SUGGESTIONS Cisco 7970 Treasure Trove

therock112

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Finally an easy to use, no nonsense, download, install and setup type of asterisk....thanks Ward and team, a fantastic job.

I have been using a@h 2.5 for some time, I am of the group "dont mess with it if it aint broke"...

Unfortunately i bought a cisco 7970 AFTER I read wards article on this phone.....and I was curious about the quality of audio, the touch screen, the colour screen...you get the drift....

anyways, I get the new pbxiaf working nicely and got this phone talking (registering properly) to the asterisk server but I am not able to get the mwi indicator to lite up..

this appears to be a known bug with the cisco firmware, the firmware I am using on the 7970 is SIP70.8-2-2SR4S and I am able to successfully make and receive calls...BUT when a vmail is left in the mailbox the vmail indicator does not lite up (no lamp or indicator on screen)...

apparently the asterisk developers have fixed this issue some time back, adding a buggyciscomwi=yes in the sip.conf is the suggested fix

for the techie among us visit for more details: http://svn.digium.com/view/asterisk/trunk/channels/chan_sip.c?view=diff&r1=48982&r2=48983

now my question is:
1.
I have tried to add buggyciscomwi=yes to sip_custom.conf but that doesnt seem to make any difference, watching the sip debug messages in asterisk, I still notice the extra (0/0) being sent to the 7970 and of course the buggy cisco 7970 firmware returns a Warning: 399 Bad MWI NOTIFY back to asterisk....( wish cisco could simply open their eyes and churcn out rfc compliant code on their incredibly nice but unfortunately reduced to the functionality of a budgetone phone due to their low quality firmware!!)
2.
the other recommend way to fix this issue would be edit the chan_sip.c (http://bugs.digium.com/view.php?id=8575) and then recompile asterisk and other modules/code etc etc which I have no idea on how to go about in pbx in a flash.
If someone can shed some ideas/suggestions on the above, would be helpful to me and possibly to others too.
thanks.
 

jroper

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Hi

Probably a silly question, but you did do an asterisk reload after editing sip.conf?

Mods are better put in sip-custom.conf as sip.conf may get overwritten by a later version of FreePBX.

The source files for asterisk are all in /usr/src

Do the modifications recommended to the appropriate files in the /usr/src/asterisk directory, then: -

make clean
./configure
make
make install
Then reboot

and you should be good to go.

Yours

Joe
 

therock112

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Hi

Probably a silly question, but you did do an asterisk reload after editing sip.conf?

Mods are better put in sip-custom.conf as sip.conf may get overwritten by a later version of FreePBX.

The source files for asterisk are all in /usr/src

Do the modifications recommended to the appropriate files in the /usr/src/asterisk directory, then: -

make clean
./configure
make
make install
Then reboot

and you should be good to go.

Yours

Joe

as I mentioned I added the buggyciscomwi=yes in sip_custom.conf and saved the changes and rebooted the system, but that didnt seem to help....

I think I may have to go via option 2 and actually edit the chan_sip.c file, recompile asterisk and keep my fingers crossed....

will let you know. thanks for the quick reply by the way and the instructions on how to recompile asterisk on pbxiaf etc.
 

therock112

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Joe,

looks like I ran into some trouble following your instruction on recompiling asterisk.

before I started i issued a amportal stop followed by a service zaptel stop

and then went thru the make clean and ./configure etc etc

everything appeared to go thru fine, when all was done...I rebooted the pbx-in-a-flash box and now according to freepbx, asterisk is indicated in red in the bottom right hand side of the page...

I am not able to issue a asterisk -vvvvvr it whines about a "Unable to connect to remote asterisk (does /var/run/asterisk.ctl exist?)"

any ideas? or suggestions.....

was there more to re-compiling asterisk....was I supposed shutdown more service....or do i need to pass some specific parameters to ./configure ??
 

therock112

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*Resolved* Cisco 7970 MWI Issue

1. adding buggycisco=yes in the sip_custom.conf did not appear to do much for me....maybe I made a goofup somewhere.....

2. editing the chan_sip.c file did do the trip...this of course required recompiling asterisk etc.....

