Re: [asterisk-users] Can't get G.729 to work...
On Tue, Dec 15, 2009 at 07:53:47PM +, Jeff LaCoursiere wrote: On Tue, 15 Dec 2009, Ben Schorr wrote: O.K., interestingly enough when I call our extensions from my mobile phone it still seems to be using ULAW, but when they dial out it seems to be using G.729 now. Is there something in Dahdi that I need to configure so that inbound calls (from the PRI on a Digium TE205) use G.729 to get to the phones too? A Dahdi channel over a PRI will always be ulaw - that is the encoding on the PRI (at least in the US). For the record: in most other parts of the world it is actually alaw. But anyway, both ulaw and alaw have very similar charasteristics, so basically a 's/ulaw/alaw/' or 's/ulaw/ulaw,alaw/' on on config snippets in this thread would generally work if you want to apply them to the AS :-) -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can't get G.729 to work...
Ben Schorr wrote:
O.K., I restored the Allow=ulaw in the sip_general_additional.conf file,
then I found the individual extension settings in the
sip_additional.conf file and I added
I would not go editing the individual files if you are using FreePBX.
As soon as you make a change in the web interface it will override any
manual changes you made.
Simply do it in the web interface for each extension. You do have a
parameter called allow and another called disallow in the web interface
when editing the extension (its under device options). Use them. For
multiple entries just separate them with a comma.
Andres
http://www.neuroredes.com
disallow=all
allow=g729
to each of the extensions at the remote site. Then I did a SIP RELOAD.
So we'll see how that goes.
Thanks again for the assist - this has been quite an education.
Ben M. Schorr
Chief Executive Officer
__
Roland Schorr Tower
www.rolandschorr.com
b...@rolandschorr.com
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Dave Fullerton
Sent: Tuesday, December 15, 2009 11:47 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Can't get G.729 to work...
I don't know how FreePBX works, but with vanilla Asterisk you would do
something like this with your sip.conf:
[general]
disallow=all
allow=ulaw
allow=g729
[localA]
callerid=Local phone A 100
username=localA
secret=blahblah1
[localB]
callerid=Local phone B 101
username=localB
secret=blah1blah
[remoteA]
callerid=Remote phone A 102
disallow=all
allow=g729
username=remoteA
secret=123456
[remoteB]
callerid=Remote phone B 103
disallow=all
allow=g729
username=remoteB
secret=654321
You can do this using templates as well, but this will make it easier
to
understand. See the disallow/allow lines on the remote peers? Those
override the settings in the general portion of your sip.conf. With
these
settings the local phones will use ulaw by default and allow g729 when
needed.
This will do what you want for the most part. Local phones will use
ulaw for all
calls between themselves and calls in and out of the PRI. Calls from a
remote
phone to a local phone will use g.729 end to end. Calls from a local
phone to a
remote phone will use ulaw between the local phone and asterisk and
g.729
between asterisk and the remote phone (this is a limitation of
asterisk's
codec negotiation). Calls from remote phones will use g.729 all the
time.
I'm sure there is a way to do this through the freepbx gui, but like I
said, I
have no experience with freepbx.
-Dave
Ben Schorr wrote:
O.K., I think I'm catching on. I only have a single SIP.CONF file
that ALL of the extensions are using so I'm gathering that I need to
set up a separate SIP.CONF file (or perhaps just an included file)
for
the 8 users at the remote office which ONLY Allows the G.729.
So now I'm figuring out how to do that.
Ben M. Schorr
Chief Executive Officer
__
Roland Schorr Tower
www.rolandschorr.com
b...@rolandschorr.com
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Dave Fullerton
Sent: Tuesday, December 15, 2009 11:05 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Can't get G.729 to work...
That's a bit misleading. Yes calls that travel over a PRI will be
using ulaw, but
only over the PRI leg of the call. The SIP leg can still be using
G.729 with
asterisk transcoding between the two legs.
Ben, You haven't shown us the contents of your sip.conf file for
the
peers
you are working on but I have a guess as to what is going on. In
one
of your
previous messages you state: I moved G.729 to the top of that list
(just
below disallow) I'm guessing your list looks something like this:
disallow=all
allow=g729
allow=ulaw
allow={maybe something else}
This will be fine for all the phones in the office but the remote
phones need
to ONLY have disallow=all and allow=g729 in their entries in
sip.conf
as Jeff's
reply stated. By having the allow=ulaw entry in there you are
giving
asterisk
permission to allow any call that is already in the ulaw format
(calls
from the
PRI) to remain in that format when contacting your remote phones.
If
you're
still stick post your sip.conf (with the passwords removed) and we
can
help
you out.
-Dave
Danny Nicholas wrote:
IMO you can only use the G.729 on a SIP call. If the call falls
onto the
PRI
Re: [asterisk-users] Can't get G.729 to work...
Really? I didn't see them in the web interface; which is why I turned
to editing the files. I'll check the web interface again, perhaps I
simply missed them.
Best wishes and aloha,
Ben M. Schorr
Chief Executive Officer
__
Roland Schorr Tower
www.rolandschorr.com
b...@rolandschorr.com
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Andres
Sent: Wednesday, December 16, 2009 5:43 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Can't get G.729 to work...
Ben Schorr wrote:
O.K., I restored the Allow=ulaw in the sip_general_additional.conf
file, then I found the individual extension settings in the
sip_additional.conf file and I added
I would not go editing the individual files if you are using FreePBX.
As soon as you make a change in the web interface it will override any
manual changes you made.
Simply do it in the web interface for each extension. You do have a
parameter called allow and another called disallow in the web
interface when
editing the extension (its under device options). Use them. For
multiple
entries just separate them with a comma.
Andres
http://www.neuroredes.com
disallow=all
allow=g729
to each of the extensions at the remote site. Then I did a SIP
RELOAD.
So we'll see how that goes.
Thanks again for the assist - this has been quite an education.
Ben M. Schorr
Chief Executive Officer
__
Roland Schorr Tower
www.rolandschorr.com
b...@rolandschorr.com
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Dave Fullerton
Sent: Tuesday, December 15, 2009 11:47 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Can't get G.729 to work...
I don't know how FreePBX works, but with vanilla Asterisk you would
do
something like this with your sip.conf:
[general]
disallow=all
allow=ulaw
allow=g729
[localA]
callerid=Local phone A 100
username=localA
secret=blahblah1
[localB]
callerid=Local phone B 101
username=localB
secret=blah1blah
[remoteA]
callerid=Remote phone A 102
disallow=all
allow=g729
username=remoteA
secret=123456
[remoteB]
callerid=Remote phone B 103
disallow=all
allow=g729
username=remoteB
secret=654321
You can do this using templates as well, but this will make it
easier
to
understand. See the disallow/allow lines on the remote peers? Those
override the settings in the general portion of your sip.conf. With
these
settings the local phones will use ulaw by default and allow g729
when
needed.
This will do what you want for the most part. Local phones will use
ulaw for all
calls between themselves and calls in and out of the PRI. Calls from
a
remote
phone to a local phone will use g.729 end to end. Calls from a local
phone to a
remote phone will use ulaw between the local phone and asterisk and
g.729
between asterisk and the remote phone (this is a limitation of
asterisk's
codec negotiation). Calls from remote phones will use g.729 all the
time.
I'm sure there is a way to do this through the freepbx gui, but like
I
said, I
have no experience with freepbx.
