similar to: What is acceptablenetworklatencyforvoipconnection?

Hello, I'm new to the list but I've been following it since its inception. First of all sorry for the length of this post, if you're not a Fedora user just skip it. I've been building Compiz packages since Fedora Core 5 came out (Xgl only) and updating since then with all the new stuff, AIGLX and whatever. I use them on a very large range of hardware. Attached are my source rpm

Il 02/08/20 19:09, Alessandro Baggi ha scritto: > > Il 02/08/20 18:54, Stephen John Smoogen ha scritto: >> On a side note, you keep emphasizing you aren't expecting an SLA.. but >> all your questions are what someone asks to have in a defined SLA. I >> have done the same thing in the past when things have gone badly, but >> couching it in 'I am not asking'

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

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

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

Muchas gracias Carlos, mas aun pues en españa es mas de medianoche del viernes, no ? eric. On Fri 04 Jul 2014 19:08:24 CLT, Carlos Ortega wrote: > Hola, > > Esta es una manera de hacerlo (he incluido un par de cambios > adicionales en las etiquetas): > > #--------------------------------- > mfctr$dia.new <- paste("dia_",mfctr$dia, sep="") > >

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.

https://bugzilla.mindrot.org/show_bug.cgi?id=1959 Bug #: 1959 Summary: Incorrect Sequence Numbers for NetFlow v9 export. Classification: Unclassified Product: softflowd Version: -current Platform: All OS/Version: All Status: NEW Severity: major Priority: P2 Component: softflowd

Fellow asterisk users, I am trying to get Single Line Appearance functionality working on a set of Linksys SPA942 phones and have not been successful. It looks like sla.conf is not getting read, only one phone reads as registered for the shared line, and a busy tone every time the shared extension is dialed. I have followed the documentation [1] and followed through other threads I saw

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

I configured my sla.conf to use with a Polycom phone. I have no idea if I did it right, however, none of the console "sla" commands exist. Do I have to something special to compile in this support, or should it just work out of the box? ~jay

I understand that the Asterisk SLA implementation is somewhat different from most key systems and PBX systems. I also understand that in Asterisk, one does not put an SLA line on hold since it is just a MeetMe conference. However, is there any way to make the BLF flash when the answering party on the Asterisk system presses the hold key on their set and leaves the calling party alone in the

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

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

Il 02/08/20 18:54, Stephen John Smoogen ha scritto: > On a side note, you keep emphasizing you aren't expecting an SLA.. but > all your questions are what someone asks to have in a defined SLA. I > have done the same thing in the past when things have gone badly, but > couching it in 'I am not asking' just makes the people being asked > grumpy. Better to be open and say

I would be incredibly grateful to anyone who'll help me translate some SAS code into R code. Say for example that I have a dataset named "dat1" that includes five variables: wshed, site, species, bda, and sla. I can calculate with the following SAS code the mean, CV, se, and number of observations of "bda" and "sla" for each combination of

Has anyone successfully gotten the features of sla.conf to work with Polycom phones? I would like to get 501's to show 3 shared lines on each of the keys. Particularly I am looking for a sample configuration for the phone, the Asterisk configuration seems simple enough. I've searched and searched and searched and have not found anything, general info about SLA yes, but nothing specific

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:

I'm trying to record all calls including calls that are part of a SLA. Using both monitor and mixmonitor the recording appears to happen (that is, asterisk logging shows it happening) however the file is never written to. It doesn't seem to be possible to use the recording that is part of the meetme app because when the SLA app creates the conference it passes NULL in the options field

Anyone have an example of using SLA with SIP handsets in Asterisk 1.6? I'm looking at the sla.tex file and wondering when it was last updated... Thanks, -Philip