Re: [Freeswitch-users] Compact, fanless appliance?

Mitch Capper wrote: > You may want to look at the Intel Atom combo machines you can get a 1.6 > ghz machine probably for around $100-150 USD in a very small form factor > and very powerful. Thanks for the tip. Any link where I could check this out? The cheapest PC's I find are > 230€ (Asus' Ee

Re: [Freeswitch-users] Compact, fanless appliance?

> The cheapest PC's I find are > 230€ (Asus' EeePC), and with not enough room > to stick a PCI card. > Well actually a barebone with VIA C7 + 1 slot for PCI card (carefull you have to remove some metal part otherwise Sangoma/Digium card will not fit *LOL*) and external PSU are around 190/215€

Re: [Freeswitch-users] Compact, fanless appliance?

Thanks Antonio for the links on Acrosser and PCEngines. It seems like PCE's alix1d is a good solution, provided 256MB is enough to hold Linux + Freeswitch + some tiny LAMP stack. Still, it looks like an Atom-included mobo like those from Asus or Gigabyte would be cheaper. The biggest issue is find

Re: [Freeswitch-users] Compact, fanless appliance?

Fred-145 ha scritto: > Thanks Antonio for the links on Acrosser and PCEngines. It seems like PCE's > alix1d is a good solution, provided 256MB is enough to hold Linux + > Freeswitch + some tiny LAMP stack. Still, it looks like an Atom-included > mobo like those from Asus or Gigabyte would be cheape

Re: [Freeswitch-users] any way ring fifo members one by one?

On May 4, 2009, at 9:25 PM, Anthony Minessale wrote: > When you are using on-hook agents, it's presumed to be under low > call volume, you can just set the agents to get popped > into the queue in nowait mode so if the caller changed his mind the > agent will get a hangup. Remember, if there

Re: [Freeswitch-users] any way ring fifo members one by one?

Hello, Anthony, I would like to provide a patch allowing having different call distribution strategies, at least for "call back" agents. Do you think the simple approach of modifying the SQL query in find_consumers (given strategy that would be set from dialplan) would be enough? Thanks, Françoi

Re: [Freeswitch-users] any way ring fifo members one by one?

On Sun, May 3, 2009 at 11:01 PM, seven wrote: > Actually, for the "call back" agents, because the fifo use originate to > start a new session, the new session won't hang up unless one agent answered > or timeout. Agents will hear nothing and wait(member_wait=wait) on the > queue or hanup(nowait)

[Freeswitch-users] Registration problem with multiple IP phones behind Linux NAT firewall router

I am having a problem with getting multiple Polycom IP phones to register to my Freeswitch server. Here is my setup (IP addresses are not actual ones, but are consistent throughout): Freeswitch server in colo facility IP addr: 1.1.1.1 (publicly routable) Linux NAT firewall rout

Re: [Freeswitch-users] Ways of Integrating Sphinx...

BTW Brian, Here is something that would make FS's VAD much better. The technique also improved Sphinx-3 performance in low-SNR enviroments and made it run over 40% faster. http://figment.cse.usf.edu/~sfefilat/data/papers/WeBT5.3.pdf Mark. -Original Message- From: Brian West To:

Re: [Freeswitch-users] Ways of Integrating Sphinx...

Wasn't aware Sphinx 3 was integrated into FreeSWITCH ... /b On May 4, 2009, at 12:00 PM, mszla...@aol.com wrote: BTW Brian, Here is something that would make FS's VAD much better. The technique also improved Sphinx-3 performance in low-SNR enviroments and made it run over 40% faster. ht

Re: [Freeswitch-users] Ways of Integrating Sphinx...

Nope it isn't but does that make a difference if pocketsphinx could use a similar upgrade? Anyway, you now have a way to make VAD better in FS. -Original Message- From: Brian West To: freeswitch-users@lists.freeswitch.org Sent: Mon, 4 May 2009 10:29 am Subject: Re: [Freeswitch-

Re: [Freeswitch-users] Ways of Integrating Sphinx...

Hi, maybe things in speex should be worth to look for too. Just my 2 cents... Regards, Tamas ps: It seems, VAD+DTX in mod_speex does not work. Brian West írta: > VAD isn't really high on my list right now. > > /b > > On May 4, 2009, at 12:46 PM, mszla...@aol.com >

Re: [Freeswitch-users] Ways of Integrating Sphinx...

VAD isn't really high on my list right now. /b On May 4, 2009, at 12:46 PM, mszla...@aol.com wrote: Nope it isn't but does that make a difference if pocketsphinx could use a similar upgrade? Anyway, you now have a way to make VAD better in FS. Brian West br...@freeswitch.org -- Meet us at

Re: [Freeswitch-users] Ways of Integrating Sphinx...

No problem but at least this reference could be used in the future. -Original Message- From: Brian West To: freeswitch-users@lists.freeswitch.org Sent: Mon, 4 May 2009 10:56 am Subject: Re: [Freeswitch-users] Ways of Integrating Sphinx... VAD isn't really high on my lis

[Freeswitch-users] XML editors and freeswitch XML spec compliance?