-Dave
Ben Schorr wrote:
O.K., I think I'm catching on. I only have a single SIP.CONF file
that ALL of the extensions are using so I'm gathering that I need
to
set up a separate SIP.CONF file (or perhaps just an included file)
for
the 8 users at the remote office which ONLY Allows the G.729.
So now I'm figuring out how to do that.
Ben M. Schorr
Chief Executive Officer
__
Roland Schorr Tower
www.rolandschorr.com
b...@rolandschorr.com
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Dave Fullerton
Sent: Tuesday, December 15, 2009 11:05 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Can't get G.729 to work...
That's a bit misleading. Yes calls that travel over a PRI will be
using ulaw, but
only over the PRI leg of the call. The SIP leg can still be using
G.729 with
asterisk transcoding between the two legs.
Ben, You haven't shown us the contents of your sip.conf file for
the
peers
you are working on but I have a guess as to what is going on. In
one
of your
previous messages you state: I moved G.729 to the top of that
list
(just
below disallow) I'm guessing your list looks something like this:
disallow=all
allow=g729
allow=ulaw
allow={maybe something else}
This will be fine for all the phones
[asterisk-users] Can't get G.729 to work...
Asterisk 1.4 - FreePBX - Polycom 330 and 501 phones. I've got G.729 loaded in the modules on the Asterisk server and on the Polycom phones I've set G.729 to be the first preference of codec, but still when I go SIP SHOW CHANNELS during active calls it still shows (ULAW) (G.711) as the codec in use. I'm a newbie at Asterisk, can anybody suggest what I might be overlooking? Best wishes and aloha, Ben M. Schorr Chief Executive Officer __ Roland Schorr Tower 1155 Fort Street Mall Honolulu, Hawaii 96813 Mobile: 808-782-6306 Fax: 808-533-3677 www.rolandschorr.com http://www.rolandschorr.com/ b...@rolandschorr.com mailto:b...@rolandschorr.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can't get G.729 to work...
On Tue, 15 Dec 2009, Ben Schorr wrote: Asterisk 1.4 - FreePBX - Polycom 330 and 501 phones. I've got G.729 loaded in the modules on the Asterisk server and on the Polycom phones I've set G.729 to be the first preference of codec, but still when I go SIP SHOW CHANNELS during active calls it still shows (ULAW) (G.711) as the codec in use. I'm a newbie at Asterisk, can anybody suggest what I might be overlooking? In the sip.conf entry for your peer you need to specify the codec negotiation order. Though you put g.729 first on the phone, asterisk probably has ulaw first, and is taking precedence. In the sip.conf entry put this: disallow=all allow=g729 j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can't get G.729 to work...
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Ben Schorr wrote: I’ve got G.729 loaded in the modules on the Asterisk server and on the Polycom phones I’ve set G.729 to be the first preference of codec, but still when I go SIP SHOW CHANNELS during active calls it still shows “(ULAW)” (G.711) as the codec in use. How about in sip.conf? Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFLJ9csCFu3bIiwtTARAnDfAJ9QL8xqGZYgeHyFwhX7Ebz+h7UVYQCdEt5k af0y9vqC4WV8CmdAN0D0ASE= =++1O -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can't get G.729 to work...
On Tue, 15 Dec 2009, Ben Schorr wrote: Ahhh...yes, I think that may have been it. I moved G.729 to the top of that list (just below disallow) and now I have a restart when convenient pending. Is that sufficient or do I have to actually reboot the server for the change to take effect? Just do a sip reload at the asterisk CLI prompt and you will be good to go. It won't cutoff any calls in progress. Then reboot your phone. Cheers, j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can't get G.729 to work...
Ahhh...yes, I think that may have been it. I moved G.729 to the top of that list (just below disallow) and now I have a restart when convenient pending. Is that sufficient or do I have to actually reboot the server for the change to take effect? Best wishes and aloha, Ben M. Schorr Chief Executive Officer __ Roland Schorr Tower www.rolandschorr.com b...@rolandschorr.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: Tuesday, December 15, 2009 8:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Can't get G.729 to work... On Tue, 15 Dec 2009, Ben Schorr wrote: Asterisk 1.4 - FreePBX - Polycom 330 and 501 phones. I've got G.729 loaded in the modules on the Asterisk server and on the Polycom phones I've set G.729 to be the first preference of codec, but still when I go SIP SHOW CHANNELS during active calls it still shows (ULAW) (G.711) as the codec in use. I'm a newbie at Asterisk, can anybody suggest what I might be overlooking? In the sip.conf entry for your peer you need to specify the codec negotiation order. Though you put g.729 first on the phone, asterisk probably has ulaw first, and is taking precedence. In the sip.conf entry put this: disallow=all allow=g729 j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can't get G.729 to work...
You should only need a reboot for DAHDI changes (not always then...) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ben Schorr Sent: Tuesday, December 15, 2009 1:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Can't get G.729 to work... Ahhh...yes, I think that may have been it. I moved G.729 to the top of that list (just below disallow) and now I have a restart when convenient pending. Is that sufficient or do I have to actually reboot the server for the change to take effect? Best wishes and aloha, Ben M. Schorr Chief Executive Officer __ Roland Schorr Tower www.rolandschorr.com b...@rolandschorr.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: Tuesday, December 15, 2009 8:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Can't get G.729 to work... On Tue, 15 Dec 2009, Ben Schorr wrote: Asterisk 1.4 - FreePBX - Polycom 330 and 501 phones. I've got G.729 loaded in the modules on the Asterisk server and on the Polycom phones I've set G.729 to be the first preference of codec, but still when I go SIP SHOW CHANNELS during active calls it still shows (ULAW) (G.711) as the codec in use. I'm a newbie at Asterisk, can anybody suggest what I might be overlooking? In the sip.conf entry for your peer you need to specify the codec negotiation order. Though you put g.729 first on the phone, asterisk probably has ulaw first, and is taking precedence. In the sip.conf entry put this: disallow=all allow=g729 j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can't get G.729 to work...
O.K., thanks, I'm catching on (slowly). Waiting for the next call to see if the SIP.CONF change did the trick. Ben M. Schorr Chief Executive Officer __ Roland Schorr Tower www.rolandschorr.com b...@rolandschorr.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Tuesday, December 15, 2009 9:15 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Can't get G.729 to work... You should only need a reboot for DAHDI changes (not always then...) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ben Schorr Sent: Tuesday, December 15, 2009 1:08 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Can't get G.729 to work... Ahhh...yes, I think that may have been it. I moved G.729 to the top of that list (just below disallow) and now I have a restart when convenient pending. Is that sufficient or do I have to actually reboot the server for the change to take effect? Best wishes and aloha, Ben M. Schorr Chief Executive Officer __ Roland Schorr Tower www.rolandschorr.com b...@rolandschorr.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: Tuesday, December 15, 2009 8:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Can't get G.729 to work... On Tue, 15 Dec 2009, Ben Schorr wrote: Asterisk 1.4 - FreePBX - Polycom 330 and 501 phones. I've got G.729 loaded in the modules on the Asterisk server and on the Polycom phones I've set G.729 to be the first preference of codec, but still when I go SIP SHOW CHANNELS during active calls it still shows (ULAW) (G.711) as the codec in use. I'm a newbie at Asterisk, can anybody suggest what I might be overlooking? In the sip.conf entry for your peer you need to specify the codec negotiation order. Though you put g.729 first on the phone, asterisk probably has ulaw first, and is taking precedence. In the sip.conf entry put this: disallow=all allow=g729 j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can't get G.729 to work...
