shawnl@up.net
2007-Apr-10 12:31 UTC
[asterisk-users] Help w/ Asterisk Cisco IP phone and SCCP
I have a new asterisk installation (1.4.2) that is working fine with SIP. Now I'm trying to add 2 cisco ip phones (7960) running SCCP (latest chan_sccp). I have the phones booted, and the tftp directory all setup, etc. But the phones do not quite work right. When I lift the handset I only get a dial-tone 1 out of 5 or so times I try, though hitting the speaker button works. I can dial from SCCP -> SIP phone with no problems, but not SIP -> SCCP or SCCP -> SCCP. I have a feeling I'm forgetting something fairly easy and stupid, but I can't seem to see what it is. Anyone have any suggestions? sccp.conf [general] keepalive = 30 context = internal bindaddr = 192.168.1.1 port = 2000 debug = 10 firstdigittimeout = 16 digittimeout = 8 [devices] type = 7960 description = Cisco1 tzoffset = 0 autologin = 104 ; speeddial = 101, 105 device => SEP00036BC3852B [lines] id = Cisco1 pin = 1234 label = 104 description = Cisco1 context = internal ;callwaiting = 1 incominglimit = 2 mailbox = 500 vmnum = 500 cid_name = Cisco1 cid_num = 104 line => 104 extensions.conf [internal] include => outbound-local include => outbound-long-distance ; Software phone exten => 101,1,Dial(SIP/test-softphone,,r) exten => 102,1,Dial(SIP/bob,20) exten => 102,2,Voicemail(u102) exten => 102,102,Voicemail(b102) exten => 102,103,Hangup() exten => 103,1,Dial(SIP/bill,20) exten => 103,2,Voicemail(u103) exten => 103,102,Voicemail(b103) exten => 103,103,Hangup() exten => 104,Dial(SCCP/SEP00036BC3852B,20) exten => 104,2,Voicemail(u104) exten => 104,102,Voicemail(b104) exten => 104,103,Hangup() exten => 105,Dial(SCCP/SEP00036B095612,20) exten => 105,2,Voicemail(u105) exten => 105,102,Voicemail(b105) exten => 105,103,Hangup()
Lacy Moore - Aspendora
2007-Apr-10 12:51 UTC
[asterisk-users] Help w/ Asterisk Cisco IP phone and SCCP
On 4/10/07, shawnl@up.net <shawnl@up.net> wrote:> exten => 104,Dial(SCCP/SEP00036BC3852B,20) > exten => 104,2,Voicemail(u104) > exten => 104,102,Voicemail(b104) > exten => 104,103,Hangup() >Off the top of my head, I would say that your dial statement should be Dial(SCCP/104,20). You should be dialing the line, not the device.
shawnl@up.net
2007-Apr-10 13:00 UTC
[asterisk-users] Help w/ Asterisk Cisco IP phone and SCCP
On Tue, Apr 10, 2007 at 02:51:31PM -0500, Lacy Moore - Aspendora wrote:> On 4/10/07, shawnl@up.net <shawnl@up.net> wrote: > >exten => 104,Dial(SCCP/SEP00036BC3852B,20) > >exten => 104,2,Voicemail(u104) > >exten => 104,102,Voicemail(b104) > >exten => 104,103,Hangup() > > > > Off the top of my head, I would say that your dial statement should be > Dial(SCCP/104,20). You should be dialing the line, not the device.Tried that as well. -Shawn
Jason Parker
2007-Apr-10 13:12 UTC
[asterisk-users] Help w/ Asterisk Cisco IP phone and SCCP
----- shawnl@up.net wrote:> [snip] > > I have a feeling I'm forgetting something fairly easy and stupid, but > I > can't seem to see what it is. Anyone have any suggestions? >Dial(SCCP/line@device) -- Jason Parker Digium
end1r@comcast.net
2007-Apr-10 13:30 UTC
[asterisk-users] Help w/ Asterisk Cisco IP phone and SCCP
Do you have any console messages? SCCP uses a "station start tone" message with a value of "Inside Dial Tone" and a direction of "Tone Output User". and the "line instance" and "Station Tone Output Direction" should be set to something other than 0. SCCP runs over TCP so you should get this message, but it would be interesting to see if you get this message the phone still doesnt play dial tone. My experiences with chan_sccp have been disappointing at best. If you can get a trace of both SCCP legs send it to me and i can take a look at it. -------------- Original message ---------------------- From: Jason Parker <jparker@digium.com>> ----- shawnl@up.net wrote: > > [snip] > > > > I have a feeling I'm forgetting something fairly easy and stupid, but > > I > > can't seem to see what it is. Anyone have any suggestions? > > > > Dial(SCCP/line@device) > > -- > Jason Parker > Digium > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users
Lacy Moore - Aspendora
2007-Apr-10 13:35 UTC
[asterisk-users] Help w/ Asterisk Cisco IP phone and SCCP
On 4/10/07, shawnl@up.net <shawnl@up.net> wrote:> exten => 104,Dial(SCCP/SEP00036BC3852B,20) > exten => 104,2,Voicemail(u104) > exten => 104,102,Voicemail(b104) > exten => 104,103,Hangup() >Actually, if this is a cut and paste, you are missing the 1. It should be: exten => 104,1,Dial... you have exten => 104,Dial... Also, jumping to n+101 is not the default in Asterisk 1.2+, you might want to search the wiki (www.voip-info.org) for priority jumping and it can explain much better than I can. A better question, I guess, is this chan_sccp or chan_skinny? If chan_sccp did you successfully compile it with the patches for 1.4?
shawnl@up.net
2007-Apr-10 17:05 UTC
[asterisk-users] Help w/ Asterisk Cisco IP phone and SCCP
On Tue, Apr 10, 2007 at 03:35:23PM -0500, Lacy Moore - Aspendora wrote:> On 4/10/07, shawnl@up.net <shawnl@up.net> wrote: > >exten => 104,Dial(SCCP/SEP00036BC3852B,20) > >exten => 104,2,Voicemail(u104) > >exten => 104,102,Voicemail(b104) > >exten => 104,103,Hangup() > > > Actually, if this is a cut and paste, you are missing the 1. It should be: > > exten => 104,1,Dial... > > you have > > exten => 104,Dial...Sorry, it was a typo> > Also, jumping to n+101 is not the default in Asterisk 1.2+, you might > want to search the wiki (www.voip-info.org) for priority jumping and > it can explain much better than I can. > > A better question, I guess, is this chan_sccp or chan_skinny? If > chan_sccp did you successfully compile it with the patches for 1.4?chan_sccp with the patches for 1.4 Thanks Shawn