Has anyone found a decent XML editor for the XML files? I loaded the files up in OrangeVolt under Eclipse, but didn't have much time to play around with it. Also, I tried opening them with an XML plugin for TextMate, and it says the files are not compliant. The reason is that the first lin

Re: [Freeswitch-users] XML editors and freeswitch XML spec compliance?

On May 4, 2009, at 1:52 PM, Jay Austad wrote: Has anyone found a decent XML editor for the XML files? I loaded the files up in OrangeVolt under Eclipse, but didn't have much time to play around with it. Also, I tried opening them with an XML plugin for TextMate, and it says the files are not

Re: [Freeswitch-users] XML editors and freeswitch XML spec compliance?

I recommend something that does syntax highlighting without all the pedantic nonsense. In a Windows environment I've used notepad++ and notepad2. In a Linux environment I've just used Kedit in the GUI environment but emacs and vim work nicely in the text-only environment. In OSX I use emacs in a te

[Freeswitch-users] ANNOUNCEMENT: FreeSWITCH mod_opal Now Officially In Beta

The FreeSWITCH team would like everyone to know that the mod_opal module is now officially in beta. Please read this article for more information: http://www.freeswitch.org/node/179 Many thanks to Robert Jongbloed and Craig Southeren of the OPAL project for their many years of support for open sou

[Freeswitch-users] Invite on SIP instead of TLS

I have updated my system from SVN 10003 to 13223. I also have updated all libraries etc. Everything works fine (SIP +TLS) when calling internal numbers (conference). However calling internally registered phones does not work. Here are some facts which do not fit together - phones are sucessfully

Re: [Freeswitch-users] Invite on SIP instead of TLS

I'm pretty sure this was fixed in 13226 please update. You're using a new feature it seems. /b On May 4, 2009, at 6:09 PM, Peter P GMX wrote: > fs_path=sip%3A723329%40217.xxx.xxx.186%3A2651 Brian West br...@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com

Re: [Freeswitch-users] any way ring fifo members one by one?

On May 4, 2009, at 9:25 PM, Anthony Minessale wrote: On Sun, May 3, 2009 at 11:01 PM, seven wrote: Actually, for the "call back" agents, because the fifo use originate to start a new session, the new session won't hang up unless one agent answered or timeout. Agents will hear nothing and

Re: [Freeswitch-users] any way ring fifo members one by one?

I think order by outbound_call_count will cause problem. Think about the fifo run a few days, and we added a new member, the outbound_call_count will always less than others in a certain time. What about use order by next_avail? On May 4, 2009, at 10:55 PM, François Delawarde wrote: Hell

[Freeswitch-users] help with mod_conference stability

We had our first big issues with our freeswitch system today. During at least 2 conferences, audio became jittery and there were three occasions where everyone was dropped from a conference. Even so, conference recording was not interrupted, and the freeswitch debug log doesn't show anything unusua

Re: [Freeswitch-users] help with mod_conference stability

Stephen Crosby wrote: > We had our first big issues with our freeswitch system today. During at > least 2 conferences, audio became jittery and there were three occasions > where everyone was dropped from a conference. Even so, conference recording > was not interrupted, and the freeswitch debug l

Re: [Freeswitch-users] help with mod_conference stability

To answer the other question, I did not try pinging the host while the problems were occurring, but I was able to call back in with no problem only a few seconds after I got dropped (less than 20), so if it is a network issue it would have to have been very brief. --Stephen On Mon, May 4, 2009 at

Re: [Freeswitch-users] help with mod_conference stability

Network problem is what I'm still thinking. Take a look at this log snippet: http://pastebin.freeswitch.org/8813 I'm CALLER_A, and you can see me calling back in on line 8 after I got dropped. But that was only seconds prior. There really seems to be nothing in the log at the time the calls were

Re: [Freeswitch-users] help with mod_conference stability

Sorry for the extra messages, but I've just discovered something that probably helps pinpoint the problem: http://pastebin.freeswitch.org/8814 It seems that the callers were disconnected, but freeswitch had to wait a timeout period before it actually hangs up which looks like about 5 minutes. So I

Re: [Freeswitch-users] help with mod_conference stability

Stephen Crosby wrote: > Network problem is what I'm still thinking. Take a look at this log snippet: > > http://pastebin.freeswitch.org/8813 > > I'm CALLER_A, and you can see me calling back in on line 8 after I got > dropped. But that was only seconds prior. There really seems to be nothing > i

Re: [Freeswitch-users] help with mod_conference stability

Stephen Crosby wrote: > Sorry for the extra messages, but I've just discovered something that > probably helps pinpoint the problem: > http://pastebin.freeswitch.org/8814 > > It seems that the callers were disconnected, but freeswitch had to wait a > timeout period before it actually hangs up whi

Re: [Freeswitch-users] help with mod_conference stability

I was on this conference and there were 4 of us that got dropped at the same time. By the way, these were all regular PSTN lines calling through our SIP provider and being routed to our freeswitch instance which is being hosted on a VPS. When you say a device on my end terminated the call, do you

Re: [Freeswitch-users] help with mod_conference stability

Stephen Crosby wrote: > I was on this conference and there were 4 of us that got dropped at the same > time. By the way, these were all regular PSTN lines calling through our SIP > provider and being routed to our freeswitch instance which is being hosted > on a VPS. > > When you say a device on