Asterisk and CISCO 7940 on screen busy message

We are a small company and we have set up the basics in Asterisk with our CISCO 7940 phones on SIP.

What i would like to do is set up a softkey or similar too press when im away or unavailable or when im on a call, my ext number or name comes up on the other SIP phones with a busy display.
I cannot find a way to make this come up in the CISCO phones. The CISCO phones are now on SIP. Is this even possible? if so where should i look?

Can anyone suggest where i can look or ask please?

no that is not possible. not with asterisk. if you want to see how many times this has been asked and debated, type “shared line appearance” in the search box here and you will see this is something often asked but never accomplished.

This is presence, rather than SLA, and the problem will be the phone, rather than Asterisk. If you want to do clever things with SIP, you need to use a phone that was designed, from the ground up, for SIP.

I’m not sure how good Asterisk’s support for reporting remote presence is, but you will need to use a recent version to have a chance of it working.

thanks for your feedback, its much appreciated. The Cisco phone has SCCP capability, not sure if Asterisk like to communicate in this format, but would this solve my problem? or look at a different phone?
many thanks again!

SCCP is supported, but the support is not as extensive as for SIP. I don’t think there is any presence support.

There is a 3rd party chan_sccp at sourceforge which I know supports busy lamp fields on the 7971s we have. I have a 7961 which I have yet to try.

Depending on the firmware you have, the Cisco should do a basic BLF for when you are on a call, however, it doesnt work with DND.

You could use a custom hint for a DND managed in Asterisk, and not by the phone. For example:
exten => 202,hint,SIP/202&custom:dnd202
exten => 202,hint,SCCP/202&custom:dnd202 if you are using the SCCP version.

Then use the devstate function to set the custom hint on and off when you call the DND feature. Check out for some info on devstate and custom hints.

However, from memory, the Cisco 7941 only has 2 buttons, so you may need to do something clever with XML on the screen instead.

Let us know how you go

This is possible, but it’s not simple. At the moment it requires patching the Asterisk source code. Patches and issue in general can be reviewed in the Digium bug tracker (Issue ID 13996.

It also requires that your Cisco phones are connecting to asterisk with SIP/TCP (not UDP).

To enable SIP/TCP on any 79xx phone which uses the newer style SEPxxxxxxxxxxxx.cnf.xml files, you need to do the following steps:

[li]Configure the address of your Asterisk server in the section of the config file like this. Replace the with the IP address or FQDN of your asterisk server:

<callManagerGroup> <members> <member priority="0"> <callManager> <ports> <ethernetPhonePort>2000</ethernetPhonePort> <sipPort>5060</sipPort> <securedSipPort>5061</securedSipPort> </ports> <processNodeName></processNodeName> </callManager> </member> </members> </callManagerGroup> [/li]
[li]Change the setting for each line on the phone to the literal string “USECALLMANAGER”. It will look something like this:

<sipLines> <line button="1"> <featureID>9</featureID> <featureLabel>Line One</featureLabel> <proxy>USECALLMANAGER</proxy> <port>5060</port> <name>sipusername</name> . . . [/li][/ul]

You’ll also need to define the Speed Dial/BLF lines in the XML file (they cannot be configured from the phone by hand). Those will look like:

<line button="3">

Lastly, you need to set up Hint()s for the extensions in your dialplan. Those will look something like this:

exten => 102,hint,SIP/steve

I’ve got it running here (in production) and it works fine, but a lot of disconnected steps have to occur to make it all come together. It’s not simple or straightforward.

thank you all so much for your input, hopefully getting close to making it work. again thanks for all your effort in looking into my problem, it is very much appreciated