

 How about contacting Digium about this?

I`ve tried, got no response (yet)
I`ve found a thread on this list with the same problem, without the
solution: 
http://www.mail-archive.com/asterisk-users@lists.digium.com/msg219314.html

- András

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Re: [asterisk-users] G729 license key registration

[Gordon Henderson]

On Tue, 29 Jun 2010, Kiss András wrote:


2010/6/25 Remco Bressers rbress...@signet.nl:

On 06/25/2010 09:48 AM, Kiss András wrote:

You selected 5, G.729 Codec
Please enter your Key-ID: G729-10D2X----X
This product key cannot be registered!  Please verify you entered the
correct product key.
Server response: 404 - Key not found.

Any suggestions?


How about contacting Digium about this?


I`ve tried, got no response (yet)
I`ve found a thread on this list with the same problem, without the
solution: 
http://www.mail-archive.com/asterisk-users@lists.digium.com/msg219314.html


Dump the digium one and download the free one and make sure you don't 
utilise more channels of g729 than you've paid licenses for... (or just 
buy more licenses when you need more channels)


Probably not the technically correct way to do things, but no-one replied 
the last time I suggested this on the list...


Or try a commercially licensed alternative - e.g. 
http://www.howlertech.com/products/howlets/ perhaps their licensing works 
better?


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[asterisk-users] G729 license key registration

[Kiss András]

Hi,

I have trouble re-registering a G729 license for Asterisk (bought 6 years ago)
My license looks like: 10D2X----X

Tried to re-register the codec according to the
http://downloads.digium.com/pub/telephony/codec_g729/README document,
but the register failed with this error message:

You selected 5, G.729 Codec
Please enter your Key-ID: 10D2X----X
This product key cannot be registered!  Please verify you entered the
correct product key.
Server response: ERR - Invalid prefix, should be 'G729'

The program can communicate correctly with digium`s server.
Tried with the G729 prefix, but it seems completely wrong:

You selected 5, G.729 Codec
Please enter your Key-ID: G729-10D2X----X
This product key cannot be registered!  Please verify you entered the
correct product key.
Server response: 404 - Key not found.


Any suggestions?

- Andras

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Re: [asterisk-users] G729 license key registration

[Remco Bressers]

On 06/25/2010 09:48 AM, Kiss András wrote:
 You selected 5, G.729 Codec
 Please enter your Key-ID: G729-10D2X----X
 This product key cannot be registered!  Please verify you entered the
 correct product key.
 Server response: 404 - Key not found.
 
 Any suggestions?

How about contacting Digium about this?

-- 
Met vriendelijke groet,
Signet bv


Remco Bressers

T 040 - 707 4 907
F 040 - 707 4 909
E rbress...@signet.nl

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Re: [asterisk-users] G729 exhaustion conditions

[Zhang Shukun]

G729 is free for use is no transcoding is done.

2010/4/19 Harel Cohen ha...@easycall.gi:
 Hi all,

 Suppose I buy and install one G729 codec. Suppose there is one call going on
 where both end-points have G729 codecs and the Asterisk is not doing any
 transcoding. Does this conversation exhaust my G729 license (even though
 this call would have worked without license in the first place) or do I
 still have the ability to use this G729 codec for other call which requires
 transcoding?

 Thank you,

 Harel Cohen

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-- 
Best regards,
Sucan

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[asterisk-users] G729 exhaustion conditions

[Harel Cohen]

Hi all,
Suppose I buy and install one G729 codec. Suppose there is one call going on 
where both end-points have G729 codecs and the Asterisk is not doing any 
transcoding. Does this conversation exhaust my G729 license (even though this 
call would have worked without license in the first place) or do I still have 
the ability to use this G729 codec for other call which requires transcoding?
Thank you,
Harel Cohen
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Re: [asterisk-users] G729 Pass through

[Dovey Forman]

The SDP response is the IP of the Trixbox server.

On Fri, Dec 11, 2009 at 1:38 PM, Christian Victor
christ...@victormedia.dewrote:

 Hi!

 Are you sure you are getting Astrisk out of the media path? I guess
 reinvite must be allowed. Then it should work without transcoding
 licenses.