On my cisco 7970 phone, I upgraded to firmware version 8.3(1)
so far the phone is operating properly....

when a vmail is left in the mailbox, the red lamp is ON, upon deleting the vmail, the lamp is off as it should be....

I will keep playing with it and see what other "weirdness" rears its ugly head....

thank you all for the help...
 

wardmundy

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Can you post your chan_sip.c file just so others can use it. Thx.
 

gdchongris

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Sccp?

Have you thought about using SCCP? I have it running very well on PBX-IAF right now

1. adding buggycisco=yes in the sip_custom.conf did not appear to do much for me....maybe I made a goofup somewhere.....

2. editing the chan_sip.c file did do the trip...this of course required recompiling asterisk etc.....

On my cisco 7970 phone, I upgraded to firmware version 8.3(1)
so far the phone is operating properly....

when a vmail is left in the mailbox, the red lamp is ON, upon deleting the vmail, the lamp is off as it should be....

I will keep playing with it and see what other "weirdness" rears its ugly head....

thank you all for the help...
 

therock112

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Can you post your chan_sip.c file just so others can use it. Thx.

absolutely..

Unzip in /usr/src/asterisk/channels folder and recompile asterisk....

<ward, I tried attaching the zipped ".c" file but the zipped file size is over 100kb, and it appears your forum wont allow this....>
 

therock112

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Have you thought about using SCCP? I have it running very well on PBX-IAF right now

to be honest, I have never played with sccp....its always been sip

I dont even know where to start...perhaps a abc type instruction or tutorial you could suggest......

what are the pro's and con's of using sccp in comparison to SIP???
 

gdchongris

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SCCP=great-imo

I first started messing with trixbox a year ago because I saw a cisco 7960 on the show 24 on fox and thought it was awesome, so I bought one on ebay like an idiot. A year later... well now I do this for a living haha.

At first I used SIP but little things annoy me. Like how slow the menus are (I don't know if this is true on the 7970 like you have but on my 7960 and 7940 they are slow, like a bad cell phone) and things like the web browser not being able to do as much... so I ventured out and got into SCCP. When I had trixbox you could install it as a package and it did all the dirty work for you (make, make compile, etc.) so I had to re-invent the wheel so to say when I switched to PBX-IAF. It's not hard though...

So here's what you do. (I have a Cisco 7960/ Cisco 7940 (firmware P00307020300), Cisco 7912 (I forget what firmware I'll have to look this up but MWI call parking and everything works), Cisco IP Communicator softphone and an ATA-186 all running SCCP with my PBX-IAF as I'm typing this)

First, get to your box's command prompt. It can be ssh, you can be in front of it, it doesn't matter. Hell you can copy-paste this if your in ssh (that's what I'd do)

I'm doing this off of memory right now so if I mess this up message me and I can send you new instructions when I'm actually in front of a box to mess with it

1. 'wget http://superb-east.dl.sourceforge.net/sourceforge/chan-sccp-b/chan_sccp_20071130.tar.gz' Download chan_sccp-b
2. 'tar xzvf chan_sccp_20071130.tar.gz' I think that's the command, I'm no linux expert
3. 'cd chan_sccp_20071130' Change to the untarred directory
4. 'make && make install' Tells your box to build chan_sccp and install it - it will ask you if you want to compile call parking, call pickup and realtime functionality - I answered yes to all of them. When it finishes it will say install error 1 along with some cp/error stat cannot create sccp.conf message - this is fine chan_sccp installed but you just need to go make the config file
5. 'nano -w /etc/asterisk/modules.conf' Pick a place anywhere under the 'autoload=yes' line and add the following line: 'noload => chan_skinny.so', press Ctrl-X and accept changes - this tells asterisk to not load the crappy chan_skinny stack because if chan_sccp and chan_skinny are running your phones won't know what to do
6. 'nano -w /etc/asterisk/sccp.conf' Now that you are back at the command prompt, we're going to make the sccp.conf file. I've attached sccp.conf in a zip file here, open it up on your computer and paste it into your ssh session, and then Ctrl-X and save your changes. Modify the default config file to match your server. Customize each setting to your liking, SCCP lines mean extensions in the FreePBX world and the devices mean your actual phones. You can set autologin=131,132,133,134 and as long as you have lines 131, 132, 133 and 134 set up one phone can have all those extensions logged in as once. I use this because 131 is my main number, 132 dials out of my PSTN trunk, and 133 and 134 are voicemail boxes for my IVR. I love this functionality it is awesome. Once you define a line in sccp.conf, go to FreePBX > Extensions, add a CUSTOM extension, set the extension number and display name, and under the DIAL field type 'SCCP/131' where 131 is your line/extension number. You can also set up voicemail here if you want said extension to have voicemail. Keep in mind once you apply your changes to sccp.conf you must do an 'amportal restart' in order for SCCP changes to take effect.
7. 'nano /tftpboot/OS79XX.txt' in the first line type the name of your firmware version and save/exit
8. Make sure your .sbn, .loads and .bin firmware files are in your tftpboot directory.
9. I have enclosed an SCCPxxxxxxxxx.cnf.xml file in a ZIP. Make these for each of your SCCP devices that you define in your .conf and change all the '10.222.34.22' IP addresses to match the IP of your PBX-IAF box. This makes it so where the phones know where to go to register
10. 'amportal stop' Kill Asterisk
11. 'amportal start' Start Asterisk which will now have chan_sccp and your sccp.conf loaded in

