similar to: Bunch of set-up/usage questions (SLA, MWI, SMS proxy's, crypto, Fax, etc)

So, for the hotel project, what was the conclusion? I don't remember seeing a summary. And were any of the phones combination Wifi/GSM? Like a Nokia E70 or E61i? I've been looking for such a phone to use myself, but so far haven't found one that I liked. Common flaws were: * poor standby time * lousy interface for doing complex configurations (esp. certificate management) *

I'm looking to ditch my GSM phone with Cingular and provide my own calling services with Asterisk hanging my phone off the Edge or 3GPP network... Luckily, I have the option (through my employer) of getting a data-only plan. My question is, other than the Nokia E61i or E70, what phones will do SIP-over-EDGE or SIP-over-3GPP? The iPhone doesn't yet have a SIP client, and it's

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:

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

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:

Hi Friends, Thank you for your response. Yesterday only, I configured my Nokia E70 mobile and its working fine. For group members convenience, here I am giving the configuration: Configuring the Nokia E70: Go to Menu - Tools - Settings - Connection - Sip Settings - Profile name: Olivetalk Service Profile: IETF Default Access Point: Olive Public user name: sip:102@202.xxx.xxx.xxx Use

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

This issue is peripherally related to the following thread Re: [Dovecot] client certs with godaddy ssl cert This is running on CentOS 5.2 with latest Atrpms for Dovecot as of this weekend. # rpm -qa | grep dovecot dovecot-sieve-1.1.5-8.el5 dovecot-1.1.4-0_81.el5 With assistance from Rainer Frey (Inxmail GmbH), I am able to successfully use client ssl certs for imap access on both my Nokia

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

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

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

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>

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.

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

Hi all, i'm trying from several days to setup a SLA on my machine with some THOMSON 2030. My goal is to bind every F key to an extension (NOT a trunk). So, F1 = 201, F2 = 202, F3 = 203, and so on... I'm googled thousand of pages and many more confusing concepts are in my mind. My server uses extensions with numbering 2XX placed in context 'phones'. I set yet in sip.conf: