asterisk users - Oct 2003 - Error

When dialling in and dialling my extension, when answered I get


" Read_channel ## vpb/1-3: Setting record mode, bridge = 0
WARNING[20499]: File chan_sip.c, Line 1111 (sip_write): Asked to
transmit frame type 8, while native formats is 4 (read/write = 4/4)
== Spawn extension (default, 1004, 1) exited non-zero on 'vpb/1-3'
-- hangup on vpb (vpb/1-3)
-- Hungup on vpb/1-3 complete
-- Event [12=>[02] Loop Drop "


And it hangs up the line any ideas ???

Regards Mick

Translation: Asked to transmit frame type G.711 A-law, while native
formats is G.711 u-law (read/write =G.711 u-law/G.711 u-law)

Looks to me like you need a disallow=all in your sip.conf and allowlines for the
codecs you want to allow, then make sure that the IP
phones you are using support at least one of the codecs you are
specifying in sip.conf

Try "show codecs" at the Asterisk CLI.

On Mon, 2003-10-13 at 04:48, mick@netexpress.com.au
wrote:
> When dialling in and dialling my extension, when answered I get
> 
> 
> " Read_channel ## vpb/1-3: Setting record mode, bridge = 0
> WARNING[20499]: File chan_sip.c, Line 1111 (sip_write): Asked to
> transmit frame type 8, while native formats is 4 (read/write = 4/4)
> == Spawn extension (default, 1004, 1) exited non-zero on 'vpb/1-3'
> -- hangup on vpb (vpb/1-3)
> -- Hungup on vpb/1-3 complete
> -- Event [12=>[02] Loop Drop "
> 
> 
> And it hangs up the line any ideas ???
> 
> Regards Mick 
> 
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
-- 
Sample configs and more: http://www.fnords.org/~eric/asterisk/

BTEL Consulting
+1-850-484-4535 x2111 (Pensacola)
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+1-877-677-9643 x2111 (Toll Free)

I just set up asterisk for the first time and I keep seeing the following across
the console:

NOTICE[1192484144]: File sched.c, Line 209 (sched_settime): Request to schedule
in the past

any idea what this means?

Thanks
Ron

Partial message is available.


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