I could have missed a step or two. Hardly slept last night. Don't let this intimidate you as I let it for awhile... it's not that bad and the benefits are well worth it.

to be honest, I have never played with sccp....its always been sip

I dont even know where to start...perhaps a abc type instruction or tutorial you could suggest......

what are the pro's and con's of using sccp in comparison to SIP???
 

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gdchongris

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a lot at once

If this confused you I am in the process of making a video tutorial that I'll probably post on youtube or something that can walk you through how to do it.

I first started messing with trixbox a year ago because I saw a cisco 7960 on the show 24 on fox and thought it was awesome, so I bought one on ebay like an idiot. A year later... well now I do this for a living haha.

At first I used SIP but little things annoy me. Like how slow the menus are (I don't know if this is true on the 7970 like you have but on my 7960 and 7940 they are slow, like a bad cell phone) and things like the web browser not being able to do as much... so I ventured out and got into SCCP. When I had trixbox you could install it as a package and it did all the dirty work for you (make, make compile, etc.) so I had to re-invent the wheel so to say when I switched to PBX-IAF. It's not hard though...

So here's what you do. (I have a Cisco 7960/ Cisco 7940 (firmware P00307020300), Cisco 7912 (I forget what firmware I'll have to look this up but MWI call parking and everything works), Cisco IP Communicator softphone and an ATA-186 all running SCCP with my PBX-IAF as I'm typing this)

First, get to your box's command prompt. It can be ssh, you can be in front of it, it doesn't matter. Hell you can copy-paste this if your in ssh (that's what I'd do)

I'm doing this off of memory right now so if I mess this up message me and I can send you new instructions when I'm actually in front of a box to mess with it

