chan_sccp on trixbox 2.4 (or 2.6?)

Found this website which was a lot of help: http://southbrain.com/south/2008/02/cisco-7960-and-asterisk-1417-...

To get chan_sccp_b working on my trixbox I did the following steps:

yum update
I updated from trixbox 2.4 to trixbox 2.6.0. I'm not sure I had to, but I couldn't figure out how to install asterisk-devel without updating and I wanted to try 2.6.0 anyway. After the upgrade process was finished and freepbx modules updated, things worked just fine.

yum install asterisk-devel gcc
In order to compile the chan_sccp_b driver I needed to have a compiler which is not installed by default on trixbox and the developement part of asterisk.

Then I just followed the instructions on that web page to get the driver compiled and installed. I had to make a few minor tweaks to their example config to get my phone working well with trixbox, but it wasn't hard.

A few things I like about SCCP:

I can onhook dial. With SIP I had to push a button or lift the handset before I could start dialing. Not needed with SCCP.
The DND button works and it right on the main screen!
I can monitor another extension using one of the six buttons on my 7960!
Plus, I answered "yes" to the PARK compile option which gives me a "Park Call" button that works nicely.

Scott,

Youre going to use a custom in the dropdown extension menu to provision the sccp phone using freepbx.

There is come manual provisioning which also has to be done but you need to set it up as an extension too.

hope this is useful,
todd

Thanks I have searched MGCP ad nauseum and had not found that link.

If I get this pile of LG MGCP phones to provision on trix I will be one happy camper. We have had to buy back a bunch of them from a hosted Sylantro venture.

Regards....

Scott

This is a lot of great info. I'm looking at converting some of my SIP 7960s to SCCP tomorrow...just to see. I can already tell that I'm going to like this more than SIP.

I'll be able to contribute at some point...right now, I'm having a great time learning here.

Yes, the 2.4.x -> 2.6.x upgrade was simply just a "yum update". It worked super for me, however, my system is a simple home system with one phone line. Your mileage may vary. Make sure you do a backup before attempting.

Conference feature has not yet been implemented in CHAN-SCCP-B.

See my notes (at the end) on evaluating SCCP-CHAN-B currently with the 7960, 7960+7914 Sidecar & shortly 7970/71 under Asterisk v1.4

http://trixbox.org/forums/trixbox-forums/trixbox-endpoints/7971-7...

ksDevGuy

Hi,

If you grab the latest SVN version and then follow the setup and include the conf features you get access to the MeetMe style CCME conf option - no Barge or cBarge (as yet) but you can press conf (meetme) then dial a conf number say "10" then let other people know the conf number and they can join you there.

Works well and nearly as easy as CCME does it

Steve

How do I get chan_sccp installed? I downloaded a version and it errors and says can not find /usr/include

Can someone help?

Thanks

This is the page that helped me the most http://southbrain.com/south/2008/02/cisco-7960-and-asterisk-1417-...

You need to setup sccp.conf and setup custom extension numbers using SCCP/1234 as the DIAL setting where 1234 is the extension number. 1234 will then appear in the autologin for the phone under the device section and the id under the line section in sccp.conf.

In the asterisk CLI you can
sccp show devices
........ to see how the phones are configured
sccp show lines
........ to see how the lines are configured

module unload chan_sccp.so
module load chan_sccp.so

will unload and load i.e reload the sccp driver which you need to do everytime you edit the config file. Alternatively you can do an 'amportal restart'

If you are still stuck post your config.

Would you please edit your post and use the [code]tags[/code] so your post is legible?

