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Old 12-14-07, 06:58 AM
therock112 therock112 is offline
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Posts: 140
Cisco 7970 Treasure Trove
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/...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.

Last edited by therock112 : 12-15-07 at 12:28 PM. Reason: issue resolved
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  #2  
Old 12-14-07, 07:22 AM
jroper jroper is offline
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Posts: 3,333
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
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  #3  
Old 12-14-07, 01:54 PM
therock112 therock112 is offline
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Posts: 140
Originally Posted by jroper View Post
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.
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  #4  
Old 12-14-07, 03:23 PM
therock112 therock112 is offline
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Posts: 140
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 ??
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  #5  
Old 12-15-07, 12:27 PM
therock112 therock112 is offline
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Join Date: Dec 2007
Posts: 140
*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...
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  #6  
Old 12-15-07, 01:07 PM
wardmundy wardmundy is offline
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Can you post your chan_sip.c file just so others can use it. Thx.
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  #7  
Old 12-15-07, 01:17 PM
gdchongris gdchongris is offline
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Location: Dallas, TX
Posts: 23
Sccp?
Have you thought about using SCCP? I have it running very well on PBX-IAF right now

Originally Posted by therock112 View Post
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...
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  #8  
Old 12-15-07, 01:26 PM
therock112 therock112 is offline
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Posts: 140
Quote:
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....>

Last edited by therock112 : 12-15-07 at 02:27 PM.
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  #9  
Old 12-15-07, 01:27 PM
therock112 therock112 is offline
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Join Date: Dec 2007
Posts: 140
Originally Posted by gdchongris View Post
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???
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  #10  
Old 12-15-07, 01:58 PM
gdchongris gdchongris is offline
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Join Date: Nov 2007
Location: Dallas, TX
Posts: 23
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.ne...0071130.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.

Originally Posted by therock112 View Post
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???
Attached Files:
File Type: zip sccp.conf.zip (3.0 KB, 159 views)
File Type: zip SEP003094C3B13E.cnf.xml.zip (1.3 KB, 151 views)
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