O.K., interestingly enough when I call our extensions from my mobile phone it still seems to be using ULAW, but when they dial out it seems to be using G.729 now. Is there something in Dahdi that I need to configure so that inbound calls (from the PRI on a Digium TE205) use G.729 to get to the phones too? Ben M. Schorr Chief Executive Officer __ Roland Schorr Tower www.rolandschorr.com b...@rolandschorr.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of j...@jeff.net Sent: Tuesday, December 15, 2009 9:13 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Can't get G.729 to work... On Tue, 15 Dec 2009, Ben Schorr wrote: Ahhh...yes, I think that may have been it. I moved G.729 to the top of that list (just below disallow) and now I have a restart when convenient pending. Is that sufficient or do I have to actually reboot the server for the change to take effect? Just do a sip reload at the asterisk CLI prompt and you will be good to go. It won't cutoff any calls in progress. Then reboot your phone. Cheers, j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can't get G.729 to work...
On Tue, 15 Dec 2009, Ben Schorr wrote: O.K., interestingly enough when I call our extensions from my mobile phone it still seems to be using ULAW, but when they dial out it seems to be using G.729 now. Is there something in Dahdi that I need to configure so that inbound calls (from the PRI on a Digium TE205) use G.729 to get to the phones too? A Dahdi channel over a PRI will always be ulaw - that is the encoding on the PRI (at least in the US). If your phones are using G.729 then transcoding will be taking place within asterisk for the bridge between the channels. My guess is you are looking at the PRI channel. There should be another channel for the phone. That should always be G.729 now. Cheers, j Ben M. Schorr Chief Executive Officer __ Roland Schorr Tower www.rolandschorr.com b...@rolandschorr.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of j...@jeff.net Sent: Tuesday, December 15, 2009 9:13 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Can't get G.729 to work... On Tue, 15 Dec 2009, Ben Schorr wrote: Ahhh...yes, I think that may have been it. I moved G.729 to the top of that list (just below disallow) and now I have a restart when convenient pending. Is that sufficient or do I have to actually reboot the server for the change to take effect? Just do a sip reload at the asterisk CLI prompt and you will be good to go. It won't cutoff any calls in progress. Then reboot your phone. Cheers, j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can't get G.729 to work...
Sorry, I think I may have misspoke... What I'm hoping for is that all of the connections between my phones (or at least a particular group of them) and my Asterisk server will use G.729. Currently it seems like it usually is, but not always, and I haven't figured out the pattern. All of our calls fall into two categories: Internal calls - one extension to another within our single Asterisk server org. External calls - To/From one of our extensions out thru the PRI line to our carrier (Hawaiian Tel) to phone numbers out in the world. For some reason it appears that inbound calls from out in the world are going to our phones using ULAW, but outbound calls to the world are using G.729. That's progress but...how can I get my Asterisk server to use G.729 to pass those incoming calls to my phones? Best wishes and aloha, Ben M. Schorr Chief Executive Officer __ Roland Schorr Tower www.rolandschorr.com b...@rolandschorr.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: Tuesday, December 15, 2009 9:54 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Can't get G.729 to work... On Tue, 15 Dec 2009, Ben Schorr wrote: O.K., interestingly enough when I call our extensions from my mobile phone it still seems to be using ULAW, but when they dial out it seems to be using G.729 now. Is there something in Dahdi that I need to configure so that inbound calls (from the PRI on a Digium TE205) use G.729 to get to the phones too? A Dahdi channel over a PRI will always be ulaw - that is the encoding on the PRI (at least in the US). If your phones are using G.729 then transcoding will be taking place within asterisk for the bridge between the channels. My guess is you are looking at the PRI channel. There should be another channel for the phone. That should always be G.729 now. Cheers, j Ben M. Schorr Chief Executive Officer __ Roland Schorr Tower www.rolandschorr.com b...@rolandschorr.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of j...@jeff.net Sent: Tuesday, December 15, 2009 9:13 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Can't get G.729 to work... On Tue, 15 Dec 2009, Ben Schorr wrote: Ahhh...yes, I think that may have been it. I moved G.729 to the top of that list (just below disallow) and now I have a restart when convenient pending. Is that sufficient or do I have to actually reboot the server for the change to take effect? Just do a sip reload at the asterisk CLI prompt and you will be good to go. It won't cutoff any calls in progress. Then reboot your phone. Cheers, j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can't get G.729 to work...
IMO you can only use the G.729 on a SIP call. If the call falls onto the PRI framework, ulaw will be forced. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ben Schorr Sent: Tuesday, December 15, 2009 2:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Can't get G.729 to work... Sorry, I think I may have misspoke... What I'm hoping for is that all of the connections between my phones (or at least a particular group of them) and my Asterisk server will use G.729. Currently it seems like it usually is, but not always, and I haven't figured out the pattern. All of our calls fall into two categories: Internal calls - one extension to another within our single Asterisk server org. External calls - To/From one of our extensions out thru the PRI line to our carrier (Hawaiian Tel) to phone numbers out in the world. For some reason it appears that inbound calls from out in the world are going to our phones using ULAW, but outbound calls to the world are using G.729. That's progress but...how can I get my Asterisk server to use G.729 to pass those incoming calls to my phones? Best wishes and aloha, Ben M. Schorr Chief Executive Officer __ Roland Schorr Tower www.rolandschorr.com b...@rolandschorr.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: Tuesday, December 15, 2009 9:54 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Can't get G.729 to work... On Tue, 15 Dec 2009, Ben Schorr wrote: O.K., interestingly enough when I call our extensions from my mobile phone it still seems to be using ULAW, but when they dial out it seems to be using G.729 now. Is there something in Dahdi that I need to configure so that inbound calls (from the PRI on a Digium TE205) use G.729 to get to the phones too? A Dahdi channel over a PRI will always be ulaw - that is the encoding on the PRI (at least in the US). If your phones are using G.729 then transcoding will be taking place within asterisk for the bridge between the channels. My guess is you are looking at the PRI channel. There should be another channel for the phone. That should always be G.729 now. Cheers, j Ben M. Schorr Chief Executive Officer __ Roland Schorr Tower www.rolandschorr.com b...@rolandschorr.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of j...@jeff.net Sent: Tuesday, December 15, 2009 9:13 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Can't get G.729 to work... On Tue, 15 Dec 2009, Ben Schorr wrote: Ahhh...yes, I think that may have been it. I moved G.729 to the top of that list (just below disallow) and now I have a restart when convenient pending. Is that sufficient or do I have to actually reboot the server for the change to take effect? Just do a sip reload at the asterisk CLI prompt and you will be good to go. It won't cutoff any calls in progress. Then reboot your phone. Cheers, j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can't get G.729 to work...