If you look in /usr/lib/asterisk/modules can you find " chan_sccp.so " ? If you run

ls /usr/lib/asterisk/modules/chan_sccp*

this should give you an output like

/usr/lib/asterisk/modules/chan_sccp.so

If not your chan_sccp driver is not in the right place

Have you edited " /etc/asterisk/modules.conf " and inserted the lines

noload => chan_skinny.so
load => chan_sccp.so

then do an " amportal restart ". What do you get when, at the asterisk CLI you run " sccp show version "

hi there. someone, please help me.

my sistem is trixbox CE release is 2.6.2.2
i can't install chan_sccp-b_20090602. i have done:

yum update
yum install asterisk-devel gcc
after i download chan_sccp-b_20090602 in /mnt/chan_sccp-b_20090602.tar.gz
gunzip chan_sccp-b_20090602.tar.gz
tar xzf chan_sccp-b_20090602.tar
cd chan_sccp-b_20090602
make

but it says "-bash: make: command not found"

any ideas?

p.s. sorry for my english

I've been pulling my hair out on this one...
I went to sourceforge and downloaded the zip archive that sourceforge suggested.

This one doesn't work
I've tried it on several virtual machines and they all report errors. Pretty much like wauters described.

I downloaded this one: http://sourceforge.net/projects/chan-sccp-b/files/chan-sccp-b/v2-... which compiles nicely. I will be testing it tonight and keep y'all posted.

Well, no luck so far...
As soon as I copy to chan_sccp file to /usr/lib/asterisk/modules/ and I try to load it: module load chan_sccp.so, I get this error:

usr/sbin/safe_asterisk: line 148:  3112 Segmentation fault      (core dumped) nice -n $PRIORITY ${ASTSBINDIR}/asterisk -f ${CLIARGS} ${ASTARGS} >&/dev/${TTY} < /dev/${TTY}
Asterisk ended with exit status 139
Asterisk exited on signal 11.
Automatically restarting Asterisk.
mpg123: no process killed

And that just keeps on going....until I reboot the server...

Does anyone know what's wrong ?

The weird thing is, that when I run make, it outputs the following:

make: svnversion: Command not found
Creating config file
====================
Checking Asterisk version...
 * found asterisk 1.4

While I'm sure I'm using Trix 2.8, which should be based on Asterisk 1.6.
Could that be the problem ?

Edit:
I will try my luck with Trix 2.6...

2nd edit:
Finally! I've been at this for days now...
SCCP channel Release: v2 - (built by 'root' on 'Fri Aug 7 01:05:31 CEST 2009')

So, it seems that the Channel SCCP does not work with Trix 2.8...

On to the next 'challenge'...

Hello I just jumped into the VOIP scene, and have successfully managed to deploy various Cisco 79xx's with trixbox 2.8.0.1 using SIP and a Sangoma card. I recently acquired a 7936 conference phone which is SCCP only, so I have been trying to install chan_sccp_20090110.tar and chan_sccp-b_20090602.tar.gz into trixbox 2.8, and found I couldn't. I even tried formatting it, and trying it off a fresh install to remove updates and software I installed, but it didn't work. As the matter of fact it broke my Asterisk. Asterisk gave errors regarding segmentation faults after restarting with amportal restart and I could not start it with anything. I had trouble removing chan_sccp so I simply reinstalled trixbox. I am pretty sure I am following all the steps correctly because I used several guides. Also when making the chan_sccp config file it states that it detects Asterisk 1.4 instead of 1.6. I have come to the same conclusion as Mike2k that trixbox 2.8 and chan_sccp are incompatible at least as of the versions I previously listed. Right now I am downloading trixbox 2.6 optimistic that it will make the conference phone work.

So here go a few questions:
Are there any possible workarounds to make trixbox 2.8 and chan_sccp compatible ?
If not is there any work in progress for it?
Am I really missing out on anything if I downgrade to trixbox 2.6 (security, performance, stability, xml services and/or directories for cisco phones, sangoma card)?
How do I uninstall chan_sccp?
Will chan_sccp run off any other distro of asterisk 1.6 or a pure asterisk installation?

I also wanted to thank all the people on the forums (especially this one) that ask and answer questions.

Vittorio

Thank you Skyking for your fast and detailed response. I have already downgraded to trixbox 2.6, and am working on getting the conference phone to talk to trixbox.

