SUGGESTIONS Cisco 7970 Treasure Trove

[wtodd]

no need to use sccp. I have used both and sip seems to work really well once "figured out"

try buggymwi=yes in sip custom conf

that works for me on 7970's and 7975's

 

[ksDevGuy]

Successful 7971G user for a couple years now, love the phone (and so does anyone that sees it, even next to a 7960 or the new Aastra 57iCT).

I am one of the earliest guys contributing to the 7960G & 7970/7971G how-to pool of facts on voip-info.org that later showed up everywhere else in How-To's (e.g. Kerry's at TB, etc.)

Have been using firmware v8.2.1 on our 7971's with good success on all basic functions (add the buggymwi=yes entry to sip_custom.conf). Hold, transfer, conference, cfwdall, etc.

Items I have been unable to figure out (and haven't tried in a long time -- as Ward said, until you get it cookie-cutter dialed in the phone can wear you out):

1) Dialplan XML -- getting it to load and enforce as it does so easily on the 7960G's (works partially, and partially does not)?

2) Busy Lamp Field (BLF) & Shared Line Appearance (SLA)?

3) Getting DTMF tones rather than the flat tone on the users side (DTMF works, just doesn't sound right)

4) Background image good color stratification in Photoshop (I can make them, they just don't look esp. slick -- i.e. converting photos to 12-bit color, how-to nicely)?

5) External lamp field add-on?

6) softkey.xml -- to support programming the bottom line of softkeys

Anyone, having tackled these items to any degree I'd love to hear from.

Equally, anyone needing working config file examples feel free to post - I'd be glad to share them (as I have in years past on TB forums).

ksDevGuy

 

[TheShniz]

Well I can answer the wallpaper question... it's the exact same way as the 794x/6x's, just with different dimensions, path, and colors.

You should be uploading to:
tftpboot/Desktops/320x212x12

As such, you can notice the different resolution... this is because with the 794x/6x's, the wallpaper is shown below the title bar and above the bottom bar, BUT with the 797x's it is show BEHIND those bars via transparency. (the thumbs are 80x53)

Don't forget to update the List XML same as before...


:: updated ::


Would you share your config? I'm always interested to see how others have programmed these & compare notes...

I forget which firmware now, but I remember having problems with DND not actually sending callers straight to voicemail, instead what it did was simply mute the ringer. We use firmware 8.3.1 if I remember right.

 

[ksDevGuy]

<g> You beat me to the punch. Was about to edit my post about wallpaper. Noticed it works & found my config/notes on it, just getting 12-bit color to look good is another a matter entirely!

A couple of my threads from TB in 2006 on the 7970/7971 phone -- my configs are still pretty close to this with minor cleanups:

http://www.trixbox.org/forums/trixbox-forums/trixbox-endpoints/cisco-7971g-ge-sip-v8-0-3

http://www.trixbox.org/forums/trixbox-forums/trixbox-endpoints/7971-7961-7941-sip-v8-x-support

ksDevGuy

 

[therock112]

finding anythig usefull on Cisco's website requires a CCNA first it seems, the following url has some useful info about image editing etc as well:

http://www.cisco.com/en/US/docs/voi...1/sip/english/adminstraton/guide/7970cst.html

 

[TheShniz]

getting 12-bit color to look good is another a matter entirely!

Most definately yes... color matching is definately non existant, this is the #1 challenge here (trial & error).

finding anythig usefull on Cisco's website requires a CCNA first it seems

lol, finding ANYTHING useful on Cisco's WebSite with ANY level of clearance requires a PhD (much distain and gnashing of teeth)... now if you want to know how to plug the phone into the wall, that documentation is superb .


I'll agree with your claim of Aastra making the best phones for Asterisk is definately an overstatement. While they are a very good starting point, they are by no means the best. There is no best, different phones for different people.

:: updated ::

Looking back through the thread, I'm interested to see/hear more about what you're doing in php/idle therock... sounds interesting.

 

[Tenshi]

I had high hopes to get my 7970 up an running in SCCP, but seems so far like a complete failure... the phone registered, but with my piaf, however it was nowhere to be seen for Asterisk and a fast busy on the phone as soon as I pressed a number!

It's a shame such a cool phone is so hard to make it work correctly!

Should we create a wiki for it?

 

[TheShniz]

^^^ Might want to try disable-iptables, otherwise more information is required.

 

[Tenshi]

yeah, even disabled it didn't work...

Changed back to sip firmware, changed my extension back and everything works fine

I guess I should have followed the usual: if it ain't broke...

 

[krzykat]

I have never played with these phones, but it looks like I will be soon. Too bad they don't make them easier to integrate.