Oh, dear. So my users with less-than-ideal bandwidth are stuck with drop-outs and poor sound quality because they can't use the reduced bandwidth codec for those calls? :-( They've been complaining that they often end up on a call where one or both parties are cutting in and out. Unfortunately it's only this one remote site, with about 8 users, who connect across a VPN to the site where the server is. We've tried increasing their bandwidth and tweaking the QOS settings on their firewalls but so far we haven't been able to solve it. I was hoping that switching to a lower bandwidth CODEC would give them the call reliability they need. If not then I guess I'm back to the drawing board, with increasingly impatient users, trying to troubleshoot their call quality issues. Ben M. Schorr Chief Executive Officer __ Roland Schorr Tower www.rolandschorr.com b...@rolandschorr.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Tuesday, December 15, 2009 10:19 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Can't get G.729 to work... IMO you can only use the G.729 on a SIP call. If the call falls onto the PRI framework, ulaw will be forced. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ben Schorr Sent: Tuesday, December 15, 2009 2:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Can't get G.729 to work... Sorry, I think I may have misspoke... What I'm hoping for is that all of the connections between my phones (or at least a particular group of them) and my Asterisk server will use G.729. Currently it seems like it usually is, but not always, and I haven't figured out the pattern. All of our calls fall into two categories: Internal calls - one extension to another within our single Asterisk server org. External calls - To/From one of our extensions out thru the PRI line to our carrier (Hawaiian Tel) to phone numbers out in the world. For some reason it appears that inbound calls from out in the world are going to our phones using ULAW, but outbound calls to the world are using G.729. That's progress but...how can I get my Asterisk server to use G.729 to pass those incoming calls to my phones? Best wishes and aloha, Ben M. Schorr Chief Executive Officer __ Roland Schorr Tower www.rolandschorr.com b...@rolandschorr.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: Tuesday, December 15, 2009 9:54 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Can't get G.729 to work... On Tue, 15 Dec 2009, Ben Schorr wrote: O.K., interestingly enough when I call our extensions from my mobile phone it still seems to be using ULAW, but when they dial out it seems to be using G.729 now. Is there something in Dahdi that I need to configure so that inbound calls (from the PRI on a Digium TE205) use G.729 to get to the phones too? A Dahdi channel over a PRI will always be ulaw - that is the encoding on the PRI (at least in the US). If your phones are using G.729 then transcoding will be taking place within asterisk for the bridge between the channels. My guess is you are looking at the PRI channel. There should be another channel for the phone. That should always be G.729 now. Cheers, j Ben M. Schorr Chief Executive Officer __ Roland Schorr Tower www.rolandschorr.com b...@rolandschorr.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of j...@jeff.net Sent: Tuesday, December 15, 2009 9:13 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Can't get G.729 to work... On Tue, 15 Dec 2009, Ben Schorr wrote: Ahhh...yes, I think that may have been it. I moved G.729 to the top of that list (just below disallow) and now I have a restart when convenient pending. Is that sufficient or do I have to actually reboot the server for the change to take effect? Just do a sip reload at the asterisk CLI prompt and you will be good to go. It won't cutoff any calls in progress. Then reboot your phone. Cheers, j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit
Re: [asterisk-users] Can't get G.729 to work...
Why not restrict these 8 users to a SIP provider like (but not) bandwidth.com? By eliminating the PRI element, you should completely resolve the problem. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ben Schorr Sent: Tuesday, December 15, 2009 2:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Can't get G.729 to work... Oh, dear. So my users with less-than-ideal bandwidth are stuck with drop-outs and poor sound quality because they can't use the reduced bandwidth codec for those calls? :-( They've been complaining that they often end up on a call where one or both parties are cutting in and out. Unfortunately it's only this one remote site, with about 8 users, who connect across a VPN to the site where the server is. We've tried increasing their bandwidth and tweaking the QOS settings on their firewalls but so far we haven't been able to solve it. I was hoping that switching to a lower bandwidth CODEC would give them the call reliability they need. If not then I guess I'm back to the drawing board, with increasingly impatient users, trying to troubleshoot their call quality issues. Ben M. Schorr Chief Executive Officer __ Roland Schorr Tower www.rolandschorr.com b...@rolandschorr.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Tuesday, December 15, 2009 10:19 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Can't get G.729 to work... IMO you can only use the G.729 on a SIP call. If the call falls onto the PRI framework, ulaw will be forced. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ben Schorr Sent: Tuesday, December 15, 2009 2:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Can't get G.729 to work... Sorry, I think I may have misspoke... What I'm hoping for is that all of the connections between my phones (or at least a particular group of them) and my Asterisk server will use G.729. Currently it seems like it usually is, but not always, and I haven't figured out the pattern. All of our calls fall into two categories: Internal calls - one extension to another within our single Asterisk server org. External calls - To/From one of our extensions out thru the PRI line to our carrier (Hawaiian Tel) to phone numbers out in the world. For some reason it appears that inbound calls from out in the world are going to our phones using ULAW, but outbound calls to the world are using G.729. That's progress but...how can I get my Asterisk server to use G.729 to pass those incoming calls to my phones? Best wishes and aloha, Ben M. Schorr Chief Executive Officer __ Roland Schorr Tower www.rolandschorr.com b...@rolandschorr.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: Tuesday, December 15, 2009 9:54 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Can't get G.729 to work... On Tue, 15 Dec 2009, Ben Schorr wrote: O.K., interestingly enough when I call our extensions from my mobile phone it still seems to be using ULAW, but when they dial out it seems to be using G.729 now. Is there something in Dahdi that I need to configure so that inbound calls (from the PRI on a Digium TE205) use G.729 to get to the phones too? A Dahdi channel over a PRI will always be ulaw - that is the encoding on the PRI (at least in the US). If your phones are using G.729 then transcoding will be taking place within asterisk for the bridge between the channels. My guess is you are looking at the PRI channel. There should be another channel for the phone. That should always be G.729 now. Cheers, j Ben M. Schorr Chief Executive Officer __ Roland Schorr Tower www.rolandschorr.com b...@rolandschorr.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of j...@jeff.net Sent: Tuesday, December 15, 2009 9:13 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Can't get G.729 to work... On Tue, 15 Dec 2009, Ben Schorr wrote: Ahhh...yes, I think that may have been it. I moved G.729 to the top of that list (just below disallow) and now I have a restart when convenient pending. Is that sufficient or do I
Re: [asterisk-users] Can't get G.729 to work...
- Ben Schorr b...@rolandschorr.com wrote: Oh, dear. So my users with less-than-ideal bandwidth are stuck with drop-outs and poor sound quality because they can't use the reduced bandwidth codec for those calls? :-( They've been complaining that they often end up on a call where one or both parties are cutting in and out. Unfortunately it's only this one remote site, with about 8 users, who connect across a VPN to the site where the server is. We've tried increasing their bandwidth and tweaking the QOS settings on their firewalls but so far we haven't been able to solve it. I was hoping that switching to a lower bandwidth CODEC would give them the call reliability they need. If not then I guess I'm back to the drawing board, with increasingly impatient users, trying to troubleshoot their call quality issues. You need to install the G.729a codec on your system so that it will transcode your calls from ulaw (on your PRI side) to g729 (on your SIP side). Keep in mind that G.729 is a patented codec which requires licensing. The two companies offering G.729 for Asterisk(that I know of, please correct me if there are others :-) ) are here: http://store.digium.com/productview.php?category_id=5product_code=8G729CODEC http://www.howlertech.com/products/howlets/ I've always used the Digium G.729 and it has worked flawlessly. I've also heard good things about Howler G.729 but never used it personally. Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can't get G.729 to work...
