Re: [asterisk-users] small homebrew pbx

On Mon, Jun 15, 2015 at 04:56:31PM +1000, Tim Groeneveld wrote: > > On Mon, 15 Jun 2015 16:46:13 +1000 [Lucio] wrote > >Hello all, > > > >I'm new here and I'm interested in building a small PBX with asterisk at > >home. I have one single PSTN line and ethernet cabling in place. I >

Re: [asterisk-users] small homebrew pbx

On Mon, 15 Jun 2015 17:26:40 +1000 Tzafrir Cohen wrote >> >> The SPA3102 can be found cheap on Ebay, and will be easy to setup in >> Asterisk. >> http://www.infoworld.com/article/2633694/data-modeling/your-pstn-and-you--linksys-spa-3102-and-asterisk.html >> >> >> Once the FXS is

Re: [asterisk-users] Calling multiple phones at ones

On Monday 15 Jun 2015, Ivan Demkovitch wrote: > Hello group! > > I’m new to Asterisk but got one running finally :) > > Now I’m trying to solve following problem. I have company Automated > Attendant and each employee have SIP phone at home, SIP phone in office, > cell phone. > > I want all thos

Re: [asterisk-users] small homebrew pbx

On Monday 15 Jun 2015, lu...@sulweb.org wrote: > Hello all, > > I'm new here and I'm interested in building a small PBX with asterisk at > home. I have one single PSTN line and ethernet cabling in place. I > already have fairly decent PC that I can use (AMD FX 8350 16GB of RAM > and RAID 10 SATA d

[asterisk-users] queue periodic-announce without stopping ringing

hello, is it possible to play queue periodic-announce without stopping agents ringing? actual situation is sequential - ring agents, play announce (for 15 sec), ring agents , ... (i need to connect agent with caller asap when agent is free) is it possible with ARI? -- --

Re: [asterisk-users] small homebrew pbx

On Mon, 15 Jun 2015, lu...@sulweb.org wrote: I'm new here and I'm interested in building a small PBX with asterisk at home. I have one single PSTN line and ethernet cabling in place. I already have fairly decent PC that I can use (AMD FX 8350 16GB of RAM and RAID 10 SATA disks). I make and rec

Re: [asterisk-users] small homebrew pbx

On Mon, 15 Jun 2015, Steve Edwards wrote: Although, if you lose power, you've probably lost your Internet connection as well so you could only make calls between extensions. And you would lose the Italian equivalent of 911. In the US, everybody over the age of 6 has a cell phone stapled to th

Re: [asterisk-users] small homebrew pbx

> I don't know this 'translates' to Italy, but this is what I would advise > somebody in the US to consider, assuming you have a reliable Internet > connection. > > 0) I hope you mean you want to run Asterisk at home instead of 'Asterisk > at Home.' A@H was an ancient distribution from around

Re: [asterisk-users] small homebrew pbx

I picked up a cheap JS200-FX on ebay: http://x100p.com/products/js200fx.php for $30, and it works great for a home install. Very low power draw as well. James Cass jcas...@gmail.com On Mon, Jun 15, 2015 at 10:50 AM, Kevin Larsen < kevin.lar...@pioneerballoon.com> wrote:

Re: [asterisk-users] Calling multiple phones at ones

On Mon, Jun 15, 2015 at 12:43 AM, Nathan Anderson wrote: > What you want is called SIP call forking, and unfortunately, last time I > checked (before Asterisk 12 and the advent of PJSIP), Asterisk's SIP channel > driver does not support it, and I would be shocked if Asterisk 12+ changes > this

Re: [asterisk-users] Calling multiple phones at ones

On Mon, 2015-06-15 at 11:03 -0500, Matthew Jordan wrote: > On Mon, Jun 15, 2015 at 12:43 AM, Nathan Anderson wrote: > > What you want is called SIP call forking, and unfortunately, last time I > > checked (before Asterisk 12 and the advent of PJSIP), Asterisk's SIP > > channel driver does not su

Re: [asterisk-users] small homebrew pbx

James Cass wrote: I picked up a cheap JS200-FX on ebay: http://x100p.com/products/js200fx.php for $30, and it works great for a home install. Very low power draw as well. James Cass jcas...@gmail.com The JS-200 runs a very old ( 1.4 ) versi

