I seem to be having a problem with transcoding and/or agreeing on a
valid codec.  I am running a new image pulled from CVS at 1:30 PM CST.
The issue occurs when I try to make a call to a toll-free number over
sipphone.com.

Here's what I see in the console:

NOTICE[1259545280]: File channel.c, Line 1478 (ast_set_read_format):
Unable to find a path from G729A to ULAW
NOTICE[1259545280]: File channel.c, Line 1448 (ast_set_write_format):
Unable to find a path from ULAW to G729A

Before somebody tells me "UTFG", I ALREADY HAVE.  Somebody else had a
similar issue last week and there was no real resolution posted.  So
here it is again.  I have all of the codecs that I support enabled in my
sip.conf.  Here is the relevant section:

;
; SIP Configuration for Asterisk
;
[general]
port = 5060                     ; Port to bind to
bindaddr = 0.0.0.0              ; Address to bind to
context = default               ; Default for incoming calls
srvlookup = yes         ; Enable SRV lookups on outbound calls
pedantic = yes                  ; Enable slow, pedantic checking for
Pingtel
;tos=lowdelay
;tos=184
maxexpirey=3600         ; Max length of incoming registration we allow
defaultexpirey=120              ; Default length of incoming/outoing
registration
;notifymimetype=text/plain      ; Allow overriding of mime type in
NOTIFY
;videosupport=yes               ; Turn on support for SIP video
disallow=all                    ; Disallow all codecs
allow=ulaw                      ; Allow codecs in order of preference
allow=alaw                      ; Allow codecs in order of preference
allow=gsm
allow=ilbc

register => 17476692375:[EMAIL PROTECTED]/1101

[sipphone]
type=peer
username=17476692375
secret=[MYSECRET]
host=proxy01.sipphone.com
fromuser=SteveSokol
fromdomain=sipphone.com
canreinvite=no

; ==END OF SIP.CONF FILE===

The issue occurs whenever any calls that route over the sipphone peer
are made to a toll-free number.  The calling phone (either my GS100 or
my X-LITE softphone) rings two or three times then gives me busy.  Here
is the entire debug output:

    -- Executing Dial("SIP/1101-1f83",
"SIP/[EMAIL PROTECTED]|20|tr") in new stack
    -- Called [EMAIL PROTECTED]
NOTICE[1234379840]: File channel.c, Line 1478 (ast_set_read_format):
Unable to find a path from G729A to ULAW
NOTICE[1234379840]: File channel.c, Line 1448 (ast_set_write_format):
Unable to find a path from ULAW to G729A
    -- SIP/sipphone.com-e7b3 is making progress passing it to
SIP/1101-1f83
    -- SIP/sipphone.com-e7b3 answered SIP/1101-1f83
    -- Attempting native bridge of SIP/1101-1f83 and
SIP/sipphone.com-e7b3
NOTICE[1242768320]: File channel.c, Line 1478 (ast_set_read_format):
Unable to find a path from G729A to ULAW
NOTICE[1242768320]: File channel.c, Line 1448 (ast_set_write_format):
Unable to find a path from ULAW to G729A
WARNING[1234379840]: File chan_sip.c, Line 1159 (sip_write): Asked to
transmit frame type 4, while native formats is 256 (read/write = 4/4)
  == Spawn extension (default, 918884510851, 1) exited non-zero on
'SIP/1101-1f83'

The problem does NOT occur when I call another sipphone.com user (i.e.
GS100 -> Asterisk -> Sipphone -> GS100).  Those calls go through just
fine.  The toll free calls were working last week.  Is it me, or is it
Sipphone.com?

Any suggestions would be greatly appreciated.

Steve


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