Well, I know I still have a LOT to learn about Asterisk but...how will they get their incoming phone calls from their DIDs (which the TelCo sends to their PRI) if I move the remote office onto a SIP provider? The PRI doesn't seem to cause any problem for the majority of the users (at the home site) it's just the 8 users at the remote site who are complaining of quality issues. Ben M. Schorr Chief Executive Officer __ Roland Schorr Tower www.rolandschorr.com b...@rolandschorr.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Tuesday, December 15, 2009 10:31 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Can't get G.729 to work... Why not restrict these 8 users to a SIP provider like (but not) bandwidth.com? By eliminating the PRI element, you should completely resolve the problem. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ben Schorr Sent: Tuesday, December 15, 2009 2:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Can't get G.729 to work... Oh, dear. So my users with less-than-ideal bandwidth are stuck with drop- outs and poor sound quality because they can't use the reduced bandwidth codec for those calls? :-( They've been complaining that they often end up on a call where one or both parties are cutting in and out. Unfortunately it's only this one remote site, with about 8 users, who connect across a VPN to the site where the server is. We've tried increasing their bandwidth and tweaking the QOS settings on their firewalls but so far we haven't been able to solve it. I was hoping that switching to a lower bandwidth CODEC would give them the call reliability they need. If not then I guess I'm back to the drawing board, with increasingly impatient users, trying to troubleshoot their call quality issues. Ben M. Schorr Chief Executive Officer __ Roland Schorr Tower www.rolandschorr.com b...@rolandschorr.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Tuesday, December 15, 2009 10:19 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Can't get G.729 to work... IMO you can only use the G.729 on a SIP call. If the call falls onto the PRI framework, ulaw will be forced. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ben Schorr Sent: Tuesday, December 15, 2009 2:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Can't get G.729 to work... Sorry, I think I may have misspoke... What I'm hoping for is that all of the connections between my phones (or at least a particular group of them) and my Asterisk server will use G.729. Currently it seems like it usually is, but not always, and I haven't figured out the pattern. All of our calls fall into two categories: Internal calls - one extension to another within our single Asterisk server org. External calls - To/From one of our extensions out thru the PRI line to our carrier (Hawaiian Tel) to phone numbers out in the world. For some reason it appears that inbound calls from out in the world are going to our phones using ULAW, but outbound calls to the world are using G.729. That's progress but...how can I get my Asterisk server to use G.729 to pass those incoming calls to my phones? Best wishes and aloha, Ben M. Schorr Chief Executive Officer __ Roland Schorr Tower www.rolandschorr.com b...@rolandschorr.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: Tuesday, December 15, 2009 9:54 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Can't get G.729 to work... On Tue, 15 Dec 2009, Ben Schorr wrote: O.K., interestingly enough when I call our extensions from my mobile phone it still seems to be using ULAW, but when they dial out it seems to be using G.729 now. Is there something in Dahdi that I need to configure so that inbound calls (from the PRI on a Digium TE205) use G.729 to get to the phones too? A Dahdi channel over a PRI will always be ulaw - that is the encoding on the PRI (at least in the US). If your phones are using G.729 then transcoding will be taking place
Re: [asterisk-users] Can't get G.729 to work...
I thought I already did that - which is how they now get some (but not yet all) of their calls on G.729. scratching head Ben M. Schorr Chief Executive Officer __ Roland Schorr Tower www.rolandschorr.com b...@rolandschorr.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Tim Nelson Sent: Tuesday, December 15, 2009 10:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Can't get G.729 to work... - Ben Schorr b...@rolandschorr.com wrote: Oh, dear. So my users with less-than-ideal bandwidth are stuck with drop-outs and poor sound quality because they can't use the reduced bandwidth codec for those calls? :-( They've been complaining that they often end up on a call where one or both parties are cutting in and out. Unfortunately it's only this one remote site, with about 8 users, who connect across a VPN to the site where the server is. We've tried increasing their bandwidth and tweaking the QOS settings on their firewalls but so far we haven't been able to solve it. I was hoping that switching to a lower bandwidth CODEC would give them the call reliability they need. If not then I guess I'm back to the drawing board, with increasingly impatient users, trying to troubleshoot their call quality issues. You need to install the G.729a codec on your system so that it will transcode your calls from ulaw (on your PRI side) to g729 (on your SIP side). Keep in mind that G.729 is a patented codec which requires licensing. The two companies offering G.729 for Asterisk(that I know of, please correct me if there are others :-) ) are here: http://store.digium.com/productview.php?category_id=5product_code=8 G729CODEC http://www.howlertech.com/products/howlets/ I've always used the Digium G.729 and it has worked flawlessly. I've also heard good things about Howler G.729 but never used it personally. Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can't get G.729 to work...
Do your routers allow giving these users maximum priority? What is the effective bandwidth on the VPN connection? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ben Schorr Sent: Tuesday, December 15, 2009 2:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Can't get G.729 to work... I thought I already did that - which is how they now get some (but not yet all) of their calls on G.729. scratching head Ben M. Schorr Chief Executive Officer __ Roland Schorr Tower www.rolandschorr.com b...@rolandschorr.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Tim Nelson Sent: Tuesday, December 15, 2009 10:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Can't get G.729 to work... - Ben Schorr b...@rolandschorr.com wrote: Oh, dear. So my users with less-than-ideal bandwidth are stuck with drop-outs and poor sound quality because they can't use the reduced bandwidth codec for those calls? :-( They've been complaining that they often end up on a call where one or both parties are cutting in and out. Unfortunately it's only this one remote site, with about 8 users, who connect across a VPN to the site where the server is. We've tried increasing their bandwidth and tweaking the QOS settings on their firewalls but so far we haven't been able to solve it. I was hoping that switching to a lower bandwidth CODEC would give them the call reliability they need. If not then I guess I'm back to the drawing board, with increasingly impatient users, trying to troubleshoot their call quality issues. You need to install the G.729a codec on your system so that it will transcode your calls from ulaw (on your PRI side) to g729 (on your SIP side). Keep in mind that G.729 is a patented codec which requires licensing. The two companies offering G.729 for Asterisk(that I know of, please correct me if there are others :-) ) are here: http://store.digium.com/productview.php?category_id=5product_code=8 G729CODEC http://www.howlertech.com/products/howlets/ I've always used the Digium G.729 and it has worked flawlessly. I've also heard good things about Howler G.729 but never used it personally. Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can't get G.729 to work...
Watch the calls on the console. Try both ways. Document what you see and your codec settings on both the phone, and sip.conf. You may have to tell the phone that the only codec it can use is G.729, don't just make that first choice. Make it the only choice. Cary Fitch -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ben Schorr Sent: Tuesday, December 15, 2009 2:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Can't get G.729 to work... I thought I already did that - which is how they now get some (but not yet all) of their calls on G.729. scratching head Ben M. Schorr Chief Executive Officer __ Roland Schorr Tower www.rolandschorr.com b...@rolandschorr.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can't get G.729 to work...
And tell Asterisk that G.729 is the only codec for that number as well! Cary Fitch ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can't get G.729 to work...