 Maybe you should take a look at the SIP DEBUG info to see what codec
 Asterisk is trying to negotiate with the trunk. You could disallow
 alaw and ulaw for a test.

 Christian

 2009/12/11 Dovey Forman dovey.for...@idt.net:
  Hi;
 
 
 
  I am running Asterisk (Trix) 1.2.17 with Aastra 6731i phones as my
  endpoints.
 
 
 
  It seems that when I enable G729 on my peers in sip.conf and make a call
 I
  am getting the following errors:
 
 
 
  Called crp_uk/806575011971553141421
 
  Dec 11 07:57:10 WARNING[31903] channel.c: Unable to find a codec
 translation
  path from g729 to ulaw
 
  Dec 11 07:57:10 WARNING[20633] chan_sip.c: Asked to transmit frame type
 256,
  while native formats is 4 (read/write = 4/4)
 
 
 
  Both my end points (Aastra phone) and my sip carrier support G729, so
 this
  should be simple pass-through.
 
 
 
  Snippet of my peer crp_uk:
 
 
 
  [crp_uk]
 
  disallow=all
 
  allow=ulaw
 
  allow=alaw
 
  allow=g729
 
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[asterisk-users] G729 Pass through

[Dovey Forman]

Hi;



I am running Asterisk (Trix) 1.2.17 with Aastra 6731i phones as my
endpoints.



It seems that when I enable G729 on my peers in sip.conf and make a call I
am getting the following errors:



Called crp_uk/806575011971553141421

Dec 11 07:57:10 WARNING[31903] channel.c: Unable to find a codec translation
path from g729 to ulaw

Dec 11 07:57:10 WARNING[20633] chan_sip.c: Asked to transmit frame type 256,
while native formats is 4 (read/write = 4/4)



Both my end points (Aastra phone) and my sip carrier support G729, so this
should be simple pass-through.



Snippet of my peer crp_uk:



[crp_uk]

disallow=all

allow=ulaw

allow=alaw

allow=g729
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Re: [asterisk-users] G729 Pass through

[James A. Shigley]

Have you paied for and imported g729 licenses from digium so that
asterisks can use g729?

 

http://store.digium.com/productview.php?category_id=5product_code=8G729
CODEC 

 

James Shigley

Monroe Telephone Answering Service

409-981-9213

Infinity 5.51,UC 4.02.3803, Blink 3.0.104

Ecreator:2.21, eResponse 1.1.7

Webportal,WebApps, 

 

CONFIDENTIALITY NOTICE: This email, including any attachments, contains
information which may be confidential or privileged. The information =is
intended to be for the use of the individual or entity named above. If
you are not the intended recipient, be aware that any disclosure,
copying, distribution or use of the contents of this information is
prohibited. If you have received this email in error, please notify the
sender immediately by reply to sender only message and destroy all
electronic and hard copies of the communication, including attachments. 

 

 

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dovey
Forman
Sent: Friday, December 11, 2009 12:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] G729 Pass through
Importance: High

 

Hi;

 

I am running Asterisk (Trix) 1.2.17 with Aastra 6731i phones as my
endpoints.

 

It seems that when I enable G729 on my peers in sip.conf and make a call
I am getting the following errors:

 

Called crp_uk/806575011971553141421

Dec 11 07:57:10 WARNING[31903] channel.c: Unable to find a codec
translation path from g729 to ulaw

Dec 11 07:57:10 WARNING[20633] chan_sip.c: Asked to transmit frame type
256, while native formats is 4 (read/write = 4/4)

 

Both my end points (Aastra phone) and my sip carrier support G729, so
this should be simple pass-through.

 

Snippet of my peer crp_uk:

 

[crp_uk]

disallow=all

allow=ulaw

allow=alaw

allow=g729

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Re: [asterisk-users] G729 Pass through

[Dovey Forman]

I have – but I don’t see why that would be required for pass – through?

The codec purchase should only be required if I wanted to leave voicemail in
G729 or MOH.



If my end points support G729 and I am advertising it in the invite, and
negotiating it with the 200OK, I don’t see why its not allowing pass
through.