Vittorio

Well, I created a new trixbox VM last night just to see what would happen and it worked fine -- no idea why my old system doesn't work though. I may just use the new one since I only had the SIP trunks setup and 2 extensions...

I remember a post back a ways that mentioned that it was simply not compatible with that version of Asterisk. I did not see a resolution.

Just for the commenters above that are having issues compiling chan_sccp

rpm remove asterisk-dev it is the 1.4 headers and are incorrect. once removed run: yum install asterisk16-dev
after the correct headers are installed chan_sccp will compile and load correctly.

Solved my own problem... Thought I'd share solution... Got this from an Asterisk 1.6 post...

Open 'sccp_pbx.c' with your favorite editor and change the line

#ifdef CS_AST_CONTROL_T38

to

#ifdef CS_AST_CONTROL_T38_PARAMETERS

Then recompiled no pronlem!!!

Hi all,
I am almost completely satisfied with my setup at the moment except that my remote extension (at home) is acting strange. When I dial an external number neither party can be heard. When I dial an internal extension, I can hear the other end but they cannot hear me. I changed the default context in sccp.conf in [general] and [lines] from 'sccp' to 'from-internal' and also tried 'default'. I had to do this to enable dialling out from the sccp extensions, otherwise I just got an engaged tone. With the contexts set to sccp, if I dial from a sip extension to the remote extension, audio works fine!? So I am assuming it has something to do with the contexts? Can anyone assist?
Here is part of my config:

[general]
servername = Asterisk ; show this name on the device registration
keepalive = 60 ; phone keep alive message evey 60 secs. Used to check the voicemail
debug = 1 ; console debug level. 1 => 10
context = from-internal
dateFormat = D/M/YA ; M-D-Y in any order. Use M/D/YA (for 12h format)
bindaddr = 192.168.0.153 ; 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 ;
allow=g729 ;

firstdigittimeout = 16 ; dialing timeout for the 1st digit
digittimeout = 4 ; more digits
digittimeoutchar = # ; you can force the channel to dial with this char in the dialing state
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
callwaiting_tone = 0x2d ; sets to 0 to disable the callwaiting tone
musicclass=default ; Sets the default music on hold class
language=en ; Default language setting
;callevents=no ; generate manager events when phone
; performs events (e.g. hold)
;accountcode=skinny ; accountcode to ease billing
;deny=0.0.0.0/0.0.0.0 ; Deny every address except for the only one allowed.
;permit=192.168.0.0/255.255.255.0 ; Accept class C 192.168.1.0
; You may have multiple rules for masking traffic.
; Rules are processed from the first to the last.
; This General rule is valid for all incoming connections. It's the 1st filter.

;localnet = 192.168.0.0/255.255.255.0 ; All RFC 1918 addresses are local networks
externip = xxxxxxxxxxxxx ; IP Address that we're going to notify in RTP media stream
;externhost = mydomain.dyndns.org ; Hostname (if dynamic) that we're going to notify in RTP media stream
; externrefresh = 60 ; expire time in seconds for the hostname (dns resolution)
dnd = on ; turn on the dnd softkey for all devices. Valid values are "off", "on" (busy signal), "reject" (busy signal), "silent" (ringer = silent)
echocancel = on ; sets the phone echocancel for all devices
silencesuppression = off ; sets the silence suppression for all devices
;callgroup=1,3-4 ; We are in caller groups 1,3,4. Valid for all lines
;pickupgroup=1,3-5 ; We can do call pick-p for call group 1,3,4,5. Valid for all lines
;amaflags = ; Sets the default AMA flag code stored in the CDR record
trustphoneip = no ; The phone has a ip address. It could be private, so if the phone is behind NAT
; we don't have to trust the phone ip address, but the ip address of the connection
;tos = 0x68 ; call control packets tos (0x68 Assured forwarding) [ASTERISK 1.2 only]
;rtptos = 0xB8 ; call rtp packets tos (0xB8 E.F.) [ASTERISK 1.2 only]
tos_sccp = cs3 ; default signaling TOS [ASTERISK 1.4 and above]
tos_audio = ef ; default audio TOS [ASTERISK 1.4 and above]
tos_video = af41 ; default video TOS [ASTERISK 1.4 and above]
cos_sccp = 3 ; default signaling COS [ASTERISK 1.6 and above]
cos_audio = 5 ; default audio COS [ASTERISK 1.6 and above]
cos_video = 4 ; default video COS [ASTERISK 1.6 and above]

