I’ve seen posts on various forums that say that a Cisco phone with SCCP firmware can, indeed, be used on an Asterisk system. Is this true, and is this a smart thing to do? Or, should I just attempt to flash SIP firmware? I’m beginning to wish I had bought a Grandstream phone or something … such a nightmare. I already have the firmware. My question is, is it a good idea to setup SCCP functionality on asterisk, and is it easy to do so? Also, which has more features?
There is a project called chan_sccp2 that adds terrific support for sccp phones. The FreePBX GUI does not support the module so you would have to configure the .conf files directly.
ahhh, well then I’m not sure I’d be able to figure it out at this stage. That definitely answers my question better than other posts I’ve seen though, so thanks!
What I mean to say is, I could probably figure out the actual chan_sccp2 file, but where do I put it? How do I configure an extension to use it? And, is SCCP better than SIP (aside from saving the nightmare of flashing the firmware)?
All configs are in /etc/asterisk. When you install chan_sccp2 it will load a template file.
The sccp2 project has a ton of cool features, and the sccp software is better on the phone itself. It even supports shared line appearances!
Installing is not a big deal but you have to build Asterisk from source code to get it to work.
Join the chan_sccp mail list, read the documentation.
Also, don’t forget to put your sccp extensions in the from-internal context so the extension have access to the FreePBX dial plan. Just set aside a range of extensions for the sccp phones.
So, you’re saying I couldn’t just put chan_sccp into my existing elastix installation? Wow, I think this is going to be way over my head …
Sure you could put it in your Elastix, I don’t know if Elastix has the development headers or not in their repo but the safest path is to make sure you have the current Kernel headers, download Asterisk from Digium, chan_sccp2 from sourceforge and then run the ./configure/make clean/make menuselect/make install process. It won’d disrupt your current settings or FreePBX.
why would I need Asterisk from Digium if I already have an Elastix distro up and running? Also, how would I configure an extension for it, if there’s no menu options in FreePBX?
why would I need Asterisk from Digium if I already have an Elastix distro up and running?
Because a distribution like Elastix distributes compiled binary. To build a module it needs to link to the source code. When developers build a distribution they decide what modules to include. If current headers are provided it is possible to complie and link a driver but this is tricky. You need the source code from Digium so you can compile the chan_sccp2 module, a third party add-in
Also, how would I configure an extension for it,
You setup extensions in the configuration file. By setting the sccp extension context to “from-internal” it will have access to all FreePBX features and extensions. You will have to manually resolve contentions and make sure the dial plans don’t overlap. I would set aside a range of extensions for sccp.
I am currently in the “beta” testing team and can tell you first hand that FreePBX is not compatible with the current more recent build of chan_sccp-V4.0 trunk (build 3120) - It has more to do with the dial plan and bugs with features than with overall stability. (no memory leaks - just feature and FreePBX dial plan bugs)
Right now the dev team is mostly on leave or “holiday” to deal with their real lives or personal issues. Since it is FREE, there is really no obligation for devs to continue development to meet any sort of time lines. Makes sense.
Honestly, right now since we have so many Cisco phone assets (20) it is hard for my company to move away from chan_sccp and at the mercy of the devs to finish. I would advise STRONGLY against using chan_sccp V4.0 or the latest Asterisk/FreePBX if you are looking to deploy within 2011 or even mid 2012 as there just aren’t enough devs to make it happen. (1 currently active out of 4?) Not that it isn’t good code or useful, just incomplete with no real defined completion date in view.
What chan_sccp development really needs is more devs willing to pitch in and fix the issues and it just isn’t happening. If you or anyone in the FreePBX community can help them out, PLEASE do so… they really need more talent to spread out all the work that needs to be done. If I was a programmer I would definately contribute all that I could.
In Short…
My advice is if you don’t have ties to Cisco sccp, RUN! RUN AWAY! Cisco is EVIL and they will not make it easy for you to use something other than CM.
You can try the chan_sccp 3.0 trunk with asterisk 1.6, but I have seen many stability issues come up with that recenlty also, with only 1 dev working on it, it doesn’t look good for the near future.
I am not giving up on chan_sccp development, but have to admit I am looking for another solution; either buying new phone assets (sip) or maybe another PBX distro that plays nicer with the current builds of chan_sccp.
PS: Cisco SIP firmware is pure crap. No/little functionality and ZERO support. (there is better support documentation on how to hang the phone on the wall) If you are going to use SIP, buy 1 phone and “try it out” to see if the sip software on the phone meets your needs.
