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lucabert at lucabert.de
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 Posted: Sun Jul 05, 2015 2:34 pm Post subject: [asterisk-users] Choosing codecs |
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Hi list!
I noticed that when the phone of my wife calls the gsm codec will be used,
but if someone calls the phone, alaw will be used:
00493511111111 calls 00493512222222:
OpenWrt*CLI> sip show channels
Peer User/ANR Call ID Format Hold Last Message Expiry Peer
192.168.200.11 00493512222222 5305ad0e07977dd 0x4 (ulaw) No Init: INVITE 0049351222
192.168.200.10 00493511111111 1481837b-c0a801 0x4 (ulaw) No Rx: INVITE 0049351111
00493512222222 calls 00493511111111:
OpenWrt*CLI> sip show channels
Peer User/ANR Call ID Format Hold Last Message Expiry Peer
192.168.200.10 00493511111111 5e21076a01b9483 0x8 (alaw) No Tx: ACK 0049351111
192.168.200.11 00493512222222 MCBsoNI2Cj266BB 0x2 (gsm) No Rx: ACK 0049351222
Could someone explain me why?
Second question: I think, ulaw/alaw are better then gsm, isn't it?
If so, how can I change it?
Thanks
Luca Bertoncello
(lucabert@lucabert.de)
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satskiy.a at gmail.com
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| Posted: Sun Jul 05, 2015 3:58 pm Post subject: [asterisk-users] Choosing codecs |
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Hi Luca
Y need to check your wifes codec priority list -seems to be GSM on the first place.
Luca Bertoncello <lucabert@lucabert.de> wrote:
| Quote: | Hi list!
I noticed that when the phone of my wife calls the gsm codec will be used,
but if someone calls the phone, alaw will be used:
00493511111111 calls 00493512222222:
OpenWrt*CLI> sip show channels
Peer User/ANR Call ID Format Hold Last Message Expiry Peer
192.168.200.11 00493512222222 5305ad0e07977dd 0x4 (ulaw) No Init: INVITE 0049351222
192.168.200.10 00493511111111 1481837b-c0a801 0x4 (ulaw) No Rx: INVITE 0049351111
00493512222222 calls 00493511111111:
OpenWrt*CLI> sip show channels
Peer User/ANR Call ID Format Hold Last Message Expiry Peer
192.168.200.10 00493511111111 5e21076a01b9483 0x8 (alaw) No Tx: ACK 0049351111
192.168.200.11 00493512222222 MCBsoNI2Cj266BB 0x2 (gsm) No Rx: ACK 0049351222
Could someone explain me why?
Second question: I think, ulaw/alaw are better then gsm, isn't it?
If so, how can I change it?
Thanks
Luca Bertoncello
(lucabert@lucabert.de)
--
_____________________________________________________________________
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asterisk_list at earth...
Guest
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| Posted: Mon Jul 06, 2015 2:48 am Post subject: [asterisk-users] Choosing codecs |
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On Sunday 05 Jul 2015, Luca Bertoncello wrote:
| Quote: | Hi list!
I noticed that when the phone of my wife calls the gsm codec will be used,
but if someone calls the phone, alaw will be used:
|
| Quote: | Could someone explain me why?
Second question: I think, ulaw/alaw are better then gsm, isn't it?
If so, how can I change it?
|
GSM is the native codec used for calls to mobile phones; it uses lossy
compression to achieve a low bit rate.
A-law is the native codec used by physical exchanges on the land line network
(PSTN and ISDN). It is non-lossy. It works by arranging the "steps" closer
together near the zero line, and further apart away from it; so the difference
between the actual signal and the nearest digital representation is small in
proportion to the signal.
To force the use of a-law, you need something like
disallow=all
allow=alaw
at the top of the configuration file for the calling technology in question
(sip.conf for SIP, chan_dahdi.conf for DAHDI, &c.). If you want to force a
specific device to use a specific codec, then put an allow in the section for
that device.
--
AJS
Note: Originating address only accepts e-mail from list! If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .
--
_____________________________________________________________________
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lucabert at lucabert.de
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| Posted: Mon Jul 06, 2015 2:56 am Post subject: [asterisk-users] Choosing codecs |
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Zitat von A J Stiles <asterisk_list@earthshod.co.uk>:
Hi,
| Quote: | GSM is the native codec used for calls to mobile phones; it uses lossy
compression to achieve a low bit rate.
A-law is the native codec used by physical exchanges on the land line network
(PSTN and ISDN). It is non-lossy. It works by arranging the "steps" closer
together near the zero line, and further apart away from it; so the
difference
between the actual signal and the nearest digital representation is small in
proportion to the signal.
|
Well, but for voice quality, which codec is better?
alaw or gsm?
Thanks
Luca Bertoncello
(lucabert@lucabert.de)
--
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asterisk_list at earth...