1. 'wget http://superb-east.dl.sourceforge.net/sourceforge/chan-sccp-b/chan_sccp_20071130.tar.gz' Download chan_sccp-b
2. 'tar xzvf chan_sccp_20071130.tar.gz' I think that's the command, I'm no linux expert
3. 'cd chan_sccp_20071130' Change to the untarred directory
4. 'make && make install' Tells your box to build chan_sccp and install it - it will ask you if you want to compile call parking, call pickup and realtime functionality - I answered yes to all of them. When it finishes it will say install error 1 along with some cp/error stat cannot create sccp.conf message - this is fine chan_sccp installed but you just need to go make the config file
5. 'nano -w /etc/asterisk/modules.conf' Pick a place anywhere under the 'autoload=yes' line and add the following line: 'noload => chan_skinny.so', press Ctrl-X and accept changes - this tells asterisk to not load the crappy chan_skinny stack because if chan_sccp and chan_skinny are running your phones won't know what to do
6. 'nano -w /etc/asterisk/sccp.conf' Now that you are back at the command prompt, we're going to make the sccp.conf file. I've attached sccp.conf in a zip file here, open it up on your computer and paste it into your ssh session, and then Ctrl-X and save your changes. Modify the default config file to match your server. Customize each setting to your liking, SCCP lines mean extensions in the FreePBX world and the devices mean your actual phones. You can set autologin=131,132,133,134 and as long as you have lines 131, 132, 133 and 134 set up one phone can have all those extensions logged in as once. I use this because 131 is my main number, 132 dials out of my PSTN trunk, and 133 and 134 are voicemail boxes for my IVR. I love this functionality it is awesome. Once you define a line in sccp.conf, go to FreePBX > Extensions, add a CUSTOM extension, set the extension number and display name, and under the DIAL field type 'SCCP/131' where 131 is your line/extension number. You can also set up voicemail here if you want said extension to have voicemail. Keep in mind once you apply your changes to sccp.conf you must do an 'amportal restart' in order for SCCP changes to take effect.
7. 'nano /tftpboot/OS79XX.txt' in the first line type the name of your firmware version and save/exit
8. Make sure your .sbn, .loads and .bin firmware files are in your tftpboot directory.
9. I have enclosed an SCCPxxxxxxxxx.cnf.xml file in a ZIP. Make these for each of your SCCP devices that you define in your .conf and change all the '10.222.34.22' IP addresses to match the IP of your PBX-IAF box. This makes it so where the phones know where to go to register
10. 'amportal stop' Kill Asterisk
11. 'amportal start' Start Asterisk which will now have chan_sccp and your sccp.conf loaded in

I could have missed a step or two. Hardly slept last night. Don't let this intimidate you as I let it for awhile... it's not that bad and the benefits are well worth it.
 

therock112

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wow gdchongris

thank you for taking the time to type this up...

I am going to give this a shot..

currently, I have the previously mentioned sip 8.3 firmware on the 7970...obviously I am going to have to reflash it with the sccp firmware....is this correct?

what are the benefits of the sccp prtocol over sip in regards to the 7970? do things like call-transfer work? call waiting? line sharing? other fancy stuff asterisk provides...do all of these work?

I really havent had a chance to test to see what functionality is not available in the current sip firmware on the phone.....I am assuming you have tried the sip firmware on your phone with asterisk....and met with some roadblocks...!!

thanks again.
 

therock112

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At first I used SIP but little things annoy me. Like how slow the menus are (I don't know if this is true on the 7970 like you have but on my 7960 and 7940 they are slow, like a bad cell phone) and things like the web browser not being able to do as much... so I ventured out and got into SCCP. When I had trixbox you could install it as a package and it did all the dirty work for you (make, make compile, etc.) so I had to re-invent the wheel so to say when I switched to PBX-IAF. It's not hard though...

actually I am running a spa2102, 3 polycom phones 2 x 501 and a 301, a cisco 7960 and a 7970.....the 3.3xx firmware on the 7970 is nice...matter of fact 7970 in general is nice, the menus are faster than the 7960....i would imagine the cisco guys fixed some of the UI interface in the 7960 but goofed up on the core working of the phone itself especially when it comes to sip!!
 

gdchongris

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Yes you will have to re-flash it with SCCP firmware. Let me know if you need help doing this. Yes I met roadblocks with SIP firmware and well... let's just say... roadblocks piss me off!

I currently use the autologin with multiple lines feature (I have not tried line sharing but if I remember when I get home I'll try it and report back), call parking, call transferring, call waiting, message waiting indicator and a few other things. Also I have two Polycom IP330's and various IAX softphones that I run on my system over SIP and IAX2 protocols and they all inter-connect fine (I can dial my SCCP phones from my SIP/IAX2 devices and vice-versa without issues). For awhile the message waiting indicator would not work on my 7912 but when I changed from TB to PBX-IAF and put in chan_sccp and upgraded my 7912's firmware it works flawlessly.

