similar to: To connect between more than 2 asterisk server [ links needed ]

Hello All, I am trying to figure out which way to go for a quad port fxo solution with a good echo can on it. My options are the sangoma remora, a mediatrix fxo, or something similar. The issue is that I would need a good EC. This would be on about a 9000 foot loop, and the lines don't function well on a spa-3000 or zaptel tdm 4 port card. Anyone have experience that drives them in a

I get the following errors when starting Asterisk. == Parsing '/etc/asterisk/zapata.conf': Found Feb 19 21:14:35 WARNING[10440]: chan_zap.c:920 zt_open: Unable to specify channel 1: No such device Feb 19 21:14:35 ERROR[10440]: chan_zap.c:6860 mkintf: Unable to open channel 1: No such device here = 0, tmp->channel = 1, channel = 1 Feb 19 21:14:35

When I switched to 1.8 from 1.4 I am getting this error pbx.c:4055 pbx_extension_helper: No application 'Wait,1' for extension (default, s, 1) http://www.voip-info.org/wiki/view/Asterisk+-+documentation+of+application+commands This page says its in 1.0 and I dont think has been removed. Did I do something wrong? Everything else seems to be ok. Thanks, Jerry

Hi Johannes, Thank you for your quick feddback. As I am using linux os build with buildroot on my raspberry pi I can't use package manager such as apt-get to install R. What I need to do is to get the precompiled R for arm arch and then when I build my linux image using buildroot I include the R precompiled package to my rootfs. Could you please send me a link where I can download

Hellow everyone, here's an other newby question. I've got a * configured with the card in the subject line. At times Asterisk fails to notice a disconet from the incoming line going into one of the FXO ports. Consequently it just keeps the line off-hook for ever and that causes my provider to mark the line aut of order. Is there any way to "help" Asterisk notice the disconect?

The maintenance (adding RAM) took a little longer than I planned: 1) Shutdown - 1 minute 2) Open Case - 30 seconds At this point, I emember that I can''t add RAM to this box without removing the Mother Board (hinge-mounted in case) - slap forehead. 3) Remove cables, PCI NIC & MB - 2 minutes 4) Add RAM - 1 Minute 5) Get the %$#@ MB back in the case and aligned -- 15 Minutes 6)

Hi, Is there a way to limit the duration of a call in the Dial command? Mainly for perpay account. Thanks __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I have small issue with Asterisk. My customers complaining that sometimes (not always) the outgoing voice (the voice which can be heard by the user a the other end) quality is very low (stutter and sudden clicks). The problem exist in only-IP configuration and in IPtoTDM connections as well. I use alaw codecs. I know that they consume a lot of bandwidth, but the upload and download stream is

I went to the Linux Solutions exhibition in Paris yesterday, visited the well stocked O'Reilly stand and saw a nice pile of Asterisk TFOT, 6 hours later there was only one left. It must say something, also it was the English version. -- Dave Cotton <dcotton@linuxautrement.com>

Hi all: Can someone give me a suggestion. I am using Ubuntu on my desktop and intall a program in wine called talkingbuddy which is a desktop bot that gives the weather time etc. The program installed fine but the animated character does not show up just a box that has a 2 on it and where the character is suppose to be in the desktop there is a gray box. I am not sure how to let the

I had instaled winrar3.51 and I get this errors: fixme:actctx:CreateActCtxW stub! err:rebar:REBAR_AdjustBands Phase 1 failed, x=692, maxx=-4, start=0, end=0 err:rebar:REBAR_AdjustBands Phase 1 failed, x=692, maxx=-4, start=0, end=0 err:rebar:REBAR_AdjustBands Phase 1 failed, x=40, maxx=-4, start=1, end=1 err:rebar:REBAR_AdjustBands Phase 1 failed, x=692, maxx=-4, start=0, end=0

Hi the list, I need to include some C code in R, but the behavior of the C code is strange : Here is my code : --- 8< --- Rprintf("\n XXXX mTraj=%f mClus=%f",mTraj[i+nbId*c],mClustersCenter[j+nbClusters*c]); Rprintf("\nDistA=%d Tmp=%d",dist,tmp); tmp = mTraj[i+nbId* c] - mClustersCenter [j+nbClusters* c]; Rprintf("\nDistB=%d

To close debian bug #271052, http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=271052 I went ahead and did the following: $ diff -u ../original/speex-1.1.6/libspeex/nb_celp.c libspeex/nb_celp.c --- ../original/speex-1.1.6/libspeex/nb_celp.c 2004-07-15 01:16:52.000000000 -0400 +++ libspeex/nb_celp.c 2005-02-27 08:24:49.000000000 -0500 @@ -1746,6 +1746,21 @@ st->submodeSelect =

Dear all, I am currently trying to cross compile R3.5.2 using X86 host and trying to cross compile for ARM target (raspberry pi 3 running on linux system generated by buildroot). The problem I am encountering is that the compilation fails because the current R buildsystem I trying to execute the R binary generated for Arm arch on the host X86 computer. My question is: Is there an R

Wow ! No reply... May be I must talk about X100P instead of X101P ? Is someone has yet encountered this kind of "no sound" problem when bridging two FXO lines like this (first as input, second as output) ? Any idea ? TIA. Best Regards, Francois BERGERET, France. ----- Original Message ----- From: "Francois BERGERET" <f6hqz-m@hamwlan.net> To: "Asterisk Users

Hi, i have a very huge number of data with the size 2375ko, i want to import them for R to xlsx but the size of excel is limited How can i resolve this problem? And please how can i define the frame Note that i have a table rows times c(1.....100) columns WLc(200...1000) and inside S c(15,.........) the table it is full values -- View this message in context:

Hi, Have someone there tried the TDM 2400 with 24 FXO? Have had echo problems? or any other problem ? Recommendations? Optional echo cancellation modules are necessary? TIA, Fernando

Hi Yair, Please let me if you managed to fix the DTMF tone issue, which you were experiencing couple of months ago. If not can you share any advancement. I'm currently experiencing the same issue, I can make outbound calls but DTMF will not work when dialing IVRs. My configuration is asterisk@home 1.5, registering to Voip provider (Symbio), codec is g.729 and dtmf mode is set to rfc2833.

Hi, I have the following situation: I've installed Asterisk at Machine 1 (M1 - IP: 192.168.1.145) and X-Lite (X_lite-Xten-Win32-1103m.exe from www.xten.com) at Machine 2 (M2 - IP: 192.168.1.100) and Machine 3 (M3 - IP: 192.168.1.200). I need to catch the SIP and MGCP messages that will appear when M2 calls to M3 and vice versa. The SIP messages are working (I don't have problems with the

Hi, I have 2 g729 licences - what I want to do is use g729 by default but if I get more than 2 calls at a time, use gsm for the others. So, I put this on all my sip providers: disallow=all allow=g729 allow=gsm However, this just seems to use gsm for everything. If I comment out the gsm line, it then uses g729. I thought it would use the codec's in the order they are allowed - is this