--Dovey


 --

*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *James A. Shigley
*Sent:* Friday, December 11, 2009 1:16 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] G729 Pass through



Have you paied for and imported g729 licenses from digium so that asterisks
can use g729?



http://store.digium.com/productview.php?category_id=5product_code=8G729CODEC



James Shigley

*Monroe** Telephone Answering Service*

409-981-9213**

Infinity 5.51,UC 4.02.3803, Blink 3.0.104

Ecreator:2.21, eResponse 1.1.7

Webportal,WebApps,



CONFIDENTIALITY NOTICE: This email, including any attachments, contains
information which may be confidential or privileged. The information =is
intended to be for the use of the individual or entity named above. If you
are not the intended recipient, be aware that any disclosure, copying,
distribution or use of the contents of this information is prohibited. If
you have received this email in error, please notify the sender immediately
by reply to sender only message and destroy all electronic and hard copies
of the communication, including attachments.



[image: cid:image003.png@01C9F268.65A4F5C0]



*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Dovey Forman
*Sent:* Friday, December 11, 2009 12:06 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] G729 Pass through
*Importance:* High



Hi;



I am running Asterisk (Trix) 1.2.17 with Aastra 6731i phones as my
endpoints.



It seems that when I enable G729 on my peers in sip.conf and make a call I
am getting the following errors:



Called crp_uk/806575011971553141421

Dec 11 07:57:10 WARNING[31903] channel.c: Unable to find a codec translation
path from g729 to ulaw

Dec 11 07:57:10 WARNING[20633] chan_sip.c: Asked to transmit frame type 256,
while native formats is 4 (read/write = 4/4)



Both my end points (Aastra phone) and my sip carrier support G729, so this
should be simple pass-through.



Snippet of my peer crp_uk:



[crp_uk]

disallow=all

allow=ulaw

allow=alaw

allow=g729
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Re: [asterisk-users] G729 Pass through

[Christian Victor]

Hi!

Are you sure you are getting Astrisk out of the media path? I guess
reinvite must be allowed. Then it should work without transcoding
licenses.

Maybe you should take a look at the SIP DEBUG info to see what codec
Asterisk is trying to negotiate with the trunk. You could disallow
alaw and ulaw for a test.

Christian

2009/12/11 Dovey Forman dovey.for...@idt.net:
 Hi;



 I am running Asterisk (Trix) 1.2.17 with Aastra 6731i phones as my
 endpoints.



 It seems that when I enable G729 on my peers in sip.conf and make a call I
 am getting the following errors:



 Called crp_uk/806575011971553141421

 Dec 11 07:57:10 WARNING[31903] channel.c: Unable to find a codec translation
 path from g729 to ulaw

 Dec 11 07:57:10 WARNING[20633] chan_sip.c: Asked to transmit frame type 256,
 while native formats is 4 (read/write = 4/4)



 Both my end points (Aastra phone) and my sip carrier support G729, so this
 should be simple pass-through.



 Snippet of my peer crp_uk:



 [crp_uk]

 disallow=all

 allow=ulaw

 allow=alaw

 allow=g729

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Re: [asterisk-users] G729 with IAX

[Steve Johnson]

Of course, as long as your endpoints support it.  Read more about it
and purchase G.729 channel licenses for Asterisk from Digium:

http://www.digium.com/en/products/g729codec.php

Once you have the codec properly installed, enable it for your peer in
your iax.conf file allow=g729.  Restart asterisk and go to it.

Also, Google is your friend. Search: g729 iax
for lots of information and examples.


On Tue, Dec 8, 2009 at 1:03 AM, DHAVAL INDRODIYA
dhaval.it01...@gmail.com wrote:
 dear All,

 can I use G729 with IAX trunk or IAX calls

 regards
 Dhaval

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[asterisk-users] G729 with IAX

[DHAVAL INDRODIYA]

dear All,

can I use G729 with IAX trunk or IAX calls

regards
Dhaval
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[asterisk-users] G729: TC400B vs Software Encoder

[Jim Boykin]

Can someone help me decide between TC400B vs Software Encoder. TC400B
is really expensive and would like to get opinion if it is worth
considering. Maximum simultaneous call we handle is hardly 10-15. Will
going to TC400B improves quality. Any other option then TC400B.

