[Asterisk-Users] Asterisk 1.2.0-RC1 Crashing with g729 codec and ATA 18
Matt Riddell
matt.riddell at sineapps.com
Sat Nov 12 18:45:14 MST 2005
PLEASE DO NOT POST IN HTML! :)
Gervais de Montbrun wrote:
YPE HTML PUBLIC =22-//W3C//DTD HTML 4.0 Transitional//EN=22>
<html><head><meta http-equiv=3D=22Content-Type=22 content=3D=22text/html; c=
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</head>
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ginbottom=3D=2210=22>
<font face=3D=22Geneva=22 size=3D=22+0=22 color=3D=22=23000000=22 style=3D=
=22font-family:Geneva;font-size:10pt;color:=23000000;=22><b>Asterisk Users =
Mailing List - Non-Commercial Discussion <<a href=3D=22mailto:asterisk-u=
sers=40lists.digium.com=22>asterisk-users=40lists.digium.com</a>> on Thu=
rsday, November 10, 2005 at 5:16 AM -0400 wrote:<br>
</b></font><span style=3D=22background-color:=23d0d0d0=22><font face=3D=22G=
eneva=22 size=3D=22+0=22 color=3D=22=23000000=22 style=3D=22font-family:Gen=
eva;font-size:12pt;color:=23000000;=22>the 12SP should work</font></span><f=
ont face=3D=22Geneva=22 size=3D=22+0=22 color=3D=22=23000000=22 style=3D=
=22font-family:Geneva;font-size:12pt;color:=23000000;=22><br>
</font><span style=3D=22background-color:=23d0d0d0=22><font face=3D=22Genev=
a=22 size=3D=22+0=22 color=3D=22=23000000=22 style=3D=22font-family:Geneva;=
font-size:12pt;color:=23000000;=22><br>
Sergio<br>
</font></span><font face=3D=22Geneva=22 size=3D=22+0=22 color=3D=22=2300000=
0=22 style=3D=22font-family:Geneva;font-size:12pt;color:=23000000;=22><br>
I half-managed to get my 12SP working with sccp and I am able to call it wi=
th my ATA. The ATA and my cordless phone is still configured using SIP.<br>
<br>
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 b=
e answered, but there is no audio at all. When I hang up, I can see that th=
e 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 tha=
t way for at least 1 minute after hanging up until I restarted asterisk<br>
<br>
What am I doing wrong?<br>
<br>
I'm running rc-1 of asterisk with the latest sccp 20051108.<br>
<br>
Thanks in advance,<br>
Gervais<br>
-----------------------------------------------<br>
<br>
/etc/asterisk/sccp.conf<br>
=5Bgeneral=5D<br>
keepalive =3D 5 <br>
context =3D default<br>
dateFormat =3D D.M.Y =
&nb=
sp; =
;&=23160;M-D-Y&=23160;in&=23160;any&=23160;order&=23160;(=
5&=23160;chars&=23160;max)<br>
bindaddr =3D 192.168.1.125 =
&nb=
sp; &=23160; ;&=23160;asterisk&=23160;box.<br>
port =3D 2000 &=
nbsp; &nbs=
p; &=
nbsp; &=23160;; listen&=23160;on&=23160;port&=23160;=
2000&=23160;(Skinny,&=23160;default)<br>
debug =3D 0<br>
<br>
=5Bdevices=5D<br>
type =3D 12<br>
description =3D Office<br>
tzoffset =3D 0<br>
autologin =3D 140<br>
speeddial =3D 500,500,500=40default<br>
device =3D> SEP003080629796<br>
<br>
<br>
=5Blines=5D<br>
id =3D 140<br>
pin =3D 1234<br>
label =3D "TLS Group"<br>
description =3D Office<br>
context =3D default<br>
callwaiting =3D 1<br>
incominglimit =3D 2<br>
mailbox =3D 1000<br>
vmnum =3D *98<br>
cid_name =3D Office<br>
cid_num =3D 140<br>
line =3D> 140<br>
<br>
/etc/asterisk/sip.conf<br>
=5Bgeneral=5D<br>
port =3D 5060<br>
bindaddr =3D 0.0.0.0<br>
context =3D default<br>
<br>
disallow=3Dall<br>
allow=3Dg729<br>
allow=3Dgsm<br>
allow=3Dspeex<br>
allow=3Dilbc<br>
<br>
=5B500=5D<br>
type=3Dfriend<br>
username=3D500<br>
callerid=3D"TLS Group"<br>
secret=3Dmypassword<br>
canreinvite=3Dno<br>
host=3Ddynamic<br>
dtmfmode=3Drfc2833<br>
mailbox=3D1000<br>
nat=3D1<br>
<br>
/etc/asterisk/extensions.conf<br>
exten =3D> 140,1,Dial(SCCP/140,20,tr)<br>
exten =3D> 140,2,Voicemail(u140)<br>
exten =3D> 140,3,Goto(mainmenu,s,2)<br>
exten =3D> 140,102,Voicemail(b140)<br>
exten =3D> 140,103,Goto(mainmenu,s,2)<br>
<br>
</font><font face=3D=22Geneva=22 size=3D=22+0=22 color=3D=22=230000DD=22 st=
yle=3D=22font-family:Geneva;font-size:12pt;color:=230000DD;=22>This is what=
is displayed in the console when I try to call the 12SP from the ATA<br>
</font><font face=3D=22Geneva=22 size=3D=22+0=22 color=3D=22=23000000=22 st=
yle=3D=22font-family:Geneva;font-size:12pt;color:=23000000;=22> =
-- Executing Dial("SIP/500-fc17", "SCCP/140=7C20=7Ctr&=
quot;) in new stack<br>
-- Called 140<br>
-- SCCP/140-00000001 is ringing<br>
-- SCCP/140-00000001 answered SIP/500-fc17<br>
Nov 10 22:06:05 WARNING=5B1693=5D: sccp_socket.c:308 sccp_socket_thread: SE=
P003080629796: Dead device does not send a keepalive message in 5 seconds. =
Will be removed<br>
</font><font face=3D=22Geneva=22 size=3D=22+0=22 color=3D=22=230000DD=22 st=
yle=3D=22font-family:Geneva;font-size:12pt;color:=230000DD;=22>The 12SP is =
dead until it gets reset. Again. No audio and phone "crashes"<br>
<br>
This is what is displayed in the console when I try to call the ATA from th=
e 12SP<br>
</font><font face=3D=22Geneva=22 size=3D=22+0=22 color=3D=22=23000000=22 st=
yle=3D=22font-family:Geneva;font-size:12pt;color:=23000000;=22>Executing Di=
al("SCCP/140-00000002", "SIP/500=40500=7C20=7Ctr") in n=
ew stack<br>
-- Called 500=40500<br>
-- SIP/500-6d74 is ringing<br>
-- SIP/500-6d74 answered SCCP/140-00000002<br>
</font><font face=3D=22Geneva=22 size=3D=22+0=22 color=3D=22=230000DD=22 st=
yle=3D=22font-family:Geneva;font-size:12pt;color:=230000DD;=22>This works a=
s expected. Calls out to PSTN works fine also.</font>
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Cheers,
Matt Riddell
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