[Asterisk-Users] Asterisk 1.2.0-RC1 Crashing with g729 codec and ATA 18

[Gervais de Montbrun]

Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com> on Thursday, November 10, 2005 at 5:16
AM -0400 wrote:
>the 12SP should work
>
>Sergio

I half-managed to get my 12SP working with sccp and I am able to call it
with my ATA. The ATA and my cordless phone is still configured using SIP.

I can call out from my Cisco 12 SP+ and everything seems to be working
fine. I can not however receive calls on the 12SP. The phone rings and it
can be answered, but there is no audio at all. When I hang up, I can see
that the phone reset. Also if I call in on the PSTN, I get similar results
except even after I hang up my 12SP the Zap channel is not released. It
stayed that way for at least 1 minute after hanging up until I restarted
asterisk

What am I doing wrong?

I'm running rc-1 of asterisk with the latest sccp 20051108.

Thanks in advance,
Gervais
-----------------------------------------------

/etc/asterisk/sccp.conf
[general]
keepalive = 5       
context = default
dateFormat = D.M.Y                                     
; M-D-Y in any order (5 chars max)
bindaddr = 192.168.1.125                             ; asterisk box.
port = 2000                                          ;
listen on port 2000 (Skinny, default)
debug = 0

[devices]
type        = 12
description = Office
tzoffset    = 0
autologin   = 140
speeddial   = 500,500,500 at default
device => SEP003080629796


[lines]
id = 140
pin = 1234
label = "TLS Group"
description = Office
context = default
callwaiting = 1
incominglimit = 2
mailbox = 1000
vmnum = *98
cid_name = Office
cid_num = 140
line => 140

/etc/asterisk/sip.conf
[general]
port = 5060
bindaddr = 0.0.0.0
context = default

disallow=all
allow=g729
allow=gsm
allow=speex
allow=ilbc

[500]
type=friend
username=500
callerid="TLS Group"
secret=mypassword
canreinvite=no
host=dynamic
dtmfmode=rfc2833
mailbox=1000
nat=1

/etc/asterisk/extensions.conf
exten => 140,1,Dial(SCCP/140,20,tr)
exten => 140,2,Voicemail(u140)
exten => 140,3,Goto(mainmenu,s,2)
exten => 140,102,Voicemail(b140)
exten => 140,103,Goto(mainmenu,s,2)

This is what is displayed in the console when I try to call the 12SP from
the ATA
   -- Executing Dial("SIP/500-fc17", "SCCP/140|20|tr") in new stack
    -- Called 140
    -- SCCP/140-00000001 is ringing
    -- SCCP/140-00000001 answered SIP/500-fc17
Nov 10 22:06:05 WARNING[1693]: sccp_socket.c:308 sccp_socket_thread:
SEP003080629796: Dead device does not send a keepalive message in 5
seconds. Will be removed
The 12SP is dead until it gets reset. Again. No audio and phone "crashes"

This is what is displayed in the console when I try to call the ATA from
the 12SP
Executing Dial("SCCP/140-00000002", "SIP/500 at 500|20|tr") in new stack
    -- Called 500 at 500
    -- SIP/500-6d74 is ringing
    -- SIP/500-6d74 answered SCCP/140-00000002
This works as expected. Calls out to PSTN works fine also.

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