thr3ads.net - asterisk users - [Asterisk-Users] :: Migrating to 1.0.3 => Attention.

If this information is useful, please help other people find it:
Share via:

Jefferson Carvalho

2004-Dec-07 16:56 UTC

[Asterisk-Users] :: Migrating to 1.0.3 => Attention. ::

Hello list ,

I?d like to announce possible problems with migrating any version prior to
1.0.2 to 1.0.3.

Pay attention :

1. Codecs

Codecs names/description have been changed .

For example :

versions <= 1.0.2

voip*CLI> show codecs

Disclaimer: this command is for informational purposes only.
        It does not indicate anything about your configuration.
          1 (1 <<  0)  G.723.1
          2 (1 <<  1)  GSM
          4 (1 <<  2)  G.711 u-law
          8 (1 <<  3)  G.711 A-law
         16 (1 <<  4)  G.726
         32 (1 <<  5)  ADPCM
         64 (1 <<  6)  16 bit Signed Linear PCM
        128 (1 <<  7)  LPC10
        256 (1 <<  8)  G.729A audio
        512 (1 <<  9)  SpeeX
       1024 (1 << 10)  iLBC
      65536 (1 << 16)  JPEG image
     131072 (1 << 17)  PNG image
     262144 (1 << 18)  H.261 Video
     524288 (1 << 19)  H.263 Video

versions > = 1.0.2

bacural*CLI> show codecs
Disclaimer: this command is for informational purposes only.
        It does not indicate anything about your configuration.
        INT    BINARY        HEX   TYPE    NAME   DESC
--------------------------------------------------------------------------------
          1 (1 <<  0)      (0x1)  audio    g723   (G.723.1)
          2 (1 <<  1)      (0x2)  audio     gsm   (GSM)
          4 (1 <<  2)      (0x4)  audio    ulaw   (G.711 u-law)
          8 (1 <<  3)      (0x8)  audio    alaw   (G.711 A-law)
         16 (1 <<  4)     (0x10)  audio    g726   (G.726)
         32 (1 <<  5)     (0x20)  audio   adpcm   (ADPCM)
         64 (1 <<  6)     (0x40)  audio    slin   (16 bit Signed Linear
PCM)
        128 (1 <<  7)     (0x80)  audio   lpc10   (LPC10)
        256 (1 <<  8)    (0x100)  audio    g729   (G.729A)
        512 (1 <<  9)    (0x200)  audio   speex   (SpeeX)
       1024 (1 << 10)    (0x400)  audio    ilbc   (iLBC)
      65536 (1 << 16)  (0x10000)  image    jpeg   (JPEG image)
     131072 (1 << 17)  (0x20000)  image     png   (PNG image)
     262144 (1 << 18)  (0x40000)  video    h261   (H.261 Video)
     524288 (1 << 19)  (0x80000)  video    h263   (H.263 Video)

For example :

If u used

allow=G723.1 , now u have to use : allow=g723 *  ( NO CAPS ) !!!
allow=G729 , now u have to use : allow=g729 * ( NO CAPS ) !!!
allow=iLBC , now u have to use : allow=ilbc *  ( NO CAPS ) !!!

Please , verify all configurations files ( sip.conf , iax.conf ) before 
upgrade.
If u see some message like , codec error , is it.

Good look ,

-Jefferson Carvalho
 Jeff Networks

Brian West

2004-Dec-07 18:56 UTC

head link

[Asterisk-Users] :: Migrating to 1.0.3 => Attention. ::

> If u used
> 
> allow=G723.1 , now u have to use : allow=g723 *  ( NO CAPS ) !!!
> allow=G729 , now u have to use : allow=g729 * ( NO CAPS ) !!!
> allow=iLBC , now u have to use : allow=ilbc *  ( NO CAPS ) !!!
> 
> Please , verify all configurations files ( sip.conf , iax.conf ) before
> upgrade.
> If u see some message like , codec error , is it.
It maybe a bug on this part.

allow=g723 
And 
allow=g723.1 

BOTH should work now.  Case should not matter as it never did in the past.

We have a codec alias table in frame.c

} ast_codec_alias_table[] = {
    {"slinear","slin"},
    {"g723.1","g723"},
};

bkw

Altus Snyman

2004-Dec-08 02:57 UTC

head link

[Asterisk-Users] :: Migrating to 1.0.3 => Attention. ::

I got a new problem
We have Grandstream bt 100 using ilbc and mitel 5055 using ulaw
,internal or external, if you try and transfer using the transfer 
buttons it does not work and will just send it to the voicemail of the 
user,or whatever the next step is
But,if you add the ,tT to the Dial app and use the # key transfer works??

Dave Cotton wrote:
>On Tue, 2004-12-07 at 20:46 -0700, Kevin P. Fleming wrote:
>
>  
>
>>Yes, those patches had two major effects:
>>
>>- "disallow=all" in a SIP peer/user entry now actually works
the way it
>>should have worked all along; it clears the list of available codecs 
>>that may have been inherited from the global list
>>
>>- allow= now allows for multiple entries on a single line, and also 
>>remembers the order you specified them, so they are presented to the 
>>peer/user in the same order
>>
>>    
>>
>
>There also seemed problems in CVS-HEAD it broke case I had iLBC and it
>took ages to start *, changed it to ilbc and everything went back to
>normal speed.
>
>  
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20041208/dccc13ac/attachment.htm

dean collins

2004-Dec-08 13:04 UTC

head link

[Asterisk-Users] :: Migrating to 1.0.3 => Attention. ::

I'll increase my bounty offer from $25 to $100 for this.


Cheers,
Dean


-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Brian West
Sent: Wednesday, December 08, 2004 2:36 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] :: Migrating to 1.0.3 => Attention. ::

http://www.voip-info.org/wiki-Asterisk+bounty+SIP+simultaneous+registry

Anyone wanna push up the bounty on this?  I'm sure we could pull this
rabbit
out of our hats.

bkw
> -----Original Message-----
> From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-
> bounces@lists.digium.com] On Behalf Of Brian West
> Sent: Wednesday, December 08, 2004 12:58 PM
> To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
> Subject: RE: [Asterisk-Users] :: Migrating to 1.0.3 => Attention. ::
> 
> w00t... what to pull out of our hats next?
> 
> bkw
> 
> > -----Original Message-----
> > From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-> > bounces@lists.digium.com] On Behalf Of Kevin P. Fleming
> > Sent: Wednesday, December 08, 2004 10:34 AM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [Asterisk-Users] :: Migrating to 1.0.3 => Attention.
::
> >
> > Brian West wrote:
> > > Not to mention that allow=all works now too.  Not to mention per
> > user/peer
> > > codec prefs.
> >
> > Well, I did mention the new prefs support, but you're right, I
forgot> > about "allow=all" now working properly as well.
> >
> > Basically, it now works the way it was always expected to work :-)
> > Thanks to Brian and Anthony for their efforts!
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users@lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:
> >    http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
_______________________________________________
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

Seemingly Similar Threads

Search for more apparently analagous threads

asterisk users - Dec 2004 - :: Migrating to 1.0.3 => Attention. ::

[Asterisk-Users] :: Migrating to 1.0.3 => Attention. ::

[Asterisk-Users] :: Migrating to 1.0.3 => Attention. ::

[Asterisk-Users] :: Migrating to 1.0.3 => Attention. ::

[Asterisk-Users] :: Migrating to 1.0.3 => Attention. ::

Seemingly Similar Threads