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Author Topic: How to setuo sccp.conf and use SCCP under Asterisk  (Read 952 times)
bulek
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« on: November 30, 2011, 01:16:33 am »

Hi,

I've 3 7970 phones and they worked quite ok under 7.10. I'd like them to work under newer versions of LMCE - they also support a lot of various features that can be put in config file. But when I take a look at this config files, they all seem a bit cryptic for me.

Does anyone know of any document that would (shortly) explain features & settings of sccp channel under asterisk ?

Regards,

Bulek
 
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Thanks in advance,

regards,

Bulek.
cafedumonde
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« Reply #1 on: December 05, 2011, 04:26:34 pm »


I have one 7970, that is working fine under 810 using SCCP. I just used the instructions in with Wiki after a fresh install.

CDM
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bulek
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« Reply #2 on: December 08, 2011, 09:25:13 pm »

I have one 7970, that is working fine under 810 using SCCP. I just used the instructions in with Wiki after a fresh install.

CDM

Hi,

thanks for response. Could you please post some more details about how it looks (Asterisk version, your /etc/sccp.conf ) ??

It will help me a lot...

Regards,

Bulek.
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Thanks in advance,

regards,

Bulek.
daballiemo
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« Reply #3 on: December 11, 2011, 11:15:08 am »

I have (among others) a 7970, this is the sscp.conf. The 7970 has 2 lines. One especially for a door bell.

Code:
; (SCCP*)
;
; An implementation of Skinny Client Control Protocol (SCCP)
;
; Sergio Chersovani (mlists@c-net.it)
; http://chan-sccp.belios.de
;
; Supported phone types
;    12 -- Cisco IP Phone 12SP+ (or other 12 variants)
;    30 -- Cisco IP Phone 30VIP (or other 30 variants)
;  7902 -- Cisco IP Phone 7902G
;  7905 -- Cisco IP Phone 7905G
;  7910 -- Cisco IP Phone 7910
;  7912 -- Cisco IP Phone 7912G
;  7935 -- Cisco IP Conference Station 7935
;  7936 -- Cisco IP Conference Station 7936
;  7920 -- Cisco IP Wireless Phone 7920
;  7940 -- Cisco IP Phone 7940
;  7960 -- Cisco IP Phone 7960
;  7970 -- Cisco IP Phone 7970
;  7914 -- Cisco IP Phone 7960 with a 7914 addon
;   ata -- Cisco ATA-186 or Cisco ATA-188 (untested)

[general]
keepalive = 60               ; phone keep alive message evey 60 secs. Used to check the voicemail
debug = 1                    ; console debug level. 1 => 10
context = from-internal      ; default context for calls
dateFormat = D.M.Y           ; M-D-Y in any order. Use M/D/YA (for 12h format)
bindaddr = 192.168.80.1 ; replace with the ip address of the asterisk server (RTP important param)
port = 2000                  ; listen on port 2000 (Skinny, default)
disallow = all               ; First disallow all codecs
allow = alaw                 ; Allow codecs in order of preference
allow = ulaw                 ;
dnd = off                    ;
firstdigittimeout = 30       ; dialing timeout for the 1st digit
digittimeout = 3             ; more digits
autoanswer_ring_time = 1     ; ringing time in seconds for the autoanswer, the default is 0
autoanswer_tone = 0x32       ; autoanswer confirmation tone. For a complete list of tones: grep SKINNY_TONE sccp_protocol.h
                             ; not all the tones can be played in a connected state, so you have to try.
remotehangup_tone = 0x32     ; passive hangup notification. 0 for none
transfer_tone = 0            ; confirmation tone on transfer. Works only between SCCP devices
language=nl                  ; Default language setting this setting also sets the voicemailbox language
musicclass=han               ; Sets the default music on hold class


[devices]
;<pl_dev_183>
type = 7970
description = SEP0015FAEAB924
autologin = 220,222
transfer = on
park = on
speeddial =
cfwdall = off
cfwdbusy = off
dtmfmode = inband
device => SEP0015FAEAB924
;</pl_dev_183>
;<pl_dev_187>
type = 7970
description = SEP000D282E7921
autologin = 222
transfer = on
park = on
speeddial =
cfwdall = off
cfwdbusy = off
dtmfmode = inband
device => SEP000D282E7921
;</pl_dev_187>

[lines]
;<pl_line_183>
id = 220
pin =
label = 220
description = 220
context = from-internal
incominglimit = 2
transfer = on
secondary_dialtone_digits = 9
secondary_dialtone_tone = 0x22
cid_name = pl_183
cid_num = 220
line => 220
;</pl_line_183>
;<pl_line_187>
id = 222
pin =
label = 222
description = 222
context = from-internal
incominglimit = 2
transfer = on
secondary_dialtone_digits = 9
secondary_dialtone_tone = 0x22
cid_name = pl_187
cid_num = 222
line => 222
;</pl_line_187>
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bulek
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« Reply #4 on: December 12, 2011, 03:56:36 pm »

I have (among others) a 7970, this is the sscp.conf. The 7970 has 2 lines. One especially for a door bell.

....

Thanks for response and info. How is your doorbell connected to asterisk - is it just ordinary extension ? Is this setup different from setup where doorbell call comes in and goes to other phones by nornal call routing ?

Thanks in advance,

regards,

Bulek.
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Thanks in advance,

regards,

Bulek.
daballiemo
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« Reply #5 on: December 12, 2011, 09:09:19 pm »

Yup, I only have the doorbell routing to one phone instead of ringing all the phone. Within asterisk I have made a ringing group that is called by the door extension, and this extra line is the only extension in this ringing group except for my mobile phone.
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bugui
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« Reply #6 on: December 13, 2011, 06:22:07 am »

Hi daballiemo,
Do you have a camera on your system?

Thanks
Hernan
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daballiemo
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« Reply #7 on: December 14, 2011, 08:31:23 pm »

Hi daballiemo,
Do you have a camera on your system?

Thanks
Hernan


I do not understand the question in relation to the subject of this thread  Huh Maybe you have to explain....
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stallione
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« Reply #8 on: December 16, 2011, 08:48:11 pm »

I think he was asking about a door-bell camera, or a camera tied to the doorbell to see who is at the door. Smiley
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daballiemo
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« Reply #9 on: December 18, 2011, 05:50:43 pm »

I think he was asking about a door-bell camera, or a camera tied to the doorbell to see who is at the door. Smiley

I see, no I don't.

kr
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