Thanks
Jim

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Re: [asterisk-users] G729: TC400B vs Software Encoder

[Jim Boykin]

Thanks. What do you suggest for system with 15 max simultaneous calls?



On Sat, Dec 5, 2009 at 7:49 PM, Kevin P. Fleming kpflem...@digium.com wrote:
 Jim Boykin wrote:
 Can someone help me decide between TC400B vs Software Encoder. TC400B
 is really expensive and would like to get opinion if it is worth
 considering. Maximum simultaneous call we handle is hardly 10-15. Will
 going to TC400B improves quality. Any other option then TC400B.

 All G.729 encoders produce the same quality, since they all do the same
 thing. The only difference between various encoders is the amount of
 load they play on the system CPU, cost, licensing models, etc.

 --
 Kevin P. Fleming
 Digium, Inc. | Director of Software Technologies
 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
 skype: kpfleming | jabber: kpflem...@digium.com
 Check us out at www.digium.com  www.asterisk.org

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Re: [asterisk-users] G729: TC400B vs Software Encoder

[Kevin P. Fleming]

Jim Boykin wrote:
 Can someone help me decide between TC400B vs Software Encoder. TC400B
 is really expensive and would like to get opinion if it is worth
 considering. Maximum simultaneous call we handle is hardly 10-15. Will
 going to TC400B improves quality. Any other option then TC400B.

All G.729 encoders produce the same quality, since they all do the same
thing. The only difference between various encoders is the amount of
load they play on the system CPU, cost, licensing models, etc.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kpflem...@digium.com
Check us out at www.digium.com  www.asterisk.org

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Re: [asterisk-users] G729: TC400B vs Software Encoder

[Steve Underwood]

On 12/05/2009 10:19 PM, Kevin P. Fleming wrote:
 Jim Boykin wrote:

 Can someone help me decide between TC400B vs Software Encoder. TC400B
 is really expensive and would like to get opinion if it is worth
 considering. Maximum simultaneous call we handle is hardly 10-15. Will
 going to TC400B improves quality. Any other option then TC400B.
  
 All G.729 encoders produce the same quality, since they all do the same
 thing. The only difference between various encoders is the amount of
 load they play on the system CPU, cost, licensing models, etc.

That's not strictly true. All fixed point G.729 encoders should produce 
a bit exact output that will match the ITU test vectors. However, most 
G.729 encoders on PCs use floating point arithmetic, and there is some 
variability in the output. It should be a pretty small difference, though.

Steve



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Re: [asterisk-users] G729: TC400B vs Software Encoder

[Tilghman Lesher]

On Saturday 05 December 2009 08:25:53 Jim Boykin wrote:
 Thanks. What do you suggest for system with 15 max simultaneous calls?

As long as your machine is at least a single CPU, 2GHz or better, you should
be fine with the software encoder.  The TC400 is really only necessary on
systems where the transcoding load alone would overtax the CPU.  It's really
difficult to estimate a good number of where you should start considering the
TC400, because there are so many other tasks where the CPU may be required
and would affect the overall result, but the number is probably somewhere
north of 40 simultaneous calls.

I'm sure somebody will chime in and say they're using 100 simultaneous calls
on the software encoder alone.  As I said, the number is extremely fungible,
based upon what else the CPU is doing.

-- 
Tilghman Lesher
Digium, Inc. | Senior Software Developer
twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
Check us out at: www.digium.com  www.asterisk.org

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Re: [asterisk-users] G729 in asterisk upgrade issue

[Luis Silva]

Hi, 

Finally I made the upgrade. Everything work well, but I have an issue with
one of the G729 licenses, I can't load it.

I have two other license files that load with no problems.

The host id is the same in all the files, can you give me an idea how to
check this problem?

  

 

Kevin P. Fleming wrote:

 

 Luis Silva wrote:

 

 I have an asterisk in 1.2 version with 30 g729 licenses. I what to 

 upgrade it to 1.4 and also upgrade the O.S. from fedora 8 to centos 5.3.

