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andrew at convergedgro...
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 Posted: Fri Nov 21, 2014 5:20 am Post subject: [asterisk-users] One way audio internal |
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Hi All
We have a strange issue with our hosted asterisk server running on Debian
Internal calls btween extensions using g729 give one way audio
As soon as we change the codec to ALAW the issues goes away.
Any ideas how to fix this?
Outbound calls via a trunk work fine with g729
Kind Regards
Andrew Colin
Converged Data (Pty) Ltd.
Licensed Telecoms Operator : (0258/IECS/JAN/09) (0258/IECNS/JAN/09)
Direct: +27 (0)10 591 4607
Mobile: +27 (0)82 310 3007
Switchboard: +27 (0)10 591 4600
Email: andrew@convergedgroup.net (andrew@convergedgroup.net)
Web: http://www.convergedgroup.net
75 Witkoppen Road, Northriding, Johannesburg, 2169
P O Box 7246, Weltevredenpark, 1715
This communication is confidential and intended solely for the addressee(s). Any unauthorized review, use, disclosure or distribution is prohibited. If you believe this message has been sent to you in error, please notify the sender by replying to this transmission and delete the message without disclosing it. Thank you.E-mail including attachments is susceptible to data corruption, interception, unauthorized amendment, tampering and viruses, and we only send and receive emails on the basis that we are not liable for any such corruption, interception, amendment, tampering or viruses or any consequences thereof. |
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asterisk_list at earth...
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| Posted: Fri Nov 21, 2014 5:47 am Post subject: [asterisk-users] One way audio internal |
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On Friday 21 Nov 2014, Andrew Colin wrote:
| Quote: | Hi All
We have a strange issue with our hosted asterisk server running on Debian
Internal calls btween extensions using g729 give one way audio
As soon as we change the codec to ALAW the issues goes away.
Any ideas how to fix this?
Outbound calls via a trunk work fine with g729
|
Unless you have serious bandwidth issues, just forget about g.729 and change
to a-law throughout. A-law is what the PSTN (in civilised countries) uses
anyway, so you won't need to transcode (which chews up processor resources
and risks compromising quality) for calls to and from the outside world.
If you really need to use g.729 and are outside the USA (therefore, beyond
the reach of software patents), there is a free version that you can use --
and this one, better than Digium's offering, comes with the Source Code so you
can be sure it isn't doing anything nasty behind the scenes.
But to be honest, you probably are better off just sticking with a-law.
--
AJS
Note: Originating address only accepts e-mail from list! If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .
--
_____________________________________________________________________
-- 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:
http://lists.digium.com/mailman/listinfo/asterisk-users |
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mitul at enterux.in
Guest
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| Posted: Fri Nov 21, 2014 5:50 am Post subject: [asterisk-users] One way audio internal |
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You probably do not have enough g729 channels license. On 21-Nov-2014 4:17 PM, "A J Stiles" <asterisk_list@earthshod.co.uk (asterisk_list@earthshod.co.uk)> wrote: | Quote: | On Friday 21 Nov 2014, Andrew Colin wrote:
| Quote: | Hi All
We have a strange issue with our hosted asterisk server running on Debian
Internal calls btween extensions using g729 give one way audio
As soon as we change the codec to ALAW the issues goes away.
Any ideas how to fix this?
Outbound calls via a trunk work fine with g729
|
Unless you have serious bandwidth issues, just forget about g.729 and change
to a-law throughout. A-law is what the PSTN (in civilised countries) uses
anyway, so you won't need to transcode (which chews up processor resources
and risks compromising quality) for calls to and from the outside world.
If you really need to use g.729 and are outside the USA (therefore, beyond
the reach of software patents), there is a free version that you can use --
and this one, better than Digium's offering, comes with the Source Code so you
can be sure it isn't doing anything nasty behind the scenes.
But to be honest, you probably are better off just sticking with a-law.
