similar to: IAX2 keep alive?

Well, I don't know how to tune it more, it connects at about that rate in a mediocre rural landline. ILBC uses samples of 30ms, so if you set the trunkfreq set to 20 you will be using more of the necesary scarce bandwidth AND dropping sample info in each frame, thus making audio choppy and unclear. Make shure to disallow all codecs and then allow only ILBC or lpc10 (search for it in

Hello Fach, I have used openvpn for a while and in the new release thereis a feature called "server mode" that makes posible to have a full network of vpn links besides a single TUN/TAP adaptor (a pure software NIC) in the server. I haven't used that feature, but I think this is what you need. Also openvpn runs on linux, *bsd, solaris, windows, and maybe in other OS. Miguel >

Hi, Some time ago, there was a discussion about the inability of nortel meridian pbx to dial analog tones thru an meridian ATA, and the work arround was to enable 808 function that makes the dtmf tones long for the current call. The nortel meridian is connected via a nortel ATA to a TDM400 to a FXO port. Anyone can say me who to actually use that function (you dial something or is pbx

I have an installation that connects in a [very] good day at 22kbps, but the normal is about 18kbps. I use de ILBC codec, and also change in iax.conf the trunkfreq = 20 to trunkfreq = 30 It works, you can understand well the other person, but don't expect miracles or an outstanding sound quality. > Dear Dan; > > Thanks alot for your kindly reply. > > Well, what u advise us

Yes and No The D500 is a terrible thing First problem: it sends some horrible DTMF, so if your voicemail is configured to send #H and *H, it will not work, configure it to send numbers, like 8H and 9H (H is a placeholder for the extension). I also managed to use the MWI (message light), it's a perl script that is in voip-info.org, but with a little correction because the wiki distorted it. If

http://bugs.freedesktop.org/show_bug.cgi?id=14426 Jaime Velasco Juan <jsagarribay at gmail.com> changed: What |Removed |Added ---------------------------------------------------------------------------- AssignedTo|xorg-team at lists.x.org |nouveau at lists.freedesktop.or | |g --- Comment #1

Hi guys, I have one E1 with 30 channels in Mexico City, I guess that if i can fill this 30 channels with REAL traffic for 2 or 3 days I can find new bugs on chan_unicall or I can see how stable it can be. Im using R2/MFC with chan_unicall the patch that Steve Underwood wrote. I will let anyone make FREE LOCAL calls to Mexico City till saturday or maybe until monday to see how stable this can

Hello all, I?ve installed a proxy Squid in my gateway and a Cache DNS Server with bind. The problem is the server is only resolving is own querys but not the client queries from my company. When I do: $service named start I see in /var/log/messages: starting BIND 9.3.4-P1 -u named -t /var/named/chroot found 1 CPU, using 1 worker thread loading configuration from '/etc/named.conf'

] Hey ho, A few days ago we released the linux version of the phone, today we are very happy to have the mac version ready for a little field test. Freely downloadable from http://www.asteriskguru.com/tools/idefisk_mac.php At the same time, we also put a newer version of the windows and linux versions online. Let us know how you feel about it, a more mac look (brushed metal) is coming.

Hi all, I´m running a server that frecuently needs to open a pptp session with a remote server outside my Company. This server is running behind a Shorewall firewall and I don´t find information in Shorewall web page because there is no information in the link http://www.shorewall.net/PPTP.htm#ClientsBehind Nowadays I can connect this server with the remote one but te session is closed after

Hi. I'm having a terrible time with call quality coming into my * box. I'm using VoicePulse over a 1.5/1.5 mbit line. Outbound calls are crystal clear on both the RX/TX sides of the conversation. Inbound calls, though, are HORRIBLY garbled on the RX side. I can barely hear the caller, but they report my quality is fine. Getting loads of garbled sounds and weird echoes. (Could just be

For 64-bit X86 code we have had good success with using up-to 128-bit integers (this includes say 36-bit or even 2-bit integers). On Mon, Feb 2, 2015 at 4:03 PM, Alejandro Velasco <gollumdelperdiguero at gmail.com> wrote: > I asked a similar question last year here. Operations on types iN with no > direct translation into one assembly instruction seem to be translated into >

Ezra, Suppose a chatroom application, a RailsApp + 2 backgroundrbs: MiddleMan1& MiddleMan2, running in 2 machines. When Chater1 login, RailsApp call MiddleMan1 to get something about Chater1 from db to memory, like his contacts or other personal settings. When Chater2 login, RailsApp call MiddleMan2 do the same thing. But who is reponsible for the Global Data? like a Online Chaters

Hello Is it correct that IAX2 uses more CPU, than SIP? Also, can it be true that IAX2 is much more sensitive against high CPU loads? Also, does Asterisk support and use multiprocessor architectures, such as Xeon? ? Regards Jon -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.394 / Virus Database: 268.8.3/362 - Release Date: 12-06-2006

Hola, he estado buscando por todos lados, como puedo saber que traigo en un una variable que es un list, y pues nomas no encuentro nada. alguien sabe alguna función como dim, en caso de los data.frame que nos da las dimensiones, o length en un vector que nos da cuantos elementos tiene algo como names para saber el nombre de los objetos dentro del list, en esencia me gustaría saber si alguien

I'm new to VOIP. We are thinking of setting up a VOIP system between a couple remote offices. I've been lurking on this group for a while. What is the consensus on these phones: http://www.netvoice.ca/grandstream/budgetone101.htm I'm confused about the SIP protocol... can a SIP phone be located behind a NATing firewall ? When people use asterisk on a broadband connection used

hi, i facing a problem here. in my sip.conf, i specify a account like this, [1234] type=friend context=from-sip username=1234 secret=1234 nat=no canreinvite=yes dtmfmode=info mailbox=1234@default disallow=all allow=ulaw so i am able to login with username 1234 and password 1234 but ther weird part is, i can also register as any number (account) without having to specify in sip.conf. thus

I have tried to use native music on hold. In dir /var/lib/asterisk/moh-native/ I have some wav files (with 755 permission). In musiconhold.conf I have [native] mode=files directory=/var/lib/asterisk/moh-native And in sip.conf I have musicclass=native When I put call on hold this is what I get at CLI. -- Executing Dial("SIP/341-5931", "SIP/344|20|wWtT") in new stack

Hola, Bueno, puedes hacer el cálculo de una forma mucho más compacta y rápida. Esta forma es especialmente recomendable cuando tienes muchas columnas y muchas filas. > library(data.table) > myDT <- as.data.table(mtcars) > myDTlong <- melt(myDT, measure.vars=1:ncol(myDT)) > myDTlong[ , list(p_value = shapiro.test(value)$p.value, v_stat = shapiro.test(value)$statistic) , by

Hi All. I have notice a problem with FM feature (screen macros) on Asterisk CVS version. When call goes via IAX and calling part "accept the call" on Dial command with option M, in macros context it's setting MACRO_RESULT=CONTINUE, but anyway it hangups both channels. If anyone faced with such problem please let me know. I need to know whether it's bug or just configuration