 

 For what I understand I can make the backup of the license files in
/var/lib/asterisk/licenses/  if I need to reinstall the operating system,
but in this case with new O.S and new version can I reuse this licenses?

 

 Yes.

 

Thanks for the information Kevin.

 

Regards,

LS

 

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Re: [asterisk-users] G729 in asterisk upgrade issue

[Kevin P. Fleming]

Luis Silva wrote:

 Finally I made the upgrade. Everything work well, but I have an issue
 with one of the G729 licenses, I can’t load it…
 
 I have two other license files that load with no problems.
 
 The host id is the same in all the files, can you give me an idea how to
 check this problem?

You will need to contact Digium's support department; it is likely that
you have a very old license file, which is formatted differently than
the current codec_g729 module expects. There was a change in our g729
license files a number of years ago, and the support department can help
upgrade your license key to the current version.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kpflem...@digium.com
Check us out at www.digium.com  www.asterisk.org


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Re: [asterisk-users] G729 in asterisk upgrade issue

[Luis Silva]

 

Kevin P. Fleming wrote:

 

 Luis Silva wrote:

 

 I have an asterisk in 1.2 version with 30 g729 licenses. I what to 

 upgrade it to 1.4 and also upgrade the O.S. from fedora 8 to centos 5.3.

 

 For what I understand I can make the backup of the license files in
/var/lib/asterisk/licenses/  if I need to reinstall the operating system,
but in this case with new O.S and new version can I reuse this licenses?

 

 Yes.

 

Thanks for the information Kevin.

 

 

Danny Nicholas wrote:

 Speaking blindly here, you should be able to unless there is some kind of
server architecture involved (probably not with /v/l/a path.

 

Danny, also thanks for the interest.

 

Regards,

LS

 

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[asterisk-users] G729 in asterisk upgrade issue

[Luis Silva]

Hi, 

I need some help. 

I have an asterisk in 1.2 version with 30 g729 licenses. I what to upgrade
it to 1.4 and also upgrade the O.S. from fedora 8 to centos 5.3.

For what I understand I can make the backup of the license files in
/var/lib/asterisk/licenses/  if I need to reinstall the operating system,
but in this case with new O.S and new version can I reuse this licenses?
 
Regards
Luis Silva

 

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Re: [asterisk-users] G729 in asterisk upgrade issue

[Danny Nicholas]

Speaking blindly here, you should be able to unless there is some kind of
server architecture involved (probably not with /v/l/a path.)

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Luis Silva
Sent: Monday, October 12, 2009 12:36 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] G729 in asterisk upgrade issue

 

Hi, 

I need some help. 

I have an asterisk in 1.2 version with 30 g729 licenses. I what to upgrade
it to 1.4 and also upgrade the O.S. from fedora 8 to centos 5.3.

For what I understand I can make the backup of the license files in
/var/lib/asterisk/licenses/  if I need to reinstall the operating system,
but in this case with new O.S and new version can I reuse this licenses?
 
Regards
Luis Silva

 

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Re: [asterisk-users] G729 in asterisk upgrade issue

[Kevin P. Fleming]

Luis Silva wrote:

 I have an asterisk in 1.2 version with 30 g729 licenses. I what to
 upgrade it to 1.4 and also upgrade the O.S. from fedora 8 to centos 5.3.
 
 For what I understand I can make the backup of the license files in 
 /var/lib/asterisk/licenses/  if I need to reinstall the operating system, but 
 in this case with new O.S and new version can I reuse this licenses?

Yes.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kpflem...@digium.com
Check us out at www.digium.com  www.asterisk.org

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Re: [asterisk-users] g729 free codec any idea

[Michelle Dupuis]

Before we call each other liars and thieves, here is a link:

http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/

As with any open source, do your own homework on licensing, AND apply your
own reasonable judgment.

At this point I would like to confess that I watch rented DVD's under
Linux...which is illegal since there is no license for decrypting commercial
DVD's under Linux.  I'm overwhelmed with guilt and feel better having that
off my chest.