SCCP vs. SIP

Pros
Call Transfer does work
Single-button call parking
Web browser ACTUALLY works correctly (at least in comparison w/ the 7960)
Single-button Do Not Disturb
Auto-login of multiple lines on one device
Option for secondary dial tone (i.e. when you press 9, this is good for people who are used to traditional press-9-to-get-out PBX or key systems)
Server-side speed dial provisioning (you can program your speed dial buttons on the server so you can auto-provision and not have to worry about losing your speed dial buttons when you reboot)
(and a bunch more that I'm forgetting right now... google it a lot of people use chan_sccp but they all use it with plain vanilla asterisk which I know nothing about... I was a TB user but now proudly a PBX-IAF user so I built the bridge for users of Cisco phones)
Ability to use the 7914 expansion/attendant console (cannot be used at all with SIP images)
Ability to use 7920, 7921? and 7935/36 conference phones (cannot be used at all with SIP images)

Cons
Have to re-flash the SIP phones/devices (but not hard)
Conference call/merging calls from phone screen - I had this in the SIP image but I cannot find it in SCCP... I have to do research on this one

Things I'm not sure of (because I haven't had time to try)
Line auto-answer/intercom (there is a way around this but I have not pursued it)
Line sharing


Always a pleasure to help. I had to learn all this the hard way because people either were not doing this or were doing it with plain Asterisk so A@H/TB/PBX-IAF users were left in the dark...
wow gdchongris

thank you for taking the time to type this up...

I am going to give this a shot..

currently, I have the previously mentioned sip 8.3 firmware on the 7970...obviously I am going to have to reflash it with the sccp firmware....is this correct?

what are the benefits of the sccp prtocol over sip in regards to the 7970? do things like call-transfer work? call waiting? line sharing? other fancy stuff asterisk provides...do all of these work?

I really havent had a chance to test to see what functionality is not available in the current sip firmware on the phone.....I am assuming you have tried the sip firmware on your phone with asterisk....and met with some roadblocks...!!

thanks again.
 

gdchongris

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the firmware version? I actually don't have a 7970 yet... I'm ordering one on monday.
 

wardmundy

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I'd have to say that the 7970 could be the perfect phone if it weren't for Cisco. We wrestled with it over a year ago on Nerd Vittles and finally got it working pretty well. Then I did some more work and actually got XML weather and news reports working as well as an AsteriDex lookup function from a live MySQL database which is extremely slick. In fact, it's the way a perfect phone should work.

Having said all of that, I have resisted the temptation to say or write anything nice about this phone because Cisco is such a crappy player in the SIP and open source marketplace. Someone else will come along like Aastra or Grandstream and finally do it right. Then we'll really have something.


In the meantime, we show off the 7970 to all of our drooling nerd friends always ending the conversation with... "You don't want one!" :cool:
 

therock112

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I'd have to say that the 7970 could be the perfect phone if it weren't for Cisco. We wrestled with it over a year ago on Nerd Vittles and finally got it working pretty well. Then I did some more work and actually got XML weather and news reports working as well as an AsteriDex lookup function from a live MySQL database which is extremely slick. In fact, it's the way a perfect phone should work.

Having said all of that, I have resisted the temptation to say or write anything nice about this phone because Cisco is such a crappy player in the SIP and open source marketplace. Someone else will come along like Aastra or Grandstream and finally do it right. Then we'll really have something.


In the meantime, we show off the 7970 to all of our drooling nerd friends always ending the conversation with... "You don't want one!" :cool:

no doubt about it, the 7970 is a slick phone, too bad its manufacturer feels so threatened of open source, asterisk etc etc

if you are able to share some code in terms of pulling config from a mysql dbase, that would be fantastic....I am headed that direction anyhow, it might save some hair pulling, kicking and screaming.....(i am a calm type of person!!)
 

wardmundy

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Against my better judgment, here is a link to all of the Cisco 7970 goodies I use and some that I've never looked at including the SugarCRM stuff.

Unzip cisco7970.zip recursively and drop into /var/www/html/cisco directory. Chmod everything to 775, and chown to asterisk:asterisk. Edit services/DBConnect.inc to include your username and password for AsteriDex. Edit each of the PHP files to include the IP address of your PBX in a Flash server. None of this worked for me for months. Then all of a sudden one day, stuff started working. Who knows why?? I don't have any idea. As of today, the Asteridex lookups work, the Weather report works for Charleston (you can change it), the News feeds from Yahoo work, and HELP works. Enjoy.
--Love, Santa

P.S. Use at your own risk!!! :cool:
 

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