--
AJS
Note: Originating address only accepts e-mail from list! If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .
--
_____________________________________________________________________
-- 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:
http://lists.digium.com/mailman/listinfo/asterisk-users
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andrew at convergedgro...
Guest
|
| Posted: Fri Nov 21, 2014 5:57 am Post subject: [asterisk-users] One way audio internal |
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I am using the free g729
Kind Regards
Andrew Colin
Converged Data (Pty) Ltd.
Licensed Telecoms Operator : (0258/IECS/JAN/09) (0258/IECNS/JAN/09)
Direct: +27 (0)10 591 4607
Mobile: +27 (0)82 310 3007
Switchboard: +27 (0)10 591 4600
Email: andrew@convergedgroup.net (andrew@convergedgroup.net)
Web: http://www.convergedgroup.net
75 Witkoppen Road, Northriding, Johannesburg, 2169
P O Box 7246, Weltevredenpark, 1715
This communication is confidential and intended solely for the addressee(s). Any unauthorized review, use, disclosure or distribution is prohibited. If you believe this message has been sent to you in error, please notify the sender by replying to this transmission and delete the message without disclosing it. Thank you.E-mail including attachments is susceptible to data corruption, interception, unauthorized amendment, tampering and viruses, and we only send and receive emails on the basis that we are not liable for any such corruption, interception, amendment, tampering or viruses or any consequences thereof.
From: Mitul Limbani [mailto:mitul@enterux.in]
Sent: Friday, November 21, 2014 12:51 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Andrew Colin
Subject: Re: [asterisk-users] One way audio internal
You probably do not have enough g729 channels license.
On 21-Nov-2014 4:17 PM, "A J Stiles" <asterisk_list@earthshod.co.uk (asterisk_list@earthshod.co.uk)> wrote:
On Friday 21 Nov 2014, Andrew Colin wrote:
| Quote: | Hi All
We have a strange issue with our hosted asterisk server running on Debian
Internal calls btween extensions using g729 give one way audio
As soon as we change the codec to ALAW the issues goes away.
Any ideas how to fix this?
Outbound calls via a trunk work fine with g729
|
Unless you have serious bandwidth issues, just forget about g.729 and change
to a-law throughout. A-law is what the PSTN (in civilised countries) uses
anyway, so you won't need to transcode (which chews up processor resources
and risks compromising quality) for calls to and from the outside world.
If you really need to use g.729 and are outside the USA (therefore, beyond
the reach of software patents), there is a free version that you can use --
and this one, better than Digium's offering, comes with the Source Code so you
can be sure it isn't doing anything nasty behind the scenes.
But to be honest, you probably are better off just sticking with a-law.
--
AJS
Note: Originating address only accepts e-mail from list! If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .
--
_____________________________________________________________________
-- 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:
http://lists.digium.com/mailman/listinfo/asterisk-users |
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mitul at enterux.in
Guest
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| Posted: Fri Nov 21, 2014 6:05 am Post subject: [asterisk-users] One way audio internal |
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Then something to do with your codec selection priority. On 21-Nov-2014 4:26 PM, "Andrew Colin" <andrew@convergedgroup.net (andrew@convergedgroup.net)> wrote: | Quote: |
I am using the free g729
Kind Regards
Andrew Colin
Converged Data (Pty) Ltd.
Licensed Telecoms Operator : (0258/IECS/JAN/09) (0258/IECNS/JAN/09)
Direct: [url=tel:%2B27%20%280%2910%20591%204607]+27 (0)10 591 4607[/url]
Mobile: [url=tel:%2B27%20%280%2982%20310%203007]+27 (0)82 310 3007[/url]
Switchboard: [url=tel:%2B27%20%280%2910%20591%204600]+27 (0)10 591 4600[/url]
Email: andrew@convergedgroup.net (andrew@convergedgroup.net)
Web: http://www.convergedgroup.net
75 Witkoppen Road, Northriding, Johannesburg, 2169
P O Box 7246, Weltevredenpark, 1715
This communication is confidential and intended solely for the addressee(s). Any unauthorized review, use, disclosure or distribution is prohibited. If you believe this message has been sent to you in error, please notify the sender by replying to this transmission and delete the message without disclosing it. Thank you.E-mail including attachments is susceptible to data corruption, interception, unauthorized amendment, tampering and viruses, and we only send and receive emails on the basis that we are not liable for any such corruption, interception, amendment, tampering or viruses or any consequences thereof.