;earlyrtp = none ; valid options: none, offhook, dial, ringout. default is none.
; The audio strem will be open in the progress and connected state.
private = on ; permit the private function softkey
privacy = full ; full = disable hints notification on devices, on = hints showed depending on private key, off = hints always showed
mwilamp = on ; Set the MWI lamp style when MWI active to on, off, wink, flash or blink
mwioncall = off ; Set the MWI on call.
;blindtransferindication = ring ; moh or ring. the blind transfer should ring the caller or just play music on hold
;protocolversion = 3 ; skinny version protocol. Just for testing. 0 to 11
;cfwdall = off ; activate the callforward ALL stuff and softkeys
;cfwdbusy = off ; activate the callforward BUSY stuff and softkeys
;cfwdnoanswer = off ; activate the callforward NOANSWER stuff and softkeys
;devicetable=sccpdevice ;datebasetable for devices
;linetable=sccpline ;datebasetable for lines
;nat=on ; Global NAT support (default Off)
;directrtp=on ; This option allow devices to do direct RTP sessions (default Off)
;allowoverlap=on ; Enable overlap dialing support. (Default is off)
;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
;jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
; sccp channel. Defaults to "no". An enabled jitterbuffer will
; be used only if the sending side can create and the receiving
; side can not accept jitter. The sccp channel can accept
; jitter, thus a jitterbuffer on the receive sccp side will be
; used only if it is forced and enabled.

;jbforce = no ; Forces the use of a jitterbuffer on the receive side of a sccp
; channel. Defaults to "no".

;jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.

;jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
; resynchronized. Useful to improve the quality of the voice, with
; big jumps in/broken timestamps, usually sent from exotic devices
; and programs. Defaults to 1000.

;jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a
; sccp channel. Two implementations are currently available
; - "fixed" (with size always equals to jbmaxsize)
; - "adaptive" (with variable size, actually the new jb of IAX2).
; Defaults to fixed.

;jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
;-----------------------------------------------------------------------------------

[devices]

