similar to: OT: VoIP SLA for SIP trunking - SMEs

Dear all, I've created a digium certified asterisk professional - dCAP linkedin group for anyone, dCAP, interested: http://www.linkedin.com/e/gis/60298/39AE1350DBF3 Best regards, Marco Mouta dCAP November 2006 -- Esta mensagem (incluindo quaisquer anexos) pode conter informa??o confidencial para uso exclusivo do destinat?rio. Se n?o for o destinat?rio pretendido, n?o dever? usar,

Hi guys, I've setup on box with a TE110P and time to time I need to access remote equipment outside of our office and use a data channel. I'm wondering if do I need to buy a POTS line only for this time to time acess or what's the easiest way to do that via my TE110P on asterisk box. I know that is possible data transmission with this Digium Card, I'm wondering how... Any tip any

Dear all, I've a live system that needs to be upgraded but, before I proceed to the upgrade I want to assure the rollback process. That's why I'm requesting your feedback, in fact this asterisk in live system isn't going so bad but.... the upgrade is essential NOTICE that the upgrade will keep the same version 1.2 not from 1.2 to 1.4 Requirements: -backup /usr/sbin/asterisk

Hi guys, First of all, I know that this server must be upgraded asap, I'm just wondering if anyone of you has already faced this problem and , if so, would the upgrade solve my problems... CAPI version 0.6 Asterisk 1.2.5 AGI scripts are being used Main problems: -Dropped Calls - ps aux | grep asterisk shows that asterisk (that is started with safe_asterisk) is generating multiple

Recent events, including vulnerabilities that were reported and the subsequent discussions about how they were handled, have made those of us that manage Asterisk development decide that it is time for the Asterisk project to have a formal security vulnerability and advisory reporting process. Over the next few weeks we will begin to formalize and document this process on the asterisk.org

Recent events, including vulnerabilities that were reported and the subsequent discussions about how they were handled, have made those of us that manage Asterisk development decide that it is time for the Asterisk project to have a formal security vulnerability and advisory reporting process. Over the next few weeks we will begin to formalize and document this process on the asterisk.org

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

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

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

Sds, Paulo Henrique Cardoso Administrador de Redes - T.I. NHS Sistemas Eletr?nicos Ltda Av. Juscelino Kubitschek de Oliveira, 5270 Cidade Industrial, Curitiba - PR Fone/Fax: (41) 2141-9246/9247 www.nhs.com.br IMPORTANTE: Esta mensagem, incluindo quaisquer anexos, ? endere?ada exclusivamente ao seu destinat?rio e poder? conter informa??es confidenciais. A revis?o, distribui??o,

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

Hi. I've been researching very deep into SLA in Asterisk 1.4, and am unclear as to if the feature exist. I know the commands and configs are available, but there is no documentation, and I've heard that it will not be supported until 1.4.1. Does anyone have a definitive answer on this? If it is available, is there any documentation on the feature? -Chris

I have read that 1.4 has shared line appearances, which I assume will work with Polycom phones. Has anyone configured this and verified it working? I was going to start playing around with it but wanted to see if anyone else has tackled it yet. Bill -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

Hello, I am looking for a "real life" example of using SLA lines under Asterisk. I'll describe my environment and would like to know how I define it in Asterisk (version 1.4 final). Suppose I have two multi lines phones. The first phone has extension 1 assigned to it, and the second phone has extension 2 assigned to it. Now, I want extension 3 to be available on both phones as

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

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>