Gervais de Montbrun
2005-Nov-10 19:28 UTC
[Asterisk-Users] Asterisk 1.2.0-RC1 Crashing with g729 codec and ATA 18
Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> on Thursday, November 10, 2005 at 5:16 AM -0400 wrote:>the 12SP should work > >SergioI 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@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@500|20|tr") in new stack -- Called 500@500 -- SIP/500-6d74 is ringing -- SIP/500-6d74 answered SCCP/140-00000002 This works as expected. Calls out to PSTN works fine also. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051110/45186822/attachment.htm
Sergio Chersovani
2005-Nov-11 02:41 UTC
[Asterisk-Users] Asterisk 1.2.0-RC1 Crashing with g729 codec and ATA 18
Gervais de Montbrun ha scritto:> **keepalive = 5set the keepalive to 60 or more> speeddial = 500,500,500@default >that phone should not be able to display a hint status so speeddial = 500,500> This is what is displayed in the console when I try to call the 12SP > from the ATAThe log could be more verbose than this. Set debug = 10 in your sccp.conf or in the console sccp debug 10 You should see what is happening with your audio stream Sergio
Matt Riddell
2005-Nov-12 18:45 UTC
[Asterisk-Users] Asterisk 1.2.0-RC1 Crashing with g729 codec and ATA 18
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; charset=3DISO-8859-1=22> <style type=3D=22text/css=22>body=7Bmargin-left:10px;margin-right:10px;margin-top:10px;margin-bottom:10px;=7D</style> </head> <body marginleft=3D=2210=22 marginright=3D=2210=22 margintop=3D=2210=22 marginbottom=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-users=40lists.digium.com=22>asterisk-users=40lists.digium.com</a>> on Thursday, November 10, 2005 at 5:16 AM -0400 wrote:<br> </b></font><span style=3D=22background-color:=23d0d0d0=22><font 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>the 12SP should work</font></span><font 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=22Geneva=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=23000000=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 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> =5Bgeneral=5D<br> keepalive =3D 5 <br> context =3D default<br> dateFormat =3D D.M.Y ;&=23160;M-D-Y&=23160;in&=23160;any&=23160;order&=23160;(5&=23160;chars&=23160;max)<br> bindaddr =3D 192.168.1.125 &=23160; ;&=23160;asterisk&=23160;box.<br> port =3D 2000 &=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 style=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 style=3D=22font-family:Geneva;font-size:12pt;color:=23000000;=22> -- Executing Dial("SIP/500-fc17", "SCCP/140=7C20=7Ctr") 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: SEP003080629796: 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 style=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 the 12SP<br> </font><font 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>Executing Dial("SCCP/140-00000002", "SIP/500=40500=7C20=7Ctr") in new 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 style=3D=22font-family:Geneva;font-size:12pt;color:=230000DD;=22>This works as 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)