similar to: FXO impedance matching

I understand that the X100P card is matched to a 600 ohm impedance but the UK BT phone system is not (I haven't been able to find much information on the impedance of the UK system). Has anyone come up with an easy way to match the impedance between the two so the X100P can work in the UK? Presumably a simple transformer won't do the job since it won't pass the DC components? --

Has anyone ever heard of this so called Dynamic Impedance matching on the ADIT 600? I called their support and they've never heard of it. We are of course having echo problems are on the far end due to digital/analog conversion on the local end using a channel bank. We have purchased an ADIT 600 and yes the complaints are "far less" however we're still getting them. While I have

I have a ATA 186 with SIP firmware 3.1 when I changed the configurations to use the g.726 codec I received many erros and the calls doesn't work. I changed the fields: - LBRCodec: 6 <- the code for g.726 - TXCodec: 6 - RxCodec: 6 The errors: Jul 9 13:15:37 NOTICE[1192491824]: rtp.c:500 ast_rtp_read: Unable to calculate samples for format G726 Jul 9 13:15:37 NOTICE[1192491824]:

I have a problem: When I'm in a call and a second call arrive (call waiting) I can't transfer the first call. If I press flash the line change to the second call, if I press flash again the line change to the first call. How I can transfer a call in this kind of situation ? Kind regards, Miguel

What FXO interface methods are folks using successfully in the UK? I'm looking for good, known-to-work solutions for commercial use for two PSTN trunks on an Asterisk box. Here's the options I have, as I see it: i) Two Digium X100Ps. Pro - cheap (c. ?120), CE approved. Con - UK line impedance mismatch, with resulting echo problems, plus needs two PCI slots. ii) Digium TDM400P with two

I donwloaded two IAX Clients (firefly and IAX phone) and they did register with *. It would make authenticated calls, but wouldn't actually register with the server. When I start the IAX Client the CLI show me the message: -- Registered '2004' (AUTHENTICATED) at 192.168.199.69:4569 After 5s: May 21 17:24:41 NOTICE[1133742896]: chan_iax2.c:5035 iax2_poke_noanswer: Peer

> > That would be called a milliwatt generator. It likely exists in their > central office, but its typically used by their technicians to ensure > new installations meet specs and sometimes in troubleshooting. Call your > telco repair number and see if they will give you the telephone number > for it. I'll try that. 1st line technical support at Telstra are not known for

More an echo algorithm question than a purely asterisk one... I have the following setup: Handset - PAP2 - Asterisk - SPA3000 - Telco And no matter what I do, I get echo on a call routed out via the PSTN when I talk into the handset, in the order of a hundred ms (my estimate, could be wildly inaccurate!). Echo will occur also when I have a handset plugged into the phone port on the SPA3000

Thank you very much for this hint. My apologies that I messed up a thread for my post - I had a message open and simply clicked on the link ... slap slap. Would anyone know of a better choice to multiplex three fxo lines into an asterisk box? I can still use three Digium X100P cards, but methinks, a seperate unit would be better. Thanks again, Peter >> I am located in the UK and

To me it's a error if I can't complete calls using the ATA configured to use the g726 codec. I just tried it usign a sipura SPA-2000 (preferred codec: G726-32) and I received NOTICES and WARNINGS, but I can't complete a call. On a zap channel: -- Executing Dial("SIP/2007-e4d8", "Zap/1/2217008") in new stack -- Called 1/2217008 -- Zap/1-1 answered

Muchas gracias a todos, estoy revisando las opciones que me dieron, Un abrazo, eric. On 10/06/15 02:29, miguel.angel.rodriguez.muinos en sergas.es wrote: > Hola Eric. > > Te recomiendo que le eches un ojo a gWidgets2: > http://cran.r-project.org/web/packages/gWidgets2/index.html > > Un Saludo. > > > El 09/06/2015 a las 22:07, eric escribió: >> Estimados, he

As Promised, I've released a new version of Firefly (ver 1.8) with IAX & SIP support back in. Get it from Virbiage site or here's the direct link http://www.virbiage.com/firefly/download/firefly-thirdparty.exe If it crashes on startup, export your Firefly tree from the registry (current user -> software -> firefly), then delete tree from your registry. If that fixes it, send

I have two contexts and there I have some sip clients and some iax clients, in the sip clients a have extentions like, 20, 21, 22, 23, etc; in the iax clients I have some extentions like 2000, 2001, 2002, 2003, etc. My extention is 2003 when I make a call the manager program show me that the extention 20 was originating the call (in red, and show my name/extention)! When I make a call to

Steve, Do you know if there are many differences between the Mexican and Brazilian variant ? Kind regards, Miguel Antonio Date: Tue, 05 Oct 2004 22:51:53 +0800 From: Steve Underwood <steveu@coppice.org> Subject: Re: [Asterisk-Users] RES: Working E1 MFC/R2 M?xico !!! To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID:

I am still having unacceptable echo on my X101P and twidling with the rx/tx gain levels and echo settings appears to have no discernable effect. Some questions for those who may have more significant electrical engineering background than I. 1. This impedance match thing ... will it affect this solution having other phones in parallel with the X101P? This is done so that I can test while not

Miguel, How many simultaneous calls (incoming and outgoing) you did ? Kind regards, Miguel Date: Sun, 3 Oct 2004 11:21:04 -0500 From: Miguel Cavazos <miguel@cavazos.com.mx> Subject: [Asterisk-Users] Working E1 MFC/R2 M?xico !!! To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID:

Hello R gurus, I have a data set from which i have to extract the gender and age matched rows from controls and disease group disease<-paste(rep(c('y','n'),11)) gender<-paste(rep(c('m','f'),11)) mcp<-rnorm(700,1400) age<-rnorm(32,34) dat<-data.frame(disease=disease,sex=gender,Dr_age=age[1:22],MCP=mcp[1:22]) I have other categorical variables also to

In my seemingly endless search for the cause of echo on my SPA3000, I wired it up in the following configuration: Analogue Handset <--> (FXS)SPA3000(FXO) <--> PAP2 And set the Line1 dialplan on the SPA3k to '(<:@gw0>S0)' which means that as soon as I pick up the handset I get linked straight through to the PAP2, which gives me dialtone. Even in this configuration, with

Hello there, I have been using pysieved and avelsieve and it has been working great. I decided to do test with the ManageSieve patch and got this problem: Nov 27 17:21:29 cambui dovecot: MANAGESIEVE(miguel): sieve-storage: using active sieve script path: ~/.dovecot.sieve Nov 27 17:21:29 cambui dovecot: MANAGESIEVE(miguel): sieve-storage: using sieve script storage directory: ~/.sieve Nov 27

offline? te lo envío a tu email, pq no creo que a toda la lista sea buena idea. El 15 de agosto de 2014, 9:47, Jorge I Velez <jorgeivanvelez en gmail.com> escribió: > De nada, Miguel. Es posible que me envies el archivo offline? --JIV > > > > 2014-08-15 17:45 GMT+10:00 Miguel Fiandor Gutiérrez < > miguel.fiandor.gutierrez en gmail.com>: > > Gracias