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= harset=3DISO-8859-1=22> <style type=3D=22text/css=22>body=7Bmargin-left:10px;margin-right:10px;marg= in-top:10px;margin-bottom:10px;=7D</style> </head> <body marginleft=3D=2210=22 marginright=3D=2210=22 margintop=3D=2210=22 mar= 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> </body></html> ----=_--000d1f7e.000d1f7d.bf99b263-- --===============8001218576608901889== Content-Type: text/plain; charset="us-ascii" MIME-Version: 1.0 Content-Transfer-Encoding: 7bit Content-Disposition: inline _______________________________________________ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --===============8001218576608901889==-- -- Cheers, Matt Riddell _______________________________________________ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://freevoip.gedameurope.com (Free Asterisk Voip Community) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) _______________________________________________ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users