Shared Line Appearances

Chris,

Can you please educate me on the shared lines in Asterisk. If I understand what I have read, Shared Lines allow the phone buttons to act like a key system.

If a call comes to shared line 1, all phones with shared line 1 will ring. When Ext 1 answers, the light for shared line 1 goes green on their phone and red on the other phones that did not answer. If Ext 1 puts shared line 1 on hold, all the shared line 1 light will blink allowing anyone with an appearance of shared line 1 to push the button and retrieve the call.

Is that correct? This might be a question best suited for the Asterisk-Users group.

http://www.voip-info.org/wiki/view/Asterisk+SLA

Have you tried to get it to work in a vanilla Asterisk install?

-Matt

I looked at sla.conf and sla.pdf, but see no support for this in PBX Mgr multi-tenant. For all involved here, I'd like to keep our configurations within the PBX Mgr interface. It appears to try and use sla.conf, I'd need to do all the configuration manually, which defeats the purpose of PBX Mgr.

It appears from what I'm finding that with Asterisk 1.4, I can potentially get the functionality by using, in sip.conf:

[general]

....

linitonpeer=yes

notifyonringing=yes

notifyonhold=yes

allowsubscribe=no

and then adding in PBX manager for the SIP extensions other options 'allowsubscribe=yes' - e.g.:

[phone1]

type=friend

call-limit=10

allowsubscribe=yes

....

and then, provisioning the phones with buttons for the 'shared' extensions of type BLF.

I imagine I"d also need to setup the phones that are shared to the secretary's multiline phone as multiple-devices in PBX manager, with them each ringing her phone - which would light and use her 'main' line/button on the phone rather than the 'shared' appearance - the BLF buttons. The BLF buttons, I gather, would just show the status of the extensions - hold, in use, ringing, etc.

Make sense? Just wondering if anyone else has tried this and any caveats they may have found.

One caveat I can think of off-hand is as calls come in, these 'line keys' get used up preventing phones from making calls.

Another caveat based on this previous caveat is up-front cost. In order to function with any usability you have to use an IP550 or larger capacity phones. A 2-line phone just isn't going to be useful if both lines are in use by other handsets and it keeps you from calling out or calling other extensions or handsets. That takes you from an IP330 at everyone's desk to something much costlier like a 650 and expansion module (depending on how many lines you want into the tenant).

I have found ring-groups and call-parking to be a better solution and give much greater flexibility. Incoming calls can still ring a litany of handsets, but a single call does not impose limitations on everyone else.

Getting a full-featured PBX to mimic a 'key' system is akin to forcing a low mileage BMW M5 to perform like a vintage bare-bones honda from the early 80's with no power steering, no power brakes, manual windows/doors/mirrors etc. Sure you could probably do it, but those not familiar with a BMW M5 might go around telling everyone just how much they suck :)

Truthfully, it used to come up; until our sales people got better at selling. Now it never comes up. Phone systems now seem to fall subject to new technologies the same as PCs. They are no longer stuck in the 80s. I've never had anyone call me and ask for a new computer that only has 8MB of ram running on a pentium-pro processor. They shouldn't call asking for a phone system that behaves like the 80s either :). I could just sell them analog ATA's if they wanted that, the cost of equipment would definitely be far less =)

reading up on sla.conf (in sla.tex under docs/) you have to create a 'trunk' for every line appearance. That's either mapped to a specific zap channel or a specific sip registration. From what I gather in the configuration you would have to have a special sip registration for each and every 'line key' appearance. This would be impossible in multi-tenant. Its not very practical to have special registrations to your sip provider for each and every tenant in MTE; its even less practical to have 6 registrations per tenant.

Erik - I agree and additionally, not seeing any support for sla.conf in thirdlane. Everything tells me that that is not the way to go. Instead, I think I'm better off trying to emulate the functionality as much as I can using something similar to what I laid out earlier.

Am hoping to get some feedback on this from Alex as well, if possible.......

Just as Erik said - this may not be practical for multi tenant or even larger single tenant installations.

That said - I did not look at this in a while - perhaps I will look again to see what we should do about SLA.

I have a small office client that needs to have this work with their Linksys SPA942 phones as the BOSS wants to know when his secretary is on the phone so he doesn't yell down the hall while she is talking to a client (especially when she tells the caller he is not in). Just to glance at the phone is a convenient way to tell also, not having to do a blind transfer to send the call where it needs to go is also good for the small office client.

This might be 'old school' but it does have a purpose in the world of 'old school' minded clients.

