[Asterisk-Users] BudgetTone 100 + NuFone

[Kristian Kielhofner]

Jean-Michel Hiver wrote:
> Hi List,
> 
> I am trying to get my budget sip phone to work with asterisk, which in 
> turn is configured to work with NuFone. I can get the phone to ring my 
> home PSTN'ed phone but as soon as I pick up my home phone it hangs.
> 
> Here's what I get in the log:
> 
> Nov  4 18:37:44 WARNING[1191013296]: chan_sip.c:1810 sip_write: Asked to 
> transmit frame type 1, while native formats is 4 (read/write = 1/1)
> Nov  4 18:37:44 WARNING[1191013296]: chan_sip.c:1810 sip_write: Asked to 
> transmit frame type 1, while native formats is 4 (read/write = 1/1)
> Nov  4 18:37:44 WARNING[1191013296]: chan_sip.c:1810 sip_write: Asked to 
> transmit frame type 1, while native formats is 4 (read/write = 1/1)
>    -- IAX2/NuFone/2 is ringing
> Nov  4 18:37:44 WARNING[1191013296]: chan_sip.c:1810 sip_write: Asked to 
> transmit frame type 1, while native formats is 4 (read/write = 1/1)
> Nov  4 18:37:44 NOTICE[1191013296]: channel.c:1696 ast_set_write_format: 
> Unable to find a path from G723 to ULAW
>    -- IAX2/NuFone/2 stopped sounds
>    -- IAX2/NuFone/2 answered SIP/2000-7e6e
> Nov  4 18:37:49 WARNING[1191013296]: channel.c:2135 
> ast_channel_make_compatible: No path to translate from SIP/2000-7e6e(4) 
> to IAX2/NuFone/2(1)
> Nov  4 18:37:49 WARNING[1191013296]: app_dial.c:1023 dial_exec: Had to 
> drop call because I couldn't make SIP/2000-7e6e compatible with 
> IAX2/NuFone/2
>    -- Hungup 'IAX2/NuFone/2'
>  == Spawn extension (from-sip, 3000, 1) exited non-zero on 'SIP/2000-7e6e'
>    -- Registered to '69.73.19.178', who sees us as 80.12.162.250:1026
> 
> Any ideas of what I am missing? It would seem that the key is "Unable to 
> find a path from G723 to ULAW" but I don't know what it means :-)
> 
> Cheers,
> Jean-Michel.

Jean-Michel,

	Asterisk does not really have support for G723.  It can do passthrough, 
but it will not transcode it.  NuFone does not support it.  That is 
where your error is coming from.  Make sure that you do this:

iax.conf:

disallow=all
allow=ilbc
allow=gsm
allow=ulaw

sip.conf:

disallow=all
allow=ilbc
allow=gsm
allow=ulaw

I think that should do what you need.  Please read up on codecs at
http://www.voip-info.org/wiki-Asterisk+codecs

--
Kristian Kielhofner