Odd Error

Asterisk Asterisk Support
#1

Google didn’t turn up many results for this error.

Should I be concerned with this error? Thanks for any help.

[quote][Apr 24 09:08:01] WARNING[7217] translate.c: no samples for ulawtolin
[Apr 24 09:08:16] WARNING[7217] translate.c: no samples for ulawtolin
[Apr 24 09:08:31] WARNING[7217] translate.c: no samples for ulawtolin
[Apr 24 09:08:46] WARNING[7217] translate.c: no samples for ulawtolin
[Apr 24 09:09:01] WARNING[7217] translate.c: no samples for ulawtolin[/quote]

Also, I’m getting these from time to time as well, which I think it due to changes with new asterisk version.

[quote][Apr 24 10:10:00] WARNING[8755] app_queue.c: The device state of this queue member, SIP/105, is still ‘Not in Use’ when it probably should not be! Please check UPGRADE.txt for correct configuration settings.
[Apr 24 10:10:36] WARNING[8846] app_queue.c: The device state of this queue member, SIP/110, is still ‘Not in Use’ when it probably should not be! Please check UPGRADE.txt for correct configuration settings.
[Apr 24 10:20:49] WARNING[8872] app_queue.c: The device state of this queue member, SIP/110, is still ‘Not in Use’ when it probably should not be! Please check UPGRADE.txt for correct configuration settings.
[Apr 24 10:21:01] WARNING[8868] app_queue.c: The device state of this queue member, SIP/105, is still ‘Not in Use’ when it probably should not be! Please check UPGRADE.txt for correct configuration settings.[/quote]

#2

It all depends on what you are trying to do with this device. No one could see through SIP/105 without your config.

#3

SIP/105 is just one of the phones within my queue.
I have 10 phones/extensions that are static within a queue. And I have a hundred different DID’s that route to that queue as normal.

[quote]extension.conf

exten => 12065******,1,Playback(/var/lib/asterisk/sounds/greeting)
exten => 12065******,2,Set(CALLERID(name)=Company)
exten => 12065******,3,Queue(CSR1)
[/quote]

[quote]
sip.conf

[105]
type=friend
context=internal
callerid=105
host=dynamic
secret=SECRET
canreinvite=no
insecure=port,invite
allow=all
nat=yes[/quote]

That’s pretty much it. It’s a very basic setup. I’m sure I missed something with the changes in Asterisk v1.4.2 - just can’t figure it out though. :frowning:

#4

Ah, you are using the device in queue.

[quote=“UPGRADE.txt”]* Queues depend on the channel driver reporting the proper state
for each member of the queue. To get proper signalling on
queue members that use the SIP channel driver, you need to
enable a call limit (could be set to a high value so it
is not put into action) and also make sure that both inbound
and outbound calls are accounted for.

Example:

   [general]
   limitonpeer = yes

   [peername]
   type=friend
   call-limit=10

[/quote]

#5

Sorry for my stupidity…

So I put the following in my queues.conf

[quote][general]
limitonpeer = yes
[/quote]

And the following in each sip device?:

[quote][peername]
type=friend
call-limit=10 [/quote]

Thanks for the help.

#6

They would all go to sip.conf, I think.