Guest
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| Posted: Mon Jul 06, 2015 3:05 am Post subject: [asterisk-users] Choosing codecs |
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On Monday 06 Jul 2015, Luca Bertoncello wrote:
| Quote: | Well, but for voice quality, which codec is better?
alaw or gsm?
|
A-law is better for voice quality (sorry, thought my original explanation was
obvious). But note that if the destination is a mobile phone, GSM will be
used anyway, at least for the link between the final cell tower and the
handset.
--
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
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lucabert at lucabert.de
Guest
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| Posted: Mon Jul 06, 2015 3:17 am Post subject: [asterisk-users] Choosing codecs |
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Zitat von A J Stiles <asterisk_list@earthshod.co.uk>:
| Quote: | On Monday 06 Jul 2015, Luca Bertoncello wrote:
| Quote: | Well, but for voice quality, which codec is better?
alaw or gsm?
|
A-law is better for voice quality (sorry, thought my original
explanation was
obvious). But note that if the destination is a mobile phone, GSM will be
used anyway, at least for the link between the final cell tower and the
handset.
|
OK, thank you...
Maybe will be your explanation other day but mondays obvious... 
So, I think, I should try to force the using of alaw for this phone,
is it right?
Usually we don't call mobile phones from our landline...
Thanks
Luca Bertoncello
(lucabert@lucabert.de)
--
_____________________________________________________________________
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New to Asterisk? Join us for a live introductory webinar every Thurs:
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asterisk_list at earth...
Guest
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| Posted: Mon Jul 06, 2015 3:39 am Post subject: [asterisk-users] Choosing codecs |
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On Monday 06 Jul 2015, Luca Bertoncello wrote:
| Quote: | So, I think, I should try to force the using of alaw for this phone,
is it right?
Usually we don't call mobile phones from our landline...
|
Yes. You should definitely be using A-law for calls to the Outside World.
If you use a different codec, then your telephone company will either transcode
it for you (if it is one they understand) or just block the call (if not).
Even if you are trying to use A-law to call a mobile phone, the transcoding to
GSM for the final leg to and from the handset will be taken care of by the
mobile company's equipment.
--
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:
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lucabert at lucabert.de
Guest
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| Posted: Mon Jul 06, 2015 3:41 am Post subject: [asterisk-users] Choosing codecs |
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Zitat von A J Stiles <asterisk_list@earthshod.co.uk>:
| Quote: | Yes. You should definitely be using A-law for calls to the Outside World.
If you use a different codec, then your telephone company will
either transcode
it for you (if it is one they understand) or just block the call (if not).
Even if you are trying to use A-law to call a mobile phone, the
transcoding to
GSM for the final leg to and from the handset will be taken care of by the
mobile company's equipment.
|
OK, I'll change the settings!
Thanks
Luca Bertoncello
(lucabert@lucabert.de)
--
_____________________________________________________________________
-- 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
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lucabert at lucabert.de
Guest
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| Posted: Mon Jul 06, 2015 3:58 am Post subject: [asterisk-users] Choosing codecs |
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Zitat von A J Stiles <asterisk_list@earthshod.co.uk>:
| Quote: | Yes. You should definitely be using A-law for calls to the Outside World.
|
Well, I wanted to change these settings, but I'm not sure, where I
have to do that...
I think in the users.conf, but I think, the "allow" keywords is for
the network...
How can I change this setting?
Thanks
Luca Bertoncello
(lucabert@lucabert.de)
--
_____________________________________________________________________
-- 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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asterisk_list at earth...
Guest
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| Posted: Mon Jul 06, 2015 4:56 am Post subject: [asterisk-users] Choosing codecs |
|
|
On Monday 06 Jul 2015, Luca Bertoncello wrote:
| Quote: | Zitat von A J Stiles <asterisk_list@earthshod.co.uk>:
| Quote: | Yes. You should definitely be using A-law for calls to the Outside
World.
|
Well, I wanted to change these settings, but I'm not sure, where I
have to do that...
I think in the users.conf, but I think, the "allow" keywords is for
the network...
How can I change this setting?
|
It will be in the /etc/asterisk/*.conf file for the appropriate calling
technology. So if the calls are going over a SIP trunk, it will be in
sip.conf . You want
disallow=all
allow=alaw
There probably will be some other allow= lines; just stick a semicolon in
front of the ones you do *not* want, to comment them out. Then issue
core reload
in Asterisk CLI, and all your calls should be A-law from now on.
--
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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lucabert at lucabert.de
Guest
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| Posted: Mon Jul 06, 2015 4:59 am Post subject: [asterisk-users] Choosing codecs |
|
|
Zitat von A J Stiles <asterisk_list@earthshod.co.uk>:
| Quote: | It will be in the /etc/asterisk/*.conf file for the appropriate calling
technology. So if the calls are going over a SIP trunk, it will be in
sip.conf . You want
disallow=all
allow=alaw
There probably will be some other allow= lines; just stick a semicolon in
front of the ones you do *not* want, to comment them out. Then issue
core reload
in Asterisk CLI, and all your calls should be A-law from now on.
|
OK, thanks a lot!
Luca Bertoncello
(lucabert@lucabert.de)
--
_____________________________________________________________________
-- 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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