It gets hard to read my logs when every time someone makes a phone call it
displays long pages of "Dropping voice frame". Anyone encounter this
before? Asterisk is bridging two SIP lines together, so the technology
should be the same. Maybe I'll try allowing only ULAW.
**************************************
Asterisk Standard debug (level 3)
***************************************
-- Called trunk2/12095387895
[Nov 26 13:49:37] WARNING[7744]: channel.c:3021 set_format: Unable to find a
codec translation path from unknown to unknown
[Nov 26 13:49:37] WARNING[7744]: channel.c:3402 ast_channel_make_compatible:
Unable to set read format on channel SIP/trunk2-0990c538 to 524288
-- SIP/trunk1-098dc208 is making progress passing it to
Local/XXXXXXXXXX at default-c101,2
-- Local/XXXXXXXXXX at default-c101,1 is making progress passing it to
SIP/trunk0-098cf098
[Nov 26 13:49:38] NOTICE[7062]: channel.c:2493 __ast_read: Dropping
incompatible voice frame on Local/XXXXXXXXXX at default-c101,1 of format ulaw
since our native format has changed to unknown
[Nov 26 13:49:38] NOTICE[7062]: channel.c:2493 __ast_read: Dropping
incompatible voice frame on Local/XXXXXXXXXX at default-c101,1 of format ulaw
since our native format has changed to unknown
[Nov 26 13:49:38] NOTICE[7062]: channel.c:2493 __ast_read: Dropping
incompatible voice frame on Local/XXXXXXXXXX at default-c101,1 of format ulaw
since our native format has changed to unknown
[Nov 26 13:49:38] NOTICE[7062]: channel.c:2493 __ast_read: Dropping
incompatible voice frame on Local/XXXXXXXXXX at default-c101,1 of format ulaw
since our native format has changed to unknown
**************************************
SIP.CONF Example Line
***************************************
[trunk0]
authuser=191691245XX
username=191691245XX
fromuser=191691245XX
secret=12345
fromdomain=richmond-1.vtnoc.net
host=richmond-1.vtnoc.net
dtmf=auto
dtmfmode=inband
insecure=port,invite
qualify=yes
type=peer
canreinvite=yes
call-limit=2
context=viatalk
--
/Nick
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