Bob - I would like to have a discussion with you as I was under the understanding the chan_sccp3 would build fine against Asterisk 1.8.
I want to understand what is going on with the project so I can speak clearly to it when FreePBX users ask questions.
If chan_sccp developers are willing to do a few small things to .conf file layout it would be very simple to add support in FreePBX for the channel type. If the chan_sccp project is willing to provide RP/M’s and keep them built against the distro kernel and Asterisk headers I would be willing to get behind and effort to build the modules.
I have several chan_sccp3/Asterisk 1.6/FreePBX 2.9 systems on CentOS without issue. They were built by hand.
It has always been my opinion that Asterisk/chan_sccp and FreePBX are truly “Open Call Manager”. It’s an opportunity waiting to happen. Cisco is out abusing customers to make them purchase CUCM updates in order to keep Smartnet on the handsets. This is on 5 year old systems. An open source Alternative with kick Mr. Chambers and team right where it hurts.
If you have a few minutes to chat, drop me your contact info via a PM.
Regards…Scott
SkykingOH: I don’t see anywhere to do an IM on here or even how to contact a user. my email is [email protected] if you need to reach me. I am in USA/Indiana.
Chan Sccp discussion and development can be located here:
Issues specfic to dial plan topic:
http://sourceforge.net/mailarchive/forum.php?thread_name=CA3D12A92535F248B7292C12C58076560128A673%40companyweb&forum_name=chan-sccp-b-discussion
http://sourceforge.net/mailarchive/forum.php?thread_name=CA3D12A92535F248B7292C12C58076560128A6F5%40companyweb&forum_name=chan-sccp-b-discussion
http://sourceforge.net/mailarchive/forum.php?thread_name=CA3D12A92535F248B7292C12C58076560128A88B%40companyweb&forum_name=chan-sccp-b-discussion
http://sourceforge.net/mailarchive/forum.php?thread_name=CA3D12A92535F248B7292C12C58076560128AB0A%40companyweb&forum_name=chan-sccp-b-discussion
Read historical messages:
https://sourceforge.net/mailarchive/forum.php?forum_name=chan-sccp-b-discussion
Bug reports:
https://sourceforge.net/tracker/?group_id=186378&atid=917045
Mailinglists:
https://sourceforge.net/mail/?group_id=186378
Almost everything in chan-sccp-b is discussed via the mailinglist. The forums are not used (much)
The development for compatibility with FreePBX definately needs a few FreePBX experts to look at the issues.
Note: I am not trying to “steal” this thread form the OP so staying relevant, I would say until the compatibility issues are sorted out with the latest FreePBX and latest chan_sccp 4.0, I would postpone deployment or try asterisk 1.6 and the 3.0 branch. Right now my interest is clearly on asterisk 1.8+ and the chan_sccp 4.0 branch since it offers more features relative to Cisco CM and future support. Support for 1.6 is fading out and for a company trying to stay on the edge, we need to innovate and move forward with the latest tech if at all possible.
Bob - Thanks for all of the info. I guess at this point I’ll flash the SIP firmware. Don’t feel bad about stealing the thread; all of your information has been useful. I wish I could help program SCCP functionality into FreePBX, but I wouldn’t even know where to begin; programming definitely isn’t my thing. Part of me kind of wishes I had bought a Grandstream phone; I may still sell the Cisco and get one if the SIP firmware is that bad. I did get a good deal on it though, so I could always keep it in hopes that someone actually makes good SCCP compatibility. Keep commenting on here if you like, I’m happy to read it all.
Also, Bob, you mentioned other PBX distros that would handle SCCP? I looked around, and was unable to find any. What have you found?
Well since this is a FreePBX forum a dont think its right for me to start discussing other products in detail.
All asterisk distributions are potentially compatible with chan_sccp.
Just so you understand… And you may already.
Linux=OS (centos, debian, redhat, etc.)
Asterisk=application (not tied to any pbx gui)
Freepbx=GUI for asterisk (really much more and adds many features)
Chan_sccp= driver for cisco phone opperability with asterisk
If you are an expert at asterisk, you may perfer not to use a gui such as freepbx, but this does not make it friendly in any way.
