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<font face="Geneva" size="+0" color="#000000" style="font-family:Geneva;font-size:10pt;color:#000000;"><b>Asterisk Users Mailing List - Non-Commercial Discussion <<a href="mailto:asterisk-users@lists.digium.com">asterisk-users@lists.digium.com</a>> on Thursday, November 10, 2005 at 5:16 AM -0400 wrote:<br>
</b></font><span style="background-color:#d0d0d0"><font face="Geneva" size="+0" color="#000000" style="font-family:Geneva;font-size:12pt;color:#000000;">the 12SP should work</font></span><font face="Geneva" size="+0" color="#000000" style="font-family:Geneva;font-size:12pt;color:#000000;"><br>
</font><span style="background-color:#d0d0d0"><font face="Geneva" size="+0" color="#000000" style="font-family:Geneva;font-size:12pt;color:#000000;"><br>
Sergio<br>
</font></span><font face="Geneva" size="+0" color="#000000" style="font-family:Geneva;font-size:12pt;color:#000000;"><br>
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.<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 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<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>
[general]<br>
keepalive = 5 <br>
context = default<br>
dateFormat = D.M.Y ; M-D-Y in any order (5 chars max)<br>
bindaddr = 192.168.1.125   ; asterisk box.<br>
port = 2000  ; listen on port 2000 (Skinny, default)<br>
debug = 0<br>
<br>
[devices]<br>
type = 12<br>
description = Office<br>
tzoffset = 0<br>
autologin = 140<br>
speeddial = 500,500,500@default<br>
device => SEP003080629796<br>
<br>
<br>
[lines]<br>
id = 140<br>
pin = 1234<br>
label = "TLS Group"<br>
description = Office<br>
context = default<br>
callwaiting = 1<br>
incominglimit = 2<br>
mailbox = 1000<br>
vmnum = *98<br>
cid_name = Office<br>
cid_num = 140<br>
line => 140<br>
<br>
/etc/asterisk/sip.conf<br>
[general]<br>
port = 5060<br>
bindaddr = 0.0.0.0<br>
context = default<br>
<br>
disallow=all<br>
allow=g729<br>
allow=gsm<br>
allow=speex<br>
allow=ilbc<br>
<br>
[500]<br>
type=friend<br>
username=500<br>
callerid="TLS Group"<br>
secret=mypassword<br>
canreinvite=no<br>
host=dynamic<br>
dtmfmode=rfc2833<br>
mailbox=1000<br>
nat=1<br>
<br>
/etc/asterisk/extensions.conf<br>
exten => 140,1,Dial(SCCP/140,20,tr)<br>
exten => 140,2,Voicemail(u140)<br>
exten => 140,3,Goto(mainmenu,s,2)<br>
exten => 140,102,Voicemail(b140)<br>
exten => 140,103,Goto(mainmenu,s,2)<br>
<br>
</font><font face="Geneva" size="+0" color="#0000DD" style="font-family:Geneva;font-size:12pt;color:#0000DD;">This is what is displayed in the console when I try to call the 12SP from the ATA<br>
</font><font face="Geneva" size="+0" color="#000000" style="font-family:Geneva;font-size:12pt;color:#000000;"> -- Executing Dial("SIP/500-fc17", "SCCP/140|20|tr") 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[1693]: sccp_socket.c:308 sccp_socket_thread: SEP003080629796: Dead device does not send a keepalive message in 5 seconds. Will be removed<br>
</font><font face="Geneva" size="+0" color="#0000DD" style="font-family:Geneva;font-size:12pt;color:#0000DD;">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 the 12SP<br>
</font><font face="Geneva" size="+0" color="#000000" style="font-family:Geneva;font-size:12pt;color:#000000;">Executing Dial("SCCP/140-00000002", "SIP/500@500|20|tr") in new stack<br>
-- Called 500@500<br>
-- SIP/500-6d74 is ringing<br>
-- SIP/500-6d74 answered SCCP/140-00000002<br>
</font><font face="Geneva" size="+0" color="#0000DD" style="font-family:Geneva;font-size:12pt;color:#0000DD;">This works as expected. Calls out to PSTN works fine also.</font>
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