As the sparks fly I'll happily walk away from this thread :)

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt Riddell
Sent: Thursday, October 08, 2009 10:47 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] g729 free codec any idea

On 9/10/09 3:31 PM, Michelle Dupuis wrote:
 I believe that Intel placed a 729 codec into the public domain (free), 
 and someone wrapped it in a nice Asterisk package for use.
 No idea where - but I do recall that it is out there, and legal. Of 
 course it's nice to support a vendor, but free alternatives can't be 
 shunned...

The original comment stands.  The codec is patented.

The implementation is not.

In order to use the implementation you need a license unless you live
somewhere that:

A) Doesn't have patents
B) Doesn't have a trade agreement with USA

Inserting a g729 codec from a licensed source other than Digium will break
the GPL (Digium issues an exception for it's g729):

http://tinyurl.com/ykpu42u

This conversation has come up hundreds of times on this mailing list and the
result is always the same - if you're happy breaking the law, go for it - if
you get most of your movies from piratebay then it probably isn't a problem
for you.

--
Cheers,

Matt Riddell
Director
___

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Re: [asterisk-users] g729 free codec any idea

[Jeff LaCoursiere]

On Fri, 9 Oct 2009, Michelle Dupuis wrote:

 Before we call each other liars and thieves, here is a link:

 http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/

 As with any open source, do your own homework on licensing, AND apply your
 own reasonable judgment.

 At this point I would like to confess that I watch rented DVD's under
 Linux...which is illegal since there is no license for decrypting commercial
 DVD's under Linux.  I'm overwhelmed with guilt and feel better having that
 off my chest.

 As the sparks fly I'll happily walk away from this thread :)


If you read the entire page of what you are linking to above you will see 
that it verifies what everyone has been saying all along - it is NOT FREE. 
If you want to use it you are required to pay royalties.  You said it was 
free.  It is not.



j

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Re: [asterisk-users] g729 free codec any idea

[Tim Nelson]

- DHAVAL INDRODIYA dhaval.it01...@gmail.com wrote: 
STUFF 
any help appreciated 

There is no *FREE* G.729 codec... 

Go to Digium.com and purchase the licensed G.729 codec. Yes, it's $10 per 
channel but as a nice side effect, you'll also get a supported, working G.729 
implementation where you can get assistance with a simple phone call instead of 
having to 'try so many modules'. 

Howler Technologies also has a G.729 implementation but I have yet to 
experience their product or support. 

Tim Nelson 
Systems/Network Support 
Rockbochs Inc. 
(218)727-4332 x105 ___
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Re: [asterisk-users] g729 free codec any idea

[Danny Nicholas]

Maybe you can bum a license or two off of the Astricon attendees.

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Nelson
Sent: Thursday, October 08, 2009 12:19 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] g729 free codec any idea

 

- DHAVAL INDRODIYA dhaval.it01...@gmail.com wrote: 
STUFF 

any help appreciated

 

There is no *FREE* G.729 codec...

Go to Digium.com and purchase the licensed G.729 codec. Yes, it's $10 per
channel but as a nice side effect, you'll also get a supported, working
G.729 implementation where you can get assistance with a simple phone call
instead of having to 'try so many modules'.

Howler Technologies also has a G.729 implementation but I have yet to
experience their product or support.

Tim Nelson
Systems/Network Support
Rockbochs Inc.
(218)727-4332 x105

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Re: [asterisk-users] g729 free codec any idea

[DHAVAL INDRODIYA]

there is free implementation of g729 codec
you can get it from http://asterisk.hosting.lv

On Thu, Oct 8, 2009 at 10:53 PM, Danny Nicholas da...@debsinc.com wrote:

  Maybe you can bum a license or two off of the Astricon attendees…


  --

 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *Tim Nelson
 *Sent:* Thursday, October 08, 2009 12:19 PM
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* Re: [asterisk-users] g729 free codec any idea



 - DHAVAL INDRODIYA dhaval.it01...@gmail.com wrote:
 STUFF

 any help appreciated



 There is no *FREE* G.729 codec...

 Go to Digium.com and purchase the licensed G.729 codec. Yes, it's $10 per
 channel but as a nice side effect, you'll also get a supported, working
 G.729 implementation where you can get assistance with a simple phone call
 instead of having to 'try so many modules'.

 Howler Technologies also has a G.729 implementation but I have yet to
 experience their product or support.