[asterisk-users] no samples for gsmtolin

Hi list! If I call a number from the phone of my wife, I get this warning: [Jun 15 20:50:18] WARNING[21921]: translate.c:206 framein: no samples for gsmtolin (more time per seconds). I didn't found any help in Google with this message... Someone wrote about "turning off silence suppression", th

Re: [asterisk-users] no samples for gsmtolin

If I call a number from the phone of my wife, I get this warning: [Jun 15 20:50:18] WARNING[21921]: translate.c:206 framein: no samples for gsmtolin I think this is related to silence suppression. Either ignore it, or find the device that does this and disable silent suppression. jg -- ___

[asterisk-users] DAHDI-Linux and DAHDI-Tools 2.10.2 Now Available

The Asterisk Development Team has announced the releases of: DAHDI-Linux-v2.10.2 DAHDI-Tools-v2.10.2 dahdi-linux-complete-2.10.2+2.10.2 This release is available for immediate download at: http://downloads.asterisk.org/pub/telephony/dahdi-linux http://downloads.asterisk.org/pub/telephony/dahdi-too

Re: [asterisk-users] Calling multiple phones at ones

On Monday, June 15, 2015 at 9:21 AM, Nilesh Govindrajan wrote: > How about ringall strategy with a queue? Not sure how that would help. Every SIP phone in the queue would still have to have a unique SIP identifier/username. -- Nathan Anderson First Step Internet, LLC nath...@fsr.com --

Re: [asterisk-users] Calling multiple phones at ones

On Monday, June 15, 2015 at 9:03 AM, Matthew Jordan wrote: > This is true for chan_sip. It is not true for the PJSIP stack. > > The PJSIP stack does allow for multiple devices to register contacts > to a single Address of Record (AoR). You can then dial contacts > individually, or dial all contac

Re: [asterisk-users] Peer unreachable after IP change

On 6/8/15 1:18 AM, Luca Bertoncello wrote: Hi list! Another day, another problem... I'm checking with Nagios my Asterisk at home, and since yesterday I noticed that, after my IP changes (Deutsche Telekom drop the DSL-line every 24 hours, so that I have a new IP every day), the peer of an VoIP-pr

[asterisk-users] FXS Solutions for modems and other non jitter tolerant devices

In the past we have used Adtran Atlas 550's to break out FXS ports for devices like modems. The great thing about the 550 is that internally it is all TDM so there is absolutely zero latency. We are able to use ATA's for faxes and analog phones but devices that use modems, they fail 99.99% of the

Re: [asterisk-users] FXS Solutions for modems and other non jitter tolerant devices

On 15-06-15 08:48 PM, Matt Darnell wrote: In the past we have used Adtran Atlas 550's to break out FXS ports for devices like modems. The great thing about the 550 is that internally it is all TDM so there is absolutely zero latency. We are able to use ATA's for faxes and analog phones but d

Re: [asterisk-users] FXS Solutions for modems and other non jitter tolerant devices

Jon Pounder wrote: Fax is really the only need recently, and even that has alternatives like emailing scans that most people prefer now. The legal and medical communities still seem to prefer faxing, in the ( mistaken? ) belief that it is more secure. In fact the medical community is

Re: [asterisk-users] FXS Solutions for modems and other non jitter tolerant devices

The mediatrix 4102s line kicks ass. On Jun 15, 2015 8:49 PM, "Matt Darnell" wrote: > In the past we have used Adtran Atlas 550's to break out FXS ports for > devices like modems. The great thing about the 550 is that internally it > is all TDM so there is absolutely zero latency. > > We are able

Re: [asterisk-users] queue periodic-announce without stopping ringing

On Mon, Jun 15, 2015 at 9:22 AM, Marek Cervenka wrote: > hello, > > is it possible to play queue periodic-announce without stopping agents > ringing? actual situation is sequential - ring agents, play announce (for 15 > sec), ring agents , ... (i need to connect agent with caller asap when agent >

[asterisk-users] Req help regarding webRTC : Attempted Attempted to set an invalid DTLS-SRTP configuration on RTP instance

Hi List, I am trying to setup a Asterisk setup in AWS instance Centos6.5 . I have installed Asterisk 13.4 with srtp,pjproject. I have configured two numbers for webRTC clients, when i try to call from a client (sipml5) to another client (sipml5) it throws the following error: "chan_sip.c:5