similar to: Help With Physical Layer

On Tue, Jun 30, 2015 at 3:34 AM, Tony Kasule <timotsmith at gmail.com> wrote: > Hello, > > Anyone to help me with this issue? It has never worked :( > > On Wed, May 20, 2015 at 11:34 AM, Tony Kasule <timotsmith at gmail.com> > wrote: > >> Hello users, >> >> I have a Digium Te235 and asterisk 13 which have worked well with 1 >> carrier but

What response do you get to *CLI> pri show spans ? On 30 June 2015 at 09:34, Tony Kasule <timotsmith at gmail.com> wrote: > Hello, > > Anyone to help me with this issue? It has never worked :( > > On Wed, May 20, 2015 at 11:34 AM, Tony Kasule <timotsmith at gmail.com> > wrote: > >> Hello users, >> >> I have a Digium Te235 and asterisk 13

------ Original Message ------ From: "Tony Kasule" <timotsmith at gmail.com> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com> Sent: 30/06/2015 8:34:47 p.m. Subject: Re: [asterisk-users] Help With Physical Layer >Hello, > >Anyone to help me with this issue? It has never worked :( Hi Tony I'm not

Hello, Anyone to help me with this issue? It has never worked :( On Wed, May 20, 2015 at 11:34 AM, Tony Kasule <timotsmith at gmail.com> wrote: > Hello users, > > I have a Digium Te235 and asterisk 13 which have worked well with 1 > carrier but we have failed to add a 2nd carrier. The second telco brings > their E1 line over finer, terminated in a RAD modem and they give

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

I am finding that I have to increase the txgain in zapata.conf to 8 when my X101P is connected to my BT phone line, otherwise people can hardly hear me. This then gives echo issues. Do other people have the same problem on BT lines. I was wondering if I should give BT a call and get them to increase the gain on the line. Strange though as the rxgain is OK and I don't have this problem with

> > 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

asterisk-users-bounces@lists.digium.com wrote: > In the following setup: > call coming from a pstn line -> into FXO card -> asterisk -> SIP > phone > > i get an incredible loud echo in the SIP phone (about 0,5-1s) > (everything i speak into SIP phone microphone i hear in its > speaker). The person calling from PSTN is not getting any echo. Make sure you're not

Tim At first should take a look to cable pinout (RAD documents) as pin 1,2, Transmit (output) and 4, 5 Receive (input) for Digium card you should use a straight cable (try to test with new cable one too). Second check Dahdi configuration parameters, use dahdi commands as; dahdi show status, service dahdi restart and check result, (could a mistake on parameter value on system.conf). Mc

Hi, Has anyone in australia got asterisk running on FreeBSD? how would i pass the opermode=AUSTRALIA parameter to the wcfxs.ko module as kldload doesnt let you pass parameters to the module like modprobe in Linux. I tried to get the sysctl variable using sysctl -a it might use but nothing applicable was visible.... i need to set the impedences to 600 ohms, can someone please advice me on

On Tue, Jun 30, 2015 at 2:15 PM, Duncan Turnbull <duncan at e-simple.co.nz> wrote: > > Hi Tony > > I'm not familiar with the card you but 120 ohm is usually twisted pair, > and 75 ohm is coax (usually). If it is changeable its usually done with > jumpers on the card. > The new Digium cards have no jumpers anymore and I don't think they support coaxial cables.

Hi list, Sometimes I receive a Red Alarm in my E1 trunk (E&M immediate start signaling), and just few seconds after this, all alarms are cleared. This problem ocurrs many times/day, and if are calls in progress, these calls just hang-up. Could it be an asterisk bug? Or may I contact the PSTN provider? Thanks Eduardo

I've been following the list for months, and I have a working Asterisk setup, but it'd be *really* useful to me at this point if someone could summarize when Asterisk has echo problems and when it doesn't. For instance, I usually hear a far-end echo when talking on my 7940, but not when using a POTS phone plugged into a TDM400 FXO port. It doesn't seem to matter if the call

RAD appear to have bucketloads of products which bridge between various interfaces (E1, BRI, POTS) and their own TDMoIP protocol. The attractive thing about them for me is their availability in Australia. The voip wiki says not much about it (http://www.voip-info.org/wiki/view/TDMoIP), and certainly nothing about if there is any way to get Asterisk to talk TDMoIP. Despite the name, TDMoIP

Hi all, Did anyone had success with Q.Sig on * 1.2, especially with Alcatel 4400 (which seems to only support Q.Sig) ? I am thinking about interconnecting 15 sites together with asterisk (probably using IAX or SIP). I have a very heterogeneous environement using both PRI and BRI, but my pilot will start with the Alcatel at the central site. Any help/example is most welcome. BR, - Patrick -

I'm just curious if anyone has the X100P & Callerid receiving working outside US. Replies are appreciated. Also if it's not working for you in a certain coutry you can respond too. regards Martin

Hi all: We are looking for someone with experience in Alcatel PBX - PRI - Asterisk integration Please get in touch off list.. We're wanting to hire a professional subcontractor, developer or company to get around some issues like these: Asterisk shows PRI to Alcatel is up, but when trying to dial from Alcatel to Asterisk results in a disc tone (Asterisk do send calls properly into

Hi All, This is our 2nd E1 client that we try to use crc4 either with the e100p or with the e405p without luck. After some trials, we ask the telco to switch off crc4 on their side and everything works flawlessly. Is there anything in the crc4 calculation that may be broken? We took a look at wct1xxx.c and wct4xx.c but there doesn't seem to be much there to be fixed (apparently the crc4

Hi all, I tried to connect the bristuffed(0.3.0-PRE-1i) * to an Alcatel PBX via BRI (zaphfc) and Q.SIG. The Alcatel PBX is connected to the outside world and should forward our calls to the telco. This setup works correctly as far as I use euroisdn as the switchtype. The first problem was that it is only possible to run the * side in CPE-mode -- I wanted NET. Anyway, I configured * this way:

I have re-labeled tick marks on the x axis. The problem is that by using axes=FALSE, the axes disappears and when they are called back using axis(side=1)..etc. the axis on sides 1 and 2 do not meet at the bottom left corner of the graph. I would also like to have the 3rd and 4th axes in there as well, all meeting in their respective corners. par(mfrow=c(1,2)) gut<-c("Full",