Ward, do you think its possible to use your weather system such that when there is a "weather alert" that is posts to the phone automatically? People in offices, don't often know of the weather, and for them to know without asking, sounds pretty cool. For example, right now, I'm in South Florida, and I don't know if the hurricane is still coming my way. I need to turn on TV or go to www.weather.com to find out. Just thinking out loud here, but if my phone told me that the storm is on its way, so I know how to prepare, would be pretty cool.

Not to mention, I think could be a nice sales pitch for a PIAF system to a business client.

 

[wardmundy]

Great idea. I'll add to the Weather Wish List once we deal with...

 

[gman]

I just wanted to mention that I have two flavors of Call Manager running inside of VMware.

Version 4 running on top of Windows 2k and version 6 running on Linux.

Both seem to be running fine despite having to be fooled into running on VMware. My 7970 registers with both but since I'm not familiar with call manager I have not yet figured out how to set things up with a sip provider.

If there are any config files that the comunity would find usefull to learn from or whatnot let me know, could posibly set up some sort of remote desktop / ssh as well if needed.

 

[TheShniz]

I'm interested to know where you got the software from, and if Cisco makes it available (I've not seen it)... reason being, I'd like to try integrating asterisk w/ it. How did you modify it for VMware, by editing anaconda to not validate for certified hardware? Anywho, you've perked my interest

 

[gman]

Version 6 was the easier one to install, just be sure to give it an ample amount of disk space. All it's going to do is give you several stern warnings about how a vmware installation is not for production and so on..

I found some information on how to set this up here:

V4:
http://www.blindhog.net/cisco-install-call-manager-4x-with-vmware/

V6:
http://www.blindhog.net/how-to-install-call-manager-6x-in-vmware/

I also wanted to try experimenting with asterisk and Call manager integration but really haven't found a good place to start. Hopefully this will help get things going

 

[someguysname]

We use Call Manager 4.2 were I work and believe it or not integrating it with asterisk was easier than falling off of a log. I followed this guide for the most part:

http://www.voip-info.org/wiki-Asterisk+Cisco+CallManager+Integration

I ignored the asterisk extensions.conf part and used a custom trunk instead with:

SIP/$OUTNUM$@callmgr1

for the custom dial string

In call manager I created a route pattern (Route Plan>Route/Hunt>Route Pattern) of 8.xxxx (I use 4 digit extensions) and sent it to the sip trunk created in call manager. Users just dial 8 and then the extension on the asterisk box they want to reach.

It was that easy.

 

[gman]

Thanks for getting me pointed in the right direction someguy, I now have my 7970 making and receiving calls through call manager directly with our sip provider.

When I try the custom dial string, I can see that I am now hitting the pbxiaf server but every time I dial an extension I get the "number not in service" error.

Does this setup let you make direct outbound calls as well? Or would I need a separate route plan set up in CM that would be something like:

9.XXXXXXXXXX
Or
9.[2-9]XX[2-9]XXXXXX


And, since it may be of use to someone else, I have been using this page to help with route plan setup:

http://www.cisco.com/en/US/products...ducts_tech_note09186a0080094b2a.shtml#topic1a

 

[someguysname]

On the general settings page of PBXIAF try setting allow anonymous inbound sip calls to yes...

I can make outbound calls through call manager from pbxiaf. I have an outbound route in pbxiaf that simply matches 9 then takes it off and sends the rest on to the custom trunk. (9|.) The user dials 9 to get "out" of asterisk and ccm...someone else installed the CCM, i just inherited it a few years back, but I just used whatever was setup on it already, no additional routes out were needed...

I also have a Sipura 3000 that plugs into a POTS line (our fax is still on POTS, we use them for emergency lines if the network is down) that basically ignores anything being dialed out of the line 1 interface on it or anything coming in the pstn...but I can dial out of it from asterisk as well....so they use an 8 to dial out of it if for some reason the call manager is down.

 

[wardmundy]

On the general settings page of PBXIAF try setting allow anonymous inbound sip calls to yes...

... and, if that works, then set it back to No and read this article for a way to allow specific anonymous calls into your system without opening Pandora's Box.

 

[someguysname]

You wont have to leave it set at yes nor will you have to make any weird changes to extensions.conf to get it to work...

It seems some of the pages at voip-info have been deleted so hang tight and I will try to find my notes on that problem or look through the way back machine...I know I had the same issue when I set mine up but that was almost a year ago.

 

[someguysname]

Well, it's been a few days and I still cant find what I had to change. I do remember testing the anonymous inbound sip calls, but I dont remember what I had to do to fix it. I can tell you that mine are not set for anonymous inbound calls and the ccm can send calls to the asterisk and asterisk can send to ccm. Good luck...