[asterisk-users] Can't get G.729 to work...

Andres andres at telesip.net
Wed Dec 16 09:43:22 CST 2009 

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
>bens at rolandschorr.com
>
>
>  
>
>>-----Original Message-----
>>From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-
>>bounces at 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
>>>bens at rolandschorr.com
>>>
>>>
>>>      
>>>
>>>>-----Original Message-----
>>>>From: asterisk-users-bounces at lists.digium.com
>>>>        
>>>>
>[mailto:asterisk-users-
>  
>
>>>>bounces at 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-bounces at lists.digium.com
>>>>>[mailto:asterisk-users-bounces at 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
>>>>>bens at rolandschorr.com
>>>>>
>>>>>
>>>>>          
>>>>>
>>>>>>-----Original Message-----
>>>>>>From: asterisk-users-bounces at lists.digium.com
>>>>>>            
>>>>>>
>>>[mailto:asterisk-users-
>>>      
>>>
>>>>>>bounces at 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
>>>>>>>bens at rolandschorr.com
>>>>>>>
>>>>>>>
>>>>>>>              
>>>>>>>
>>>>>>>>-----Original Message-----
>>>>>>>>From: asterisk-users-bounces at lists.digium.com
>>>>>>>>                
>>>>>>>>
>>>>>[mailto:asterisk-users-
>>>>>          
>>>>>
>>>>>>>>bounces at lists.digium.com] On Behalf Of jeff at 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
>>>>>>>>                
>>>>>>>>
>>_______________________________________________
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>>    
>>
>
>_______________________________________________
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>
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>  
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