Yes, the routers are another issue we're dealing with. We've configured them to prioritize traffic to/from our Asterisk server but I'm not convinced that setting is really working as expected. So we're working with the vendor on that. The effective bandwidth is 6Mb/sec down x 1.4Mb/sec up (so basically 1.4x1.4 on the VPN). For 8 users, where rarely more than 2-3 of them are on the phone at any given time, that should be sufficient I think. They DO have to share the connection with their web browsing and e-mail and such but as best we've been able to tell they aren't saturating their connections - usually not more than 4-5 of the 8 are using their computers at any given time and most of them just do e-mail and local apps that shouldn't touch the Internet connection. Frankly I'm puzzled that they have these issues and the problems rarely seem to happen when I call them. I'll go to their site and make a few calls from one of their phones and it sounds perfect to me. But three days later all I hear is how frustrated they are because these new VOIP phones suck and they can never hear anybody and... sigh Ben M. Schorr Chief Executive Officer __ Roland Schorr Tower www.rolandschorr.com b...@rolandschorr.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Tuesday, December 15, 2009 10:54 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Can't get G.729 to work... Do your routers allow giving these users maximum priority? What is the effective bandwidth on the VPN connection? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ben Schorr Sent: Tuesday, December 15, 2009 2:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Can't get G.729 to work... I thought I already did that - which is how they now get some (but not yet all) of their calls on G.729. scratching head Ben M. Schorr Chief Executive Officer __ Roland Schorr Tower www.rolandschorr.com b...@rolandschorr.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Tim Nelson Sent: Tuesday, December 15, 2009 10:29 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Can't get G.729 to work... - Ben Schorr b...@rolandschorr.com wrote: Oh, dear. So my users with less-than-ideal bandwidth are stuck with drop-outs and poor sound quality because they can't use the reduced bandwidth codec for those calls? :-( They've been complaining that they often end up on a call where one or both parties are cutting in and out. Unfortunately it's only this one remote site, with about 8 users, who connect across a VPN to the site where the server is. We've tried increasing their bandwidth and tweaking the QOS settings on their firewalls but so far we haven't been able to solve it. I was hoping that switching to a lower bandwidth CODEC would give them the call reliability they need. If not then I guess I'm back to the drawing board, with increasingly impatient users, trying to troubleshoot their call quality issues. You need to install the G.729a codec on your system so that it will transcode your calls from ulaw (on your PRI side) to g729 (on your SIP side). Keep in mind that G.729 is a patented codec which requires licensing. The two companies offering G.729 for Asterisk(that I know of, please correct me if there are others :-) ) are here: http://store.digium.com/productview.php?category_id=5product_code=8 G729CODEC http://www.howlertech.com/products/howlets/ I've always used the Digium G.729 and it has worked flawlessly. I've also heard good things about Howler G.729 but never used it personally. Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can't get G.729 to work...
- Ben Schorr b...@rolandschorr.com wrote: I thought I already did that - which is how they now get some (but not yet all) of their calls on G.729. scratching head VERY SIMPLIFIED VIEW Allowing G.729 in your configurations (disallow=all, allow=g729) enables those endpoints to use that codec *THROUGH* the Asterisk system to other endpoints who support that codec. The PRI coming into the system can only use a single codec which is ulaw in your case (some locations use alaw). So, since both endpoints cannot agree on g729, they fall back to a codec they both know, ulaw. Now, if you had G.729 codec licensed, Asterisk would simply take the ulaw audio from the PRI, transcode it to G.729, and pass it to your phone allowing the phone to operate in G.729 for these calls also. /VERY SIMPLIFIED VIEW The actual underlying processes that determine how all of this happens is much more complex, specifically the SDP/SIP negotiations that occur. Tim Nelson Systems/Network Support Rockbochs Inc. (218)727-4332 x105 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can't get G.729 to work...
That's a bit misleading. Yes calls that travel over a PRI will be using
ulaw, but only over the PRI leg of the call. The SIP leg can still be
using G.729 with asterisk transcoding between the two legs.
Ben, You haven't shown us the contents of your sip.conf file for the
peers you are working on but I have a guess as to what is going on. In
one of your previous messages you state: I moved G.729 to the top of
that list (just below disallow) I'm guessing your list looks something
like this:
disallow=all
allow=g729
allow=ulaw
allow={maybe something else}
This will be fine for all the phones in the office but the remote phones
need to ONLY have disallow=all and allow=g729 in their entries in
sip.conf as Jeff's reply stated. By having the allow=ulaw entry in there
you are giving asterisk permission to allow any call that is already in
the ulaw format (calls from the PRI) to remain in that format when
contacting your remote phones. If you're still stick post your sip.conf
(with the passwords removed) and we can help you out.
-Dave
Danny Nicholas wrote:
IMO you can only use the G.729 on a SIP call. If the call falls onto the
PRI framework, ulaw will be forced.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ben Schorr
Sent: Tuesday, December 15, 2009 2:11 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Can't get G.729 to work...
Sorry, I think I may have misspoke...
What I'm hoping for is that all of the connections between my phones (or
at least a particular group of them) and my Asterisk server will use
G.729. Currently it seems like it usually is, but not always, and I
haven't figured out the pattern.
All of our calls fall into two categories:
Internal calls - one extension to another within our single Asterisk
server org.
External calls - To/From one of our extensions out thru the PRI line to
our carrier (Hawaiian Tel) to phone numbers out in the world.
For some reason it appears that inbound calls from out in the world are
going to our phones using ULAW, but outbound calls to the world are
using G.729.
That's progress but...how can I get my Asterisk server to use G.729 to
pass those incoming calls to my phones?
Best wishes and aloha,
Ben M. Schorr
Chief Executive Officer
__
Roland Schorr Tower
www.rolandschorr.com
b...@rolandschorr.com
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere
Sent: Tuesday, December 15, 2009 9:54 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Can't get G.729 to work...
On Tue, 15 Dec 2009, Ben Schorr wrote:
O.K., interestingly enough when I call our extensions from my mobile
phone it still seems to be using ULAW, but when they dial out it
seems
to be using G.729 now.
Is there something in Dahdi that I need to configure so that inbound
calls (from the PRI on a Digium TE205) use G.729 to get to the
phones
too?
A Dahdi channel over a PRI will always be ulaw - that is the encoding
on the
PRI (at least in the US). If your phones are using G.729 then
transcoding will
be taking place within asterisk for the bridge between the channels.
My guess is you are looking at the PRI channel. There should be
another
channel for the phone. That should always be G.729 now.
Cheers,
j
Ben M. Schorr
Chief Executive Officer
__
Roland Schorr Tower
www.rolandschorr.com
b...@rolandschorr.com
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of j...@jeff.net
Sent: Tuesday, December 15, 2009 9:13 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Can't get G.729 to work...
On Tue, 15 Dec 2009, Ben Schorr wrote:
Ahhh...yes, I think that may have been it. I moved G.729 to the
top
of that list (just below disallow) and now I have a restart when
convenient pending. Is that sufficient or do I have to actually
reboot the server for the change to take effect?
Just do a sip reload at the asterisk CLI prompt and you will be
good
to go. It
won't cutoff any calls in progress. Then reboot your phone.
Cheers,
j
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can't get G.729 to work...
O.K., I think I'm catching on. I only have a single SIP.CONF file that
ALL of the extensions are using so I'm gathering that I need to set up a
separate SIP.CONF file (or perhaps just an included file) for the 8
users at the remote office which ONLY Allows the G.729.
So now I'm figuring out how to do that.
Ben M. Schorr
Chief Executive Officer
__
Roland Schorr Tower
www.rolandschorr.com
b...@rolandschorr.com
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Dave Fullerton
Sent: Tuesday, December 15, 2009 11:05 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Can't get G.729 to work...
That's a bit misleading. Yes calls that travel over a PRI will be
using ulaw, but
only over the PRI leg of the call. The SIP leg can still be using
G.729 with
asterisk transcoding between the two legs.