;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
type = 7971 ; device type (see below)
autologin = 219 ; lines list. You can add an empty line for an empty button (7960, 7970, 7940, 7920)
description = Alex ; internal description. Not important
;keepalive = 60 ; set 0 to disable the keepalive check.
;tzoffset = +2
transfer = on ; enable or disable the transfer capability. It does remove the transfer softkey
park = on ; take a look to the compile howto. Park stuff is not compiled by default
;speeddial = ; you can add an empty speedial if you want an empty button (7960, 7970, 7920)
;speeddial = 1000,name ; speeddial number and name
speeddial = 200,Reception,200@default ; Number, Name (label), BLF Hint
speeddial = 201,Rohan,201@default ; Number, Name (label), BLF Hint
speeddial = 203,Maz,203@default ; Number, Name (label), BLF Hint
speeddial = 204,Prepress,204@default ; Number, Name (label), BLF Hint
speeddial = 208,Warehouse,209@default ; Number, Name (label), BLF Hint
speeddial = 209,Production,209@default ; Number, Name (label), BLF Hint
speeddial = 207,Digital,207@default ; Number, Name (label), BLF Hint
;serviceURL = ServiceURL1
;rtptos = 184 ; sets the default rtp packets TOS [ASTERISK 1.2 only]
tos_sccp = cs3 ; default signaling TOS [ASTERISK 1.4 and above]
tos_audio = ef ; default audio TOS [ASTERISK 1.4 and above]
tos_video = af41 ; default video TOS [ASTERISK 1.4 and above]
cos_sccp = 3 ; default signaling COS [ASTERISK 1.6 and above]
cos_audio = 5 ; default audio COS [ASTERISK 1.6 and above]
cos_video = 4 ; default video COS [ASTERISK 1.6 and above]
cfwdall = off ; activate the callforward stuff and softkeys
cfwdbusy = off
cfwdnoanswer = off
pickupexten = off ; enable Pickup function to direct pickup an extension
;pickupcontext = sccp ; context where direct pickup search for extensions. if not set it will be ignored.
pickupmodeanswer = on ; on = asterisk way, the call has been answered when picked up
; off = call manager way, the phone who picked up the call rings the call
dtmfmode = inband ; inband or outofband. outofband is the native cisco dtmf tone play.
; Some phone model does not play dtmf tones while connected (bug?), so the default is inband
imageversion = P00405000700 ; useful to upgrade old firmwares (the ones that do not load *.xml from the tftp server)
;deny=0.0.0.0/0.0.0.0 ; Same as general
;permit=192.168.1.5/255.255.255.255 ; This device can register only using this ip address
dnd = on ; turn on the dnd softkey for this device. Valid values are "off", "on" (busy signal), "reject" (busy signal), "silent" (ringer = silent) or user to toggle on phone
trustphoneip = no ; The phone has a ip address. It could be private, so if the phone is behind NAT
; we don't have to trust the phone ip address, but the ip address of the connection
;nat=on ; Device NAT support (default Off)
;directrtp=on ; This option allow devices to do direct RTP sessions (default Off)
;earlyrtp = none ; valid options: none, offhook, dial, ringout. default is none.
; The audio strem will be open in the progress and connected state.
private = on ; permit the private function softkey for this device
privacy = full ; full = disable hints notification on devices, on = hints showed depending on private key, off = hints always showed
mwilamp = on ; Set the MWI lamp style when MWI active to on, off, wink, flash or blink
mwioncall = off ; Set the MWI on call.
;setvar=testvar=value
device => xxxxxxxxxxxx ; device name SEP
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;

[lines]

;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;
id = 219 ; future use
pin = 1234 ; future use
label = Extension 219 ; button line label (7960, 7970, 7940, 7920)
description = Alex ; top diplay description
context = from-internal
incominglimit = 2 ; more than 1 incoming call = call waiting.
transfer = on ; per line transfer capability. on, off, 1, 0
mailbox = 219@default ; voicemail.conf (syntax: vmbox[@context][:folder])
vmnum = *97 ; speeddial for voicemail administration, just a number to dial
meetmenum = 700 ; this extension will receive meetme requests, SCCP_MEETME_ROOM channel variable will
; contain the room number dialed into simpleswitch.
cid_name = Alex ; caller id name
cid_num = 219
trnsfvm = 1000 ; extension to redirect the caller (e.g for voicemail)
secondary_dialtone_digits = 9 ; digits for the secondary dialtone (max 9 digits)
secondary_dialtone_tone = 0x22 ; outside dialtone
musicclass=default ; Sets the default music on hold class
language=en ; Default language setting
;accountcode=79501 ; accountcode to ease billing
echocancel = on ; sets the phone echocancel for this line
silencesuppression = off ; sets the silence suppression for this line
;callgroup=1,3-4 ; We are in caller groups 1,3,4. Valid for this line
;pickupgroup=1,3-5 ; We can do call pick-p for call group 1,3,4,5. Valid for this line
;amaflags = ; Sets the default AMA flag code stored in the CDR record for this line
;setvar=testvar2=value
line => 219
;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;;

Thanks,
Alex.

does confrn softkey working with your chan_sccp setup on trixbox ce?

-Larry

MulderLR