Greg

You don't need SLA to know when someone is on the phone, you need busy lamps aka buddy watching. you are describing an entirely different need.

wanting to know when someone is on the phone or not is not the same as neutering handsets in such a fashion that all 20 handsets have a limitation of only 4 people talking, ever.

the use of busy-lamps/buddy-watch is part of the sip subscription/notification already built into asterisk/thirdlane. Additionally, graphical software such as iSymphony does this, as well as track advanced status markings like 'in a meeting, out to lunch, on break, available'. Both are achieved without imposing some artificial limitation on the customer.

Hey Erik,

Could you please describe how to enable/configure this? That would be great.

Greg

gregshap.

BLF "Busy Lamp Field" is accomplished by using the HINT CMD. This is already setup in Thirdlane and no additional configuration is required on the Thirdlane side of things. On AASTRA Phones, you setup a BLF button and set the extension which you want to monitor. I have not done this on a linksys phone so I am not sure what you would use for the button. I do belive there used to be an issue with Asterisk that you had to define calllimit=1 to get this to work, but I think that was quite some time ago.

Thanks,

Chris

you have to set a call limit in 1.2 and 1.4 but it can be arbitrarily high such as 10.

if you're using Polycom phones, this should work out of the box, if you have any free line buttons on your phone. Add the other person's extension to the contact list on the phone. By default, the first entry in the contact list will show up as a speed dial on the first empty line button if you had empty line buttons available. The icon on that speed dial button will change to show the phone's current status, based on subscription notifications from asterisk, provided by the 'hint' lines in extensions.include (which Thirdlane PBX happily puts in there for you by default when you create an extension)

According to Alex, BLF won't work for Linksys or Polycom phones.

justdave - you say this works on Polycom out of the box? Since we auto-provision the customer's Polycom phones, I've tried to set them up a shared extension (registration on 2 phones), but the first one to register with Asterisk gets all the calls. I don't need to go this direction if I read you right....it's setup on the phone in the Contact list on the phone alone? How does that communication back with Asterisk then to use the 'hint' lines?

BusyLamp *DOES* work with polycom phones, you just have to know how. You don't use the BLF option in PBX manager, it is handled on the phone.

eeman - I see how to setup the Buddy List (Contact List) that is supposed to watch the extensions and show their status. What I don't see and cannot find in the manual is how this is shown in the lights on the phone watching and also how it will show as busy when the watched extension is on a call. Seems to always show just Online or Offline depending on whether the phone is registered with the server or not. Running 'sip show subscriptions' does show the watched lines ok, but that's about as far as it seems to go.

Am I not looking at this right? Or is there something I'm missing still?

there is a 'watch buddy' option in the contact information. That will track status. If its not there it is because you dont have

feature.1.name="presence"

feature.1.enabled="1"

set in your polycom configs.

Unused linekeys are automatically populated with your speed-dial contacts based on their speed dial index number. It is the same way you populate the expansion module.

eeman - interesting - thanks. Did this yesterday and the buddy lines were added to the end of the configured extensions on the phone and expansion module. They only showed online/offline, but did work as far as hitting those buttons to transfer calls. However today the phone rebooted, grab it's config from the provisioning server as it has been (which configures the phone for 7 extensions) and all the extensions from the provisioning config file MAC_ADDR-registration.cfg were not on the phone - only a single button which all incoming calls are seen on and then the buddy buttons - which are non-functional.

Are these Polycom phone setups always such a PITA?? Have lots of Linksys, Cisco and Grandstream phones out and have no problems with them, but these Polycom phones are less than cooperative!

Ideas?

I only use Polycom phones and can say that once you learn how to use them and get them set up, you wont ever want to use anything else :). The sound quality is better than cisco and the sheer number of things you can alter and re-specify is mind blowing. The 4-way visual conference in the feature pack is pretty awesome as well. We have one customer who wrote an xml application to do contact lookups in their MS Active directory database and place calls based on the results.

give me the specifics on the phone model you are using and any add-ons, and how you set it up in pbx manager etc. also tell me the versions of software you are running for:

pbxmanager

asterisk

zaptel or dahdi

polycom bootrom

polycom sip firmware

some older pbxmanager versions tried to assign 12 extensions to a 601/650. I can show you where in the polycom configs you would alter this. I am also trying to work with alex on an a gui modification to the provisioning screen so that you can tie a registration to 4 (arbitrary number) lines instead of just a single line or 'all lines'. This would take some manual file alteration off your plate.

Erik - thanks - appreciate your willingness to help on this!