Ben, You haven't shown us the contents of your sip.conf file for the
peers
you are working on but I have a guess as to what is going on. In one
of your
previous messages you state: I moved G.729 to the top of that list
(just
below disallow) I'm guessing your list looks something like this:
disallow=all
allow=g729
allow=ulaw
allow={maybe something else}
This will be fine for all the phones in the office but the remote
phones need
to ONLY have disallow=all and allow=g729 in their entries in sip.conf
as Jeff's
reply stated. By having the allow=ulaw entry in there you are giving
asterisk
permission to allow any call that is already in the ulaw format (calls
from the
PRI) to remain in that format when contacting your remote phones. If
you're
still stick post your sip.conf (with the passwords removed) and we can
help
you out.
-Dave
Danny Nicholas wrote:
IMO you can only use the G.729 on a SIP call. If the call falls
onto the
PRI framework, ulaw will be forced.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ben
Schorr
Sent: Tuesday, December 15, 2009 2:11 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Can't get G.729 to work...
Sorry, I think I may have misspoke...
What I'm hoping for is that all of the connections between my phones
(or
at least a particular group of them) and my Asterisk server will use
G.729. Currently it seems like it usually is, but not always, and I
haven't figured out the pattern.
All of our calls fall into two categories:
Internal calls - one extension to another within our single Asterisk
server org.
External calls - To/From one of our extensions out thru the PRI line
to
our carrier (Hawaiian Tel) to phone numbers out in the world.
For some reason it appears that inbound calls from out in the world
are
going to our phones using ULAW, but outbound calls to the world are
using G.729.
That's progress but...how can I get my Asterisk server to use G.729
to
pass those incoming calls to my phones?
Best wishes and aloha,
Ben M. Schorr
Chief Executive Officer
__
Roland Schorr Tower
www.rolandschorr.com
b...@rolandschorr.com
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere
Sent: Tuesday, December 15, 2009 9:54 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Can't get G.729 to work...
On Tue, 15 Dec 2009, Ben Schorr wrote:
O.K., interestingly enough when I call our extensions from my
mobile
phone it still seems to be using ULAW, but when they dial out it
seems
to be using G.729 now.
Is there something in Dahdi that I need to configure so that
inbound
calls (from the PRI on a Digium TE205) use G.729 to get to the
phones
too?
A Dahdi channel over a PRI will always be ulaw - that is the
encoding
on the
PRI (at least in the US). If your phones are using G.729 then
transcoding will
be taking place within asterisk for the bridge between the
channels.
My guess is you are looking at the PRI channel. There should be
another
channel for the phone. That should always be G.729 now.
Cheers,
j
Ben M. Schorr
Chief Executive Officer
__
Roland Schorr Tower
www.rolandschorr.com
b...@rolandschorr.com
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of j...@jeff.net
Sent: Tuesday, December 15, 2009 9:13 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Can't get G.729 to work...
On Tue, 15 Dec 2009, Ben Schorr wrote:
Ahhh...yes, I think
Re: [asterisk-users] Can't get G.729 to work...
O.K., so for now (as a test) I just commented out the allow=ULAW line
in the SIP.conf (actually it's sip_general_additional.conf on this
FreePBX box) and that does seem to be forcing all traffic to G.729.
I think ultimately I'd like to let the local users use ULAW because it
seems to sound better and just force the 8 remote users to use G.729,
but for now I can live with this while I figure out how to do that.
Thanks for all of your help - and I welcome any additional pointers
you'd like to offer.
Ben M. Schorr
Chief Executive Officer
__
Roland Schorr Tower
www.rolandschorr.com
b...@rolandschorr.com
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Ben Schorr
Sent: Tuesday, December 15, 2009 11:16 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Can't get G.729 to work...
O.K., I think I'm catching on. I only have a single SIP.CONF file
that ALL of the
extensions are using so I'm gathering that I need to set up a separate
SIP.CONF file (or perhaps just an included file) for the 8 users at
the remote
office which ONLY Allows the G.729.
So now I'm figuring out how to do that.
Ben M. Schorr
Chief Executive Officer
__
Roland Schorr Tower
www.rolandschorr.com
b...@rolandschorr.com
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Dave Fullerton
Sent: Tuesday, December 15, 2009 11:05 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Can't get G.729 to work...
That's a bit misleading. Yes calls that travel over a PRI will be
using ulaw, but
only over the PRI leg of the call. The SIP leg can still be using
G.729 with
asterisk transcoding between the two legs.
Ben, You haven't shown us the contents of your sip.conf file for the
peers
you are working on but I have a guess as to what is going on. In one
of your
previous messages you state: I moved G.729 to the top of that list
(just
below disallow) I'm guessing your list looks something like this:
disallow=all
allow=g729
allow=ulaw
allow={maybe something else}
This will be fine for all the phones in the office but the remote
phones need
to ONLY have disallow=all and allow=g729 in their entries in
sip.conf
as Jeff's
reply stated. By having the allow=ulaw entry in there you are giving
asterisk
permission to allow any call that is already in the ulaw format
(calls
from the
PRI) to remain in that format when contacting your remote phones. If
you're
still stick post your sip.conf (with the passwords removed) and we
can
help
you out.
-Dave
Danny Nicholas wrote:
IMO you can only use the G.729 on a SIP call. If the call falls
onto the
PRI framework, ulaw will be forced.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ben
Schorr
Sent: Tuesday, December 15, 2009 2:11 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Can't get G.729 to work...
Sorry, I think I may have misspoke...
What I'm hoping for is that all of the connections between my
phones
(or
at least a particular group of them) and my Asterisk server will
use
G.729. Currently it seems like it usually is, but not always, and
I
haven't figured out the pattern.
All of our calls fall into two categories:
Internal calls - one extension to another within our single
Asterisk
server org.
External calls - To/From one of our extensions out thru the PRI
line
to
our carrier (Hawaiian Tel) to phone numbers out in the world.
For some reason it appears that inbound calls from out in the
world
are
going to our phones using ULAW, but outbound calls to the world
are
using G.729.
That's progress but...how can I get my Asterisk server to use
G.729
to
pass those incoming calls to my phones?
Best wishes and aloha,
Ben M. Schorr
Chief Executive Officer
__
Roland Schorr Tower
www.rolandschorr.com
b...@rolandschorr.com
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere
Sent: Tuesday, December 15, 2009 9:54 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Can't get G.729 to work...
On Tue, 15 Dec 2009, Ben Schorr wrote:
O.K., interestingly enough when I call our extensions from my
mobile
phone it still seems to be using ULAW, but when they dial out it
seems
to be using G.729 now.
Is there something
Re: [asterisk-users] Can't get G.729 to work...
I don't know how FreePBX works, but with vanilla Asterisk you would do
something like this with your sip.conf:
[general]
disallow=all
allow=ulaw
allow=g729
[localA]
callerid=Local phone A 100
username=localA
secret=blahblah1
[localB]
callerid=Local phone B 101
username=localB
secret=blah1blah
[remoteA]
callerid=Remote phone A 102
disallow=all
allow=g729
username=remoteA
secret=123456
[remoteB]
callerid=Remote phone B 103
disallow=all
allow=g729
username=remoteB
secret=654321
You can do this using templates as well, but this will make it easier to
understand. See the disallow/allow lines on the remote peers? Those
override the settings in the general portion of your sip.conf. With
these settings the local phones will use ulaw by default and allow g729
when needed.