From: Mitul Limbani [mailto:mitul@enterux.in (mitul@enterux.in)]
Sent: Friday, November 21, 2014 12:51 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Andrew Colin
Subject: Re: [asterisk-users] One way audio internal
You probably do not have enough g729 channels license.
On 21-Nov-2014 4:17 PM, "A J Stiles" <asterisk_list@earthshod.co.uk (asterisk_list@earthshod.co.uk)> wrote:
On Friday 21 Nov 2014, Andrew Colin wrote:
| Quote: | Hi All
We have a strange issue with our hosted asterisk server running on Debian
Internal calls btween extensions using g729 give one way audio
As soon as we change the codec to ALAW the issues goes away.
Any ideas how to fix this?
Outbound calls via a trunk work fine with g729
|
Unless you have serious bandwidth issues, just forget about g.729 and change
to a-law throughout. A-law is what the PSTN (in civilised countries) uses
anyway, so you won't need to transcode (which chews up processor resources
and risks compromising quality) for calls to and from the outside world.
If you really need to use g.729 and are outside the USA (therefore, beyond
the reach of software patents), there is a free version that you can use --
and this one, better than Digium's offering, comes with the Source Code so you
can be sure it isn't doing anything nasty behind the scenes.
But to be honest, you probably are better off just sticking with a-law.
--
AJS
Note: Originating address only accepts e-mail from list! If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .
--
_____________________________________________________________________
-- 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:
http://lists.digium.com/mailman/listinfo/asterisk-users
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andrew at convergedgro...
Guest
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| Posted: Fri Nov 21, 2014 6:12 am Post subject: [asterisk-users] One way audio internal |
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I currently am running on a
Intel(R) Xeon(R) CPU E5-2670 v2 @ 2.50GHz
Codec im using is
codec_g729-ast18-icc-glibc-x86_64-core2.so |
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asterisk_list at earth...
Guest
|
| Posted: Fri Nov 21, 2014 6:50 am Post subject: [asterisk-users] One way audio internal |
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On Friday 21 Nov 2014, Andrew Colin wrote:
| Quote: | I am using the free g729
|
OK, so there shouldn't be any licencing problems (unless for some reason your
Asterisk is wanting to use the paid-for g.729 aot the Free one. Look at the
CLI output very, very carefully to see if this might be happening).
Did it ever work properly? If your kernel, C library or some other
fundamental system component has been updated since you installed g.729, then
it might have been broken by the upgrade. Navigating to the folder with the
Source Code and re-running `make` followed by `make install` ought to fix it.
But why are you using g.729 anyway? What special reason have you for doing it
differently than the rest of the world?
--
AJS
Note: Originating address only accepts e-mail from list! If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .
--
_____________________________________________________________________
-- 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:
http://lists.digium.com/mailman/listinfo/asterisk-users |
|
| Back to top |
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andrew at convergedgro...
Guest
|
| Posted: Fri Nov 21, 2014 9:14 am Post subject: [asterisk-users] One way audio internal |
|
|
I currently am running on a
Intel(R) Xeon(R) CPU E5-2670 v2 @ 2.50GHz
Codec im using is
codec_g729-ast18-icc-glibc-x86_64-core2.so
Kind Regards
Andrew Colin
Converged Data (Pty) Ltd.
