asterisk users - Jul 2004 - ATA 186, firmware SIP 3.1 and codec g.726

I have a ATA 186 with SIP firmware 3.1 when I changed the configurations to
use the g.726 codec I received many erros and the calls doesn't work.

I changed the fields:

- LBRCodec: 6 <- the code for g.726
- TXCodec: 6
- RxCodec: 6

The errors:

Jul  9 13:15:37 NOTICE[1192491824]: rtp.c:500 ast_rtp_read: Unable to
calculate samples for format G726
Jul  9 13:15:37 NOTICE[1192491824]: channel.c:1508 ast_set_read_format:
Unable to find a path from G726 to SLINR
Jul  9 13:15:37 NOTICE[1192491824]: channel.c:1478 ast_set_write_format:
Unable to find a path from ILBC to G726
Jul  9 13:15:37 WARNING[1192491824]: codec_ilbc.c:141 ilbctolin_framein:
Huh?  An ilbc frame that isn't a multiple of 50 bytes long from RTP (40)?
Jul  9 13:15:37 WARNING[1192491824]: chan_sip.c:1333 sip_write: Asked to
transmit frame type 1024, while native formats is 16 (read/write = 64/1024)

I will appreciate any help.

Kind regards,

Miguel

I see no errors.. I see three NOTICES and two WARNINGS.

bkw
> -----Original Message-----
> From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-
> admin@lists.digium.com] On Behalf Of miguel@amplanet.com.br
> Sent: Friday, July 09, 2004 11:31 AM
> To: asterisk-users@lists.digium.com
> Subject: [Asterisk-Users] ATA 186, firmware SIP 3.1 and codec g.726
>
> I have a ATA 186 with SIP firmware 3.1 when I changed the configurations
> to
> use the g.726 codec I received many erros and the calls doesn't work.
>
> I changed the fields:
>
> - LBRCodec: 6 <- the code for g.726
> - TXCodec: 6
> - RxCodec: 6
>
> The errors:
>
> Jul  9 13:15:37 NOTICE[1192491824]: rtp.c:500 ast_rtp_read: Unable to
> calculate samples for format G726
> Jul  9 13:15:37 NOTICE[1192491824]: channel.c:1508 ast_set_read_format:
> Unable to find a path from G726 to SLINR
> Jul  9 13:15:37 NOTICE[1192491824]: channel.c:1478 ast_set_write_format:
> Unable to find a path from ILBC to G726
> Jul  9 13:15:37 WARNING[1192491824]: codec_ilbc.c:141 ilbctolin_framein:
> Huh?  An ilbc frame that isn't a multiple of 50 bytes long from RTP
(40)?
> Jul  9 13:15:37 WARNING[1192491824]: chan_sip.c:1333 sip_write: Asked to
> transmit frame type 1024, while native formats is 16 (read/write >
64/1024)
>
> I will appreciate any help.
>
> Kind regards,
>
> Miguel
>
>
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You was correct, I just upgraded to the lastest CVS and the g.726 codec is
working.

But, when I use the ATA-186 the call works but I received the NOTICE below,
with sipura all is ok.
  
  -- Executing Dial("SIP/2007-ca45", "Zap/1/2132979") in new
stack
    -- Called 1/2132979
    -- Zap/1-1 answered SIP/2007-ca45
Jul 11 09:51:07 NOTICE[1200884528]: rtp.c:285 process_rfc3389: RFC3389
support incomplete.  Turn off on client if possible
Jul 11 09:51:07 NOTICE[1200884528]: rtp.c:316 process_rfc3389: Don't know
how to handle RFC3389 for receive codec 16
Jul 11 09:51:16 NOTICE[1200884528]: rtp.c:316 process_rfc3389: Don't know
how to handle RFC3389 for receive codec 16
    -- Hungup 'Zap/1-1'
  == Spawn extension (from-sip, 02132979, 1) exited non-zero on
'SIP/2007-ca45'


Subject: Re: [Asterisk-Users] RE: ATA 186, firmware SIP 3.1 and codec g.726
	+ now SIPURA SPA-2000
From: Eric Wieling <eric@fnords.org>
To: asterisk-users@lists.digium.com
Organization: BTEL Consulting
Date: Sat, 10 Jul 2004 10:25:59 -0500
Reply-To: asterisk-users@lists.digium.com

On Fri, 2004-07-09 at 13:55, miguel@amplanet.com.br
wrote:
> To me it's a error if I can't complete calls using the ATA
configured to
use
> the g726 codec.
> 
> I just tried it usign a sipura SPA-2000 (preferred codec: G726-32) and I
> received NOTICES and WARNINGS, but I can't complete a call.
It looks to me that you are using CVS -stable (which seems to support
G726 PASSTHRU) and not CVS -head (which supports G726 TRANSCODING, which
is what you need).  What does "show version" at the CLI show.

-- 
          Eric Wieling * BTEL Consulting * 504-899-1387 x2111
"In a related story, the IRS has recently ruled that the cost of Windows
upgrades can NOT be deducted as a gambling loss."

Thank you, now all is working good.


From: "Senad Jordanovic" <senad@boltblue.com>
To: <asterisk-users@lists.digium.com>
Subject: RE: [Asterisk-Users] RE: ATA 186, firmware SIP 3.1 and codec g.726
Date: Sun, 11 Jul 2004 15:42:02 +0100
Reply-To: asterisk-users@lists.digium.com

miguel@amplanet.com.br wrote:
> You was correct, I just upgraded to the lastest CVS and the g.726
> codec is working. 
> 
> But, when I use the ATA-186 the call works but I received the NOTICE
> below, with sipura all is ok. 
> 
>   -- Executing Dial("SIP/2007-ca45", "Zap/1/2132979")
in new stack
>     -- Called 1/2132979
>     -- Zap/1-1 answered SIP/2007-ca45
> Jul 11 09:51:07 NOTICE[1200884528]: rtp.c:285 process_rfc3389:
>     RFC3389 support incomplete.  Turn off on client if possible Jul
>   11 09:51:07 NOTICE[1200884528]: rtp.c:316 process_rfc3389: Don't
> know how to handle RFC3389 for receive codec 16 Jul 11 09:51:16
> NOTICE[1200884528]: rtp.c:316 process_rfc3389: Don't know how to
> handle RFC3389 for receive codec 16 -- Hungup 'Zap/1-1' == Spawn
> extension (from-sip, 02132979, 1) exited non-zero on
'SIP/2007-ca45'
Set AudioMode to "0x00140014" in ATA.