This will do what you want for the most part. Local phones will use ulaw
for all calls between themselves and calls in and out of the PRI. Calls
from a remote phone to a local phone will use g.729 end to end. Calls
from a local phone to a remote phone will use ulaw between the local
phone and asterisk and g.729 between asterisk and the remote phone (this
is a limitation of asterisk's codec negotiation). Calls from remote
phones will use g.729 all the time.
I'm sure there is a way to do this through the freepbx gui, but like I
said, I have no experience with freepbx.
-Dave
Ben Schorr wrote:
O.K., I think I'm catching on. I only have a single SIP.CONF file that
ALL of the extensions are using so I'm gathering that I need to set up a
separate SIP.CONF file (or perhaps just an included file) for the 8
users at the remote office which ONLY Allows the G.729.
So now I'm figuring out how to do that.
Ben M. Schorr
Chief Executive Officer
__
Roland Schorr Tower
www.rolandschorr.com
b...@rolandschorr.com
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Dave Fullerton
Sent: Tuesday, December 15, 2009 11:05 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Can't get G.729 to work...
That's a bit misleading. Yes calls that travel over a PRI will be
using ulaw, but
only over the PRI leg of the call. The SIP leg can still be using
G.729 with
asterisk transcoding between the two legs.
Ben, You haven't shown us the contents of your sip.conf file for the
peers
you are working on but I have a guess as to what is going on. In one
of your
previous messages you state: I moved G.729 to the top of that list
(just
below disallow) I'm guessing your list looks something like this:
disallow=all
allow=g729
allow=ulaw
allow={maybe something else}
This will be fine for all the phones in the office but the remote
phones need
to ONLY have disallow=all and allow=g729 in their entries in sip.conf
as Jeff's
reply stated. By having the allow=ulaw entry in there you are giving
asterisk
permission to allow any call that is already in the ulaw format (calls
from the
PRI) to remain in that format when contacting your remote phones. If
you're
still stick post your sip.conf (with the passwords removed) and we can
help
you out.
-Dave
Danny Nicholas wrote:
IMO you can only use the G.729 on a SIP call. If the call falls
onto the
PRI framework, ulaw will be forced.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ben
Schorr
Sent: Tuesday, December 15, 2009 2:11 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Can't get G.729 to work...
Sorry, I think I may have misspoke...
What I'm hoping for is that all of the connections between my phones
(or
at least a particular group of them) and my Asterisk server will use
G.729. Currently it seems like it usually is, but not always, and I
haven't figured out the pattern.
All of our calls fall into two categories:
Internal calls - one extension to another within our single Asterisk
server org.
External calls - To/From one of our extensions out thru the PRI line
to
our carrier (Hawaiian Tel) to phone numbers out in the world.
For some reason it appears that inbound calls from out in the world
are
going to our phones using ULAW, but outbound calls to the world are
using G.729.
That's progress but...how can I get my Asterisk server to use G.729
to
pass those incoming calls to my phones?
Best wishes and aloha,
Ben M. Schorr
Chief Executive Officer
__
Roland Schorr Tower
www.rolandschorr.com
b...@rolandschorr.com
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere
Sent: Tuesday, December 15, 2009 9:54 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re
Re: [asterisk-users] Can't get G.729 to work...
At 12:45 PM 12/15/2009, you wrote: The PRI doesn't seem to cause any problem for the majority of the users (at the home site) it's just the 8 users at the remote site who are complaining of quality issues. So out of curiosity, if you were to limit the phone usage for an hour at the remote site to a max of 2 calls at a time, does the voice quality problem go away? If so, how many calls does it take to make it come back? That test will tell you if it's a bandwidth issue and how much of a bandwidth issue it is. Ira ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can't get G.729 to work...
O.K., I restored the Allow=ulaw in the sip_general_additional.conf file,
then I found the individual extension settings in the
sip_additional.conf file and I added
disallow=all
allow=g729
to each of the extensions at the remote site. Then I did a SIP RELOAD.
So we'll see how that goes.
Thanks again for the assist - this has been quite an education.
Ben M. Schorr
Chief Executive Officer
__
Roland Schorr Tower
www.rolandschorr.com
b...@rolandschorr.com
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Dave Fullerton
Sent: Tuesday, December 15, 2009 11:47 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Can't get G.729 to work...
I don't know how FreePBX works, but with vanilla Asterisk you would do
something like this with your sip.conf:
[general]
disallow=all
allow=ulaw
allow=g729
[localA]
callerid=Local phone A 100
username=localA
secret=blahblah1
[localB]
callerid=Local phone B 101
username=localB
secret=blah1blah
[remoteA]
callerid=Remote phone A 102
disallow=all
allow=g729
username=remoteA
secret=123456
[remoteB]
callerid=Remote phone B 103
disallow=all
allow=g729
username=remoteB
secret=654321
You can do this using templates as well, but this will make it easier
to
understand. See the disallow/allow lines on the remote peers? Those
override the settings in the general portion of your sip.conf. With
these
settings the local phones will use ulaw by default and allow g729 when
needed.
This will do what you want for the most part. Local phones will use
ulaw for all
calls between themselves and calls in and out of the PRI. Calls from a
remote
phone to a local phone will use g.729 end to end. Calls from a local
phone to a
remote phone will use ulaw between the local phone and asterisk and
g.729
between asterisk and the remote phone (this is a limitation of
asterisk's
codec negotiation). Calls from remote phones will use g.729 all the
time.
I'm sure there is a way to do this through the freepbx gui, but like I
said, I
have no experience with freepbx.
-Dave
Ben Schorr wrote:
O.K., I think I'm catching on. I only have a single SIP.CONF file
that ALL of the extensions are using so I'm gathering that I need to
set up a separate SIP.CONF file (or perhaps just an included file)
for
the 8 users at the remote office which ONLY Allows the G.729.
So now I'm figuring out how to do that.
Ben M. Schorr
Chief Executive Officer
__
Roland Schorr Tower
www.rolandschorr.com
b...@rolandschorr.com
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Dave Fullerton
Sent: Tuesday, December 15, 2009 11:05 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Can't get G.729 to work...
That's a bit misleading. Yes calls that travel over a PRI will be
using ulaw, but
only over the PRI leg of the call. The SIP leg can still be using
G.729 with
asterisk transcoding between the two legs.
Ben, You haven't shown us the contents of your sip.conf file for
the
peers
you are working on but I have a guess as to what is going on. In
one
of your
previous messages you state: I moved G.729 to the top of that list
(just
below disallow) I'm guessing your list looks something like this:
disallow=all
allow=g729
allow=ulaw
allow={maybe something else}
This will be fine for all the phones in the office but the remote
phones need
to ONLY have disallow=all and allow=g729 in their entries in
sip.conf
as Jeff's
reply stated. By having the allow=ulaw entry in there you are
giving
asterisk
permission to allow any call that is already in the ulaw format
(calls
from the
PRI) to remain in that format when contacting your remote phones.
If
you're
still stick post your sip.conf (with the passwords removed) and we
can
help
you out.
-Dave
Danny Nicholas wrote:
IMO you can only use the G.729 on a SIP call. If the call falls
onto the
PRI framework, ulaw will be forced.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ben
Schorr
Sent: Tuesday, December 15, 2009 2:11 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Can't get G.729 to work...
Sorry, I think I may have misspoke...
What I'm hoping for is that all of the connections between my
phones
(or
at least a particular group of them) and my Asterisk server will
use
G.729. Currently it seems like it usually is, but not always, and
I
haven't figured out the pattern.
All of our calls fall into two categories