So really all you need to make chan_sccp work is the OS, application and the chan_sccp driver and the ability to wget and compile the sources. Right now the problem I am having is relative to the dialplan freepbx creates does not check to make sure the chan_sccp channel is open before sending the call. So I am stuck with the chan_sccp devs saying this is a freepbx problem, not a chan_sccp problem. My hunt for another solution will most likely lead me back here to post a help thread on adding chan_sccp CHANAVAIL checks into the FreePBX dialplan. Perhaps this is something already in the works. (this is not an issue with sip or dundi)
I really really love freepbx and all that it stands for, so moving to the other “fork” really isnt my plan.
I think if you have an option right now to not use Cisco, I would go that route. Plenty of quality SIP products available.
Bob,
You can discuss anything you want in these forums.
Can you please post dialplan of how the status check is accomplished?
I can tell you that FreePBX works with H.323, MGCP and SIP so I want to see exactly what is going on.
I have also tried chan_sccp with FreePBX and it seemed to work so I would like to know if this is a recent change.
You can recreate the problem by registering a sccp extension and then calling ANY unregistered extension on FreePBX. CLI will display that the call went to VM, but you will hear no audio followed by a crash when you hangup.
Where exeactly this is happening I don’t know, I looked at the dialplan myself and could not figure it out either as I am just not that good. It seems the call is routing correctly to a closed audio channel? Not sure if this helps.
I think I was told by Diederik (chan_sccp dev) that it was the FreePBX dialplan, but he did not specify exactly where in the dialplan.
"I tried reproducing the problem running a plain asterisk 1.8.7.1 without
incident. I’m not running FreePBX and not so many people on the List are
(as far as i know).
Normally CHANAVAIL should receive information that the phone is not
registered and never actually DIAL the extension, that doesn’t happen
which is also very suspect."
Diederik
Now my argument with that is, it doesn’t route the call to the extension, it goes to the VM box, but with a closed audio channel. Doesn’t make sense. Perhaps it is opening the audio channel to the unregistered extension instead of the VM box and when the call is ended asterisk doesnt know how to handle it and crashes?
I know this doesn’t help much, and all the backtraces dont show any relevant info, so at this point we need a FreePBX expert to say “AH HAA” I see where this is happening! - Specific only to chan_sccp, since sip and dahdi seem to not have this problem. I can call sip to sip and sip to sccp and dahdi to sip or sccp and its fine … any call from sccp to unregistered ext results in no audio and then crashes asterisk on hangup.
This thread gives the most info from backtrace and CLI
http://sourceforge.net/mailarchive/forum.php?thread_name=CA3D12A92535F248B7292C12C58076560128A67E%40companyweb&forum_name=chan-sccp-b-discussion
*This shows a call from sccp ext 4021 to unregistered sip ext 4444
I have to tell you Bob, none of this makes much sense. FreePBX does not control the audio or use the voicemail module in any non-standard way.
I assume the sccp extension was added as a custom extension with a dial string of sccp/ext# ?
Any time I try and discuss a sccp FreePBX module in the mail list it never generates much interest. Never understood the prejudice, why they can’t see the possibilities.
Anyway, if I am bored over the Thanksgiving holiday I will try and reproduce.
Regards…Scott
So I tried another distro of FreePBX called Elastix and the problem seemed to have gone away. I installed and setup the sccp phones right away and tested. No crash and everything worked the way it should. Then I updated all the modules to the Latest versions and the crash came back.
So somewhere in the updates is the problem but I do not know exactly which update does it at this point. I have reinstalled and am going to update one by one and test. My guess is that it is the Core update to 2.9.0.7 is what creates the problems. In version 2.8.1.4 there is no problem.
Also Diedrik is back from his leave, so perhaps I will email him this thread and see if I can get him to take a look. Chances are though since this problem happens AFTER a Core update of FreePBX modules, he will agree that something is amiss in the FreePBX code and it is best that he uses his time to resolve bugs and add features and let the FreePBX devs do their part.
I cant stress how important having the advanced sccp finctionality in FreePBX is, it makes FreePBX one of the most important servers in my facility and without it we would be forced to continue with Cisco CM. So for now I will lock down the production machine to 2.8.1.4 and Asterisk 1.8.7.0 and move forward on the test machine.
[BTW, Elastix is definately worth taking a look at if you want all your communications applications in 1 server. Harnessing HylaFax,FreePBX, IM, Video conferencing and more all into 1 box definately has my interest.]
So it’s not the distro, it’s just 2.9.
That should not be that hard to trace down.