similar to: Shared Line Appearances in 1.4

Greetings, Back in September someone asked about documentation for the new SLA feature in 1.4, however they received no replies. I thought I might ask the same question now in December. Apart from sla.conf.sample and a few comments in app_meetme.c I have been unable to find useful documentation. Is anyone using this feature right now? Is there a helpful source for information this highly

If SLA supports IP trunks, can shared extension appearance be achieved using a local SIP trunk in place of an extension? Basically, I'm trying to allow some stations (Polycom IP 650) to have a shared extension amongst all of them. Ideally, I'd like for the LED to show if that extension is in use, and I'd like for the extension to ring all stations on that extension when a call comes

anyone using/experimenting with this new feature in asterisk 1.4? is anybody able to post some info how to use and what features are supported? I have general knowledge how SLA should work, ie. monitor status of another line like BLF with additional features like answer ringing call, barge into existing call on shared line and make conference call or resume call, that was put on hold on

I've noticed sla.conf in Asterisk 1.4. I'd love to test it, but how does it work? There's bupkiss docs, and until I have a clue how to use it, I can't test it. Doug.

Hi all, I'm using 1.2.9.1, with the "metermaid" patches to show parking spot status on Snom BLF lights. I see from http://www.asterisk.org/node/97 that the metermaid code has changed substantially since 1.2.9.1. Is anyone successfully using the "new" metermaid functionality in 1.4.x? I'd like to hear any good/bad experiences before I attempt even a test

I'd like to be able to use my Snom 360 LEDs to view the status of parking slots, so I'm trying to install the "metermaid" patch (http://bugs.digium.com/view.php?id=5779). Can someone help an svn newbie figure out how to install this patch? I've done the following: svn checkout http://svn.digium.com/svn/asterisk/team/oej/metermaids cd metermaids make clean make but I get

Hi all, Is there anyway you get Asterisk to let you park a call a second time via pressing the # key + 700 (or whatever is in features.conf) ? If you get a call out of the parking lot & hit # again it just sends the DTMF tone down the line and nothing happens.. A few posts have hinted that you cannot blind transfer a parked call, but surely there must be some way around this as I can

Size: - one location - 15 IP Phones ( 1 dect) - Create new voip trunk (current are ISDN) (30 number block) - LTS is important - an SLA is optional at the moment there is no one On 11.12.2017 22:31, Ron Wheeler wrote: > You might want to add some details > - size of the project > ?-- number of locations > ?-- number of extensions > - are you converting your trunks? > - what are

Hello, I'm trying to setup shared lines with Asterisk 1.4.26 and Snom phones. It seems that Asterisk works correctly (I get "State: SLA_TRUNK_STATE_RINGING" from the CLI) but the lamps on the phone are not blinking even if I setup one function key on my phones as shared line with number: <sip:line1 at 192.168.0.123> where line1 is my DAHDI/32 Any hints appreciated. :)

Does anyone have Shared (bridged) Line Appearance working in Asterisk 1.4? Specifically with the Aastra 55i. Specifically, I am using the Aastra 55i with the expansion module. We want to see if other handsets are being used. (BLF) Getting BLF to work would be a great start. It sounds like setting up the hints properly will achieve this. right? Not totally sure how this should be configured.

I am learning about SIP phones little by little. I've been working with * for about 2 weeks now with FXS and FXO ports and analog phones, but we want to evaluate the utility of going to SIP phones directly from here rather than investing in analog phones first. One of the questions I have is multiple line appearance on phones. Since the phone basically acts like a computer, there is no

Hi - I'm wondering if any has experience with the Uniden UIP-200 phones. The product info says that the 8 led buttons at the top are all programmable. Can they be programmed as separate line appearances (ala Snom 200, Cisco 7960, Zultys Zip4x4, etc)? In other words - is the phone capable of multiple SIP registrations? Also, the post about these phones at voip-info.org mentions some

Has anyone had any success with getting SLA going between 2 SIP phones? (Particularly a set of Cisco 79xx's) The SLA document that comes with the asterisk source is about as clear as mud. Does anyone have a working sip.conf, sla.conf, and extensions.conf that I can use for reference? The part I'm most confused about is how to build the lines in sip.conf and how the phones should

Has anyone done SLA with Polycom phones? I've got a large project coming up where the customer is keen on SLA for trunks and extensions. Trunks will be on a PRI. We may do this with Cisco phones if they work better. Mark Willis

Hello, every body Anyone set up, the sla sharing line appearances, in asterisk, I'm setting, tha but, don't, work, I change the sla.conf, extensions.conf, and sip.cfg, but don't work fine. Any one, could setup, tha? Regards Carlos Rojas -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, Before I dig into the SIP traces maybe someone has a clue: We have a few Snom phones, and the call-limit on the extension associated with them is set to 1 (so the other Snom phones can blink the relevant LED). When I call a SNOM phone (300 with the latest firmware) and the call is answered the phone is in "in use" state (verbose printing from Asterisk). When I hit the

Hi All. I've been experimenting with SLA on Asterisk 1.4.13 (patched up to 1.4.14). I am using a SIP channel for my "trunk" line. On the whole things are good, but I have noticed that if I misdial an outgoing call, i.e. I get 404 "Not Found" in the SIP trace, then the trunk line just drops, rather than presenting an error tone or message to the user.

Hello, do you know if it is possible to define the SLA configuration in the database for realtime usage with asterisk? Leandro -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20141114/7c4f09a4/attachment.html>

I have a client with 20 GXP-2000s. Everything seems to be working fine. However, after a couple of weeks of use, the client is having a hard time adjusting to the new IP based phone systems and only misses one feature from their old Lucent system. That is, they had 8 analog lines before and all their old Lucent phones showed a button for each line. So, it was easy for anyone to say,

In the real world (or at least in my world) we use undersubscribed internet connections that come with a service level agreement (SLA) that guarantees that the jitter, delay, and packet loss with be within defined parameters in the service agreement. With most DSL and Cable you will not get a SLA, with the cheapest T1s you might get one, but the only penalty to the ISP if they do not meet is a