Cisco 9971 phone configuration working example with setup tips

It’s a 9951, now that you say it only uses SIP, I could have spent a night of my life trying to fix something that is not a problem where all I need to do is get a good config on it.

This is a funny illustration because folks on the forum seem to think that we have magic wands and know all the answers. On new stuff I have to just put my head down and work it out like anyone else would.

I will let you know when I get home.

Have had these suckers running well now for about a year, but the amount of time initially I spent trying to get them working in the first place was unreal. Practically no one in Asterisk-land had seen one and all the Cisco folks kind of thought I was mad…It took me literally months!!

Lovely phones when you get them all working, 2-way video too, BLF on call lists, visual voicemail, all the trick stuff you would take for granted on an ‘easier’ phone.

There are some clever people out there still hacking this stuff, folks like Gareth on JIRA who has done a massive amount of coding work on the Asterisk patches to get the BLF, presence, call forwarding, etc, etc functionality working between an essentially proprietary/closed bit of kit (never mind it runs SIP, its Cisco SIP!!!) and the open-source world.

I think I have seen presence lists demonstrated on CUCM. How does this work with the Asterisk platform (not technically, from the user perspective).

Also I have read through the thread and I probably missed it, who wrote the voicemail.php script and is it available for distribution?

If I get the patches done I will wrap all this up in an RPM and distribute it as contributed.

When using the call list feature on the phone, (all calls, missed etc), the presence state of the particular extension in the list is indicated by an icon on the right hand side of the entry to indicate the extension is online, busy, ringing on a call etc. This was introduced I think about 2 firmware revisions ago. So extends the presence functionality quite nicely beyond just the extensions programmed against the speed dial buttons.

The voicemail.php script was originally written by a Alberto Montilla, a Cisco employee as part of their open source initiative, Cisco SPA XML Services for Asterisk environments.

See: https://supportforums.cisco.com/docs/DOC-17245

This was then taken forward, debugged and translated into english by Adam Goodfriend. See
https://supportforums.cisco.com/docs/DOC-19218

I adapted it for my setup, and have been using it happily since.

Although it’s in the public domain, Philippe L. did not want me to re-post it here for fear of potential patent disputes and legal infringement!! Apparently visual voicemail is hot for this with the patent (ambulance chasing) lawyers.

Anyway I can PM you my edited copy of the script, should you wish.

I have been trying to use the voicemail.php from cisco’s site and was getting no where with it…could see the messages if i used a browser but got the XML Error [4]: Parse Error when trying from the phone. Any help with that would be great.

Reading thru the cisco post from your link…your everywhere man. I’m getting the same errors you were, error 500 and parse error. Reading thru that post now to see if I can get it going. If there is something else I need, a PM would be excellent.

Got the voicemail.php partially working…Thanks for the link to the helpful info. I can now choose Visual Voicemail and see the voicemail info, but when i choose to listen to the message, its not played. I can delete the messages, but the script seems to hang when i try to use the “back” button and can never fully exit out correctly. When I do get back to the main screen, the back button is displayed kinda greyed out, but none of my other menu options come back “New Call, Forward All, etc.” unless i pickup the handset and hangup.

You were so right, can’t believe that I spent a night trying to change software that was already loaded.

Anyway, so close. Are any of these errors show stoppers?

6:08:33p XMLDefault.cnf.xml (TFTP)
6:09:03p Updating Trust List
6:09:04p No Trust List installed
6:09:04p SEPD824BDBA6E66.cnf.xml (TFTP)
11:09:06p VPN Error: VPN is not Configured.
11:09:10p File Not Found : English_United_Kingdom/lk-sip.jar
11:09:10p Error Updating Locale
11:09:10p File Not Found : /g4-tones.xml
11:09:10p Error Updating Locale
11:09:10p TFTP Error : dialplan.xml

Again more dumb user stuff, got past that error. Trying to correlate this error to the XML tag:

680 ERR 00:00:47.411443 CVM-SIP : sip_transport_setup_cc_conn : ccm id <5> out of bounds.

For anyone lurking, you need to go to web interface of phone and download the messages file. Open it with a real programmers editor like Notepad ++

Can I please have guidance on how to get past the No Trust List installed message in the log as I can’t get my 9971 to register and seeing as Trixbox is seeing no registration requests and also in the log is “Error Verifying Config Info” I figure it’s the phone that’s not happy (it keeps pulling the config file every 30 seconds too). Any help would really be appreciated!!

You can disregard the No trust list as I get that on mine and it still registers. You want to look at the Error Verifying Config. There is something in your sepXXX.cnf.xml file that the phone doesnt like. Try to keep the file simple. I have had the issue with all Cisco phones i have setup at first. Can be as simple as the phone label name is too long. I have a few configs that work with Trixbox and FreePBX (Asterisk 1.8.12 on both) if you need something to compare to. Im still trying to get the 2way video to work, receiving but not sending.

I’ve tried two configs, the one provided on this thread and another more substantially longer one from another page. I got the 7970’s working with SIP 7 years ago (in the end) and I figured this would be the same and for most part it looks similar! I’d be interested in taking a look at your working Trixbox configs though, thank you.

PM me and I will send that to you along with the DefaultFP.xml as well. I havent figured out the trick to get XML to post correctly to this site yet.

@troy

Have just PM’d you my modified voicemail php file. I am not allowed to publish here.

Regarding your video issues, have you added the necessary configuration in sip_custom_post.conf to enable video for the Cisco handsets? i.e you will need code as follows for the 1.8.12.1 patch (note this will change for later revision of the patch as Gareth has hardcoded the FMTP into the patch itself I believe, see JIRA for more info). Here is an extract of what you will need to see for extension ‘100’ in the sip_custom_post.conf file:

[100](+) cisco_usecallmanager=yes dndbusy=yes video_fmtp=profile-level-id=42801E\;packetization-mode=0\;level-asymmetry-allowed=1 video_btias=1000000 video_imageattr=recv [x=640,y=480,q=0.50]

Repeat for all other Cisco extension that you want to enable video for.

Within FPBX I have maximum video bit rate set to 1000 kb/s and only enabled one video codec, namely H264

Have you checked your TCP / UDP transport settings for the extension? Mine as follows:

from SEP config file:

<transportLayerProtocol>2</transportLayerProtocol>

Transport setting for the extension within FPBX is “UDP only”.

Forget any messages about CTL or trust lists, as troy has noted the phone will register fine without a valid CTL. Your issue is that your SEP config file is imperfect - the phone will parse the config file and if it finds an error, no matter how minor will reject it and revert to the previous know good config. If that does not exist it will refuse to register. Simples.

Sure did, the amazing thing is there isn’t even a value of 5 in the config file. It was set to 4 (as per the example above) switched to 2 still getting the same error.

I sent you a PM

Ataman - Please do not post the same question multiple times, especially in threads that are not completely related. It does not increase your chances of getting answered, in fact it has the opposite effect because as soon as someone scanning the forums sees a duplicate post they are going to characterize the user as annoying.

One post per subject please.

To get you started I deleted one of you three posts (I am one of the volunteer moderators).

Thanks for your attention to this matter.

Thanks

Hello every body who configured Cisco 6900, 9900 series phones, do your three way conferencing is working? Please i need your help