The phone is: PolycomSoundPointIP-SPIP_601-UA/2.1.3.0028 and has the line expansion module attached. We are running the following software versions:

PBX Manager: 6.0.1.63

Asterisk: 1.4.21.1 (64 bit)

Using neither zaptel or dahdi - we have SIP gateway service with Verizon - all calls are to and from their SIP servers (via VPN tunnels).

The provisioning file I have setup for this phone is as follows. Lines 4 through 7 are what I setup for the extensions on other phones in that office (extension numbers without the trailing 1) to ring this phone too.

reg.1.displayName="Main Line"

reg.1.server.1.address="voip01.axint.net"

reg.1.server.1.transport="UDPonly"

reg.1.address="100-Customer"

reg.1.label="100"

reg.1.auth.userId="100-Customer"

reg.1.auth.password="klajdlks"

reg.1.callsPerLineKey="1"

reg.1.lineKeys="1"

reg.2.displayName="Main Line 2"

reg.2.server.1.address="voip01.axint.net"

reg.2.server.1.transport="UDPonly"

reg.2.address="200-Customer"

reg.2.label="200"

reg.2.auth.userId="200-Customer"

reg.2.auth.password="klajdlks"

reg.2.callsPerLineKey="1"

reg.2.lineKeys="1"

reg.3.displayName="Tracy"

reg.3.server.1.address="voip01.axint.net"

reg.3.server.1.transport="UDPonly"

reg.3.address="0655-Customer"

reg.3.label="0655"

reg.3.auth.userId="0655-Customer"

reg.3.auth.password="klajdlks"

reg.3.lineKeys="1"

reg.4.displayName="Tracy"

reg.4.server.1.address="voip01.axint.net"

reg.4.server.1.transport="UDPonly"

reg.4.address="06441-Customer"

reg.4.label="06441"

reg.4.auth.userId="06441-Customer"

reg.4.auth.password="klajdlks"

reg.4.lineKeys="1"

reg.5.displayName="Tracy"

reg.5.server.1.address="voip01.axint.net"

reg.5.server.1.transport="UDPonly"

reg.5.address="06411-Customer"

reg.5.label="06411"

reg.5.auth.userId="06411-Customer"

reg.5.auth.password="klajdlks"

reg.5.lineKeys="1"

reg.6.displayName="Tracy"

reg.6.server.1.address="voip01.axint.net"

reg.6.server.1.transport="UDPonly"

reg.6.address="57311-Customer"

reg.6.label="57311"

reg.6.auth.userId="57311-Customer"

reg.6.auth.password="klajdlks"

reg.6.lineKeys="1"

reg.7.displayName="Tracy"

reg.7.server.1.address="voip01.axint.net"

reg.7.server.1.transport="UDPonly"

reg.7.address="57341-Customer"

reg.7.label="57341"

reg.7.auth.userId="57341-Customer"

reg.7.auth.password="klajdlks"

reg.7.lineKeys="1"

wow, what a mess =)

reg.1.displayName="John Doe"

reg.1.server.1.address="172.16.100.1"

reg.1.server.1.transport="UDPonly"

reg.1.server.2.transport="UDPonly"

reg.1.address="2103"

reg.1.label="2103"

reg.1.auth.userId="2103"

reg.1.auth.password="something"

reg.1.outboundProxy.address=""

reg.1.outboundProxy.port="5060"

reg.1.outboundProxy.transport=""

reg.1.lineKeys="6"

reg.1.callsPerLineKey="8"

in this example the 6 phone line keys are set to extension 2103

then you create a buddy , specify the extension, select yes to 'watch buddy' and give it a speed dial index of 1, it will be the first listed on the expansion module.

for these 'watched buddies' go into their extension configuration in pbx manager and limit their calls to 10 calls. reload asterisk.

now when a watched buddy calls out, your LED on the expansion module will turn red. Likewise when the buddy gets called the same will happen. Additionally you can press the button next to the buddy to call them.

Coming soon is the ability to use less line keys then maximum on polycom (assign a registration to a selected number of buttons) and use the rest for buddy watch (BLF).

Best regards,

Alex

Erik - thanks - will give it a try and see how it works for them. Appreciate it!

Alex - look forward to more direct support in PBX manager for Polycom - getting more and more people wanting to use them.

Erik - thanks much for your help on this issue. Been a couple of days now and it seems to be working just fine. Now to work on getting issues with their Messages buttons not working and volume changes not persisting between calls......

you must be using an old version of pbxmanager, a few versions back included a new default polycom_local.cfg in /etc/provisioning that persists volume settings and overrides the messages button to dial a specific extension (*98).