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.
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.
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.
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.
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.
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:
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.
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).