Licensed Telecoms Operator : (0258/IECS/JAN/09) (0258/IECNS/JAN/09)
Direct: +27 (0)10 591 4607
Mobile: +27 (0)82 310 3007
Switchboard: +27 (0)10 591 4600
Email: andrew@convergedgroup.net (andrew@convergedgroup.net)
Web: http://www.convergedgroup.net
75 Witkoppen Road, Northriding, Johannesburg, 2169
P O Box 7246, Weltevredenpark, 1715
This communication is confidential and intended solely for the addressee(s). Any unauthorized review, use, disclosure or distribution is prohibited. If you believe this message has been sent to you in error, please notify the sender by replying to this transmission and delete the message without disclosing it. Thank you.E-mail including attachments is susceptible to data corruption, interception, unauthorized amendment, tampering and viruses, and we only send and receive emails on the basis that we are not liable for any such corruption, interception, amendment, tampering or viruses or any consequences thereof.
From: Mitul Limbani [mailto:mitul@enterux.in]
Sent: Friday, November 21, 2014 1:04 PM
To: Andrew Colin
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] One way audio internal
Then something to do with your codec selection priority.
On 21-Nov-2014 4:26 PM, "Andrew Colin" <andrew@convergedgroup.net (andrew@convergedgroup.net)> wrote:
I am using the free g729
Kind Regards
Andrew Colin
Converged Data (Pty) Ltd.
Licensed Telecoms Operator : (0258/IECS/JAN/09) (0258/IECNS/JAN/09)
Direct: [url=tel:%2B27%20%280%2910%20591%204607]+27 (0)10 591 4607[/url]
Mobile: [url=tel:%2B27%20%280%2982%20310%203007]+27 (0)82 310 3007[/url]
Switchboard: [url=tel:%2B27%20%280%2910%20591%204600]+27 (0)10 591 4600[/url]
Email: andrew@convergedgroup.net (andrew@convergedgroup.net)
Web: http://www.convergedgroup.net
75 Witkoppen Road, Northriding, Johannesburg, 2169
P O Box 7246, Weltevredenpark, 1715
This communication is confidential and intended solely for the addressee(s). Any unauthorized review, use, disclosure or distribution is prohibited. If you believe this message has been sent to you in error, please notify the sender by replying to this transmission and delete the message without disclosing it. Thank you.E-mail including attachments is susceptible to data corruption, interception, unauthorized amendment, tampering and viruses, and we only send and receive emails on the basis that we are not liable for any such corruption, interception, amendment, tampering or viruses or any consequences thereof.
From: Mitul Limbani [mailto:mitul@enterux.in (mitul@enterux.in)]
Sent: Friday, November 21, 2014 12:51 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Andrew Colin
Subject: Re: [asterisk-users] One way audio internal
You probably do not have enough g729 channels license.
On 21-Nov-2014 4:17 PM, "A J Stiles" <asterisk_list@earthshod.co.uk (asterisk_list@earthshod.co.uk)> wrote:
On Friday 21 Nov 2014, Andrew Colin wrote:
| Quote: | Hi All
We have a strange issue with our hosted asterisk server running on Debian
Internal calls btween extensions using g729 give one way audio
As soon as we change the codec to ALAW the issues goes away.
Any ideas how to fix this?
Outbound calls via a trunk work fine with g729
|
Unless you have serious bandwidth issues, just forget about g.729 and change
to a-law throughout. A-law is what the PSTN (in civilised countries) uses
anyway, so you won't need to transcode (which chews up processor resources
and risks compromising quality) for calls to and from the outside world.
If you really need to use g.729 and are outside the USA (therefore, beyond
the reach of software patents), there is a free version that you can use --
and this one, better than Digium's offering, comes with the Source Code so you
can be sure it isn't doing anything nasty behind the scenes.
But to be honest, you probably are better off just sticking with a-law.
--
AJS
Note: Originating address only accepts e-mail from list! If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .
--
_____________________________________________________________________
-- 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:
http://lists.digium.com/mailman/listinfo/asterisk-users |
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