 Tim Nelson
 Systems/Network Support
 Rockbochs Inc.
 (218)727-4332 x105

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Re: [asterisk-users] g729 free codec any idea

[Moises Silva]

On Thu, Oct 8, 2009 at 1:33 PM, DHAVAL INDRODIYA
dhaval.it01...@gmail.comwrote:

 there is free implementation of g729 codec
 you can get it from http://asterisk.hosting.lv


I'm not an expert on patents, but even when you have access to the g729
implementation, the algorithm is patented, so, as the disclaimer says in the
web page, you still need to pay royalty fees to the g729 patent holders
somehow. Unless you live in a country where patents do not matter.

-- 
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] g729 free codec any idea

[Dovid Bender]

Just because something is available doesn't mean that it is legal. You can get 
TV Shows, Movies etc. on the internet but just because it's there it doesn't 
mean that you should use it. Digium supports the Asterisk project. Shouldn't 
you show your appreciation back to them ?
  - Original Message - 
  From: DHAVAL INDRODIYA 
  To: Asterisk Users Mailing List - Non-Commercial Discussion 
  Sent: Thursday, October 08, 2009 19:33
  Subject: Re: [asterisk-users] g729 free codec any idea


  there is free implementation of g729 codec 


  you can get it from http://asterisk.hosting.lv


  On Thu, Oct 8, 2009 at 10:53 PM, Danny Nicholas da...@debsinc.com wrote:

Maybe you can bum a license or two off of the Astricon attendees…






From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Nelson
Sent: Thursday, October 08, 2009 12:19 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] g729 free codec any idea



- DHAVAL INDRODIYA dhaval.it01...@gmail.com wrote: 
STUFF 

any help appreciated



There is no *FREE* G.729 codec...

Go to Digium.com and purchase the licensed G.729 codec. Yes, it's $10 per 
channel but as a nice side effect, you'll also get a supported, working G.729 
implementation where you can get assistance with a simple phone call instead of 
having to 'try so many modules'.

Howler Technologies also has a G.729 implementation but I have yet to 
experience their product or support.

Tim Nelson
Systems/Network Support
Rockbochs Inc.
(218)727-4332 x105


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--


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Re: [asterisk-users] g729 free codec any idea

[Michelle Dupuis]

I believe that Intel placed a 729 codec into the public domain (free), and
someone wrapped it in a nice Asterisk package for use.
 
No idea where - but I do recall that it is out there, and legal.  Of course
it's nice to support a vendor, but free alternatives can't be shunned...

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dovid Bender
Sent: Thursday, October 08, 2009 8:37 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] g729 free codec any idea


Just because something is available doesn't mean that it is legal. You can
get TV Shows, Movies etc. on the internet but just because it's there it
doesn't mean that you should use it. Digium supports the Asterisk project.
Shouldn't you show your appreciation back to them ?

- Original Message - 
From: DHAVAL INDRODIYA mailto:dhaval.it01...@gmail.com  
To: Asterisk Users Mailing List -  mailto:asterisk-users@lists.digium.com
Non-Commercial Discussion 
Sent: Thursday, October 08, 2009 19:33
Subject: Re: [asterisk-users] g729 free codec any idea

there is free implementation of g729 codec  

you can get it from http://asterisk.hosting.lv


On Thu, Oct 8, 2009 at 10:53 PM, Danny Nicholas da...@debsinc.com wrote:


Maybe you can bum a license or two off of the Astricon attendees.

 


  _  


From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Nelson
Sent: Thursday, October 08, 2009 12:19 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] g729 free codec any idea

 

- DHAVAL INDRODIYA dhaval.it01...@gmail.com wrote: 
STUFF 

any help appreciated

 

There is no *FREE* G.729 codec...

Go to Digium.com and purchase the licensed G.729 codec. Yes, it's $10 per
channel but as a nice side effect, you'll also get a supported, working
G.729 implementation where you can get assistance with a simple phone call
instead of having to 'try so many modules'.

Howler Technologies also has a G.729 implementation but I have yet to
experience their product or support.

Tim Nelson
Systems/Network Support
Rockbochs Inc.
(218)727-4332 x105


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  _  




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