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Dear listers, I'm facing a puzzling situation with Digium TDM2400 card (12 FXO / 12 FXS). Once a day or so we detect 1 or 2 zombie FXO channels. These can be either outbound or inbound calls. I thought this could be related to obsolete DAHDI or Asterisk versions, so I upgraded to 2.4.0 and 1.6.2.15 respectively (OS: Ubuntu 10.04 64 bits). To no avail; the zombie channels keep showing up.

Hello everyone, We are using an Asterisk server (Ubuntu 10.04 64 bits, Asterisk 1.6.2.7, TDM2400 - 12 FXS/12 FXO), but I noticed the following dmesg alert: asterisk[4687]: segfault at 60 ip ****** sp ****** error 4 in asterisk[400000+187000] The server is running without troubles. I know I should worry, just don't know where to start. Any advices? Alex Saavedra -------------- next part

hello everyone, I would like to as for advice for the use of ?lmer? (package ?lme4?) and writing the proper syntax to best describe my data using a mixed-effects model. I have just started to use these models, and although I have read some good examples (Extending the Linear Model with R, Faraway 2005; and the R book, Crawley 2007), I am still not sure of the syntax to test my hypothesis.

Hello everyone, I have noticed thar our dahdi-channels.conf has some repeating directives, for instance for channel 2 (FXO) we have these settings: ;;; line="2 WCTDM/0/1 FXSKS" signalling=fxs_ks callerid=asreceived group=0 context=from-pstn channel => 2 callerid= group= context=default As you can see, a few directives are repeated (callerid, group, context). This was generated by

see a copy of your rsyncd.conf? Tim Conway tim.conway@philips.com 303.682.4917 Philips Semiconductor - Longmont TC 1880 Industrial Circle, Suite D Longmont, CO 80501 Available via SameTime Connect within Philips, n9hmg on AIM perl -e 'print pack(nnnnnnnnnnnn, 19061,29556,8289,28271,29800,25970,8304,25970,27680,26721,25451,25970), ".\n" ' "There are some who call me....

Hello, I just upgraded DAHDI from 2.3.0 to 2.4.0, both dahdi-linux and dahdi-tools (Ubuntu 10.04 64 bits). It took a few minutes, and it was straightforward. Everything is working. Now I'm preparing to upgrade Asterisk from 1.6.2.7 to 1.6.2.15. I made a backup of configuration files, codec licenses and CDR. Is there something else I should be aware of before upgrading? Thank you, Alex

Leg/Transaction Does Not Exist" and obviously not taken into account as endpoint GUI remains unchanged. Looking deeper into this here are : NOTIFY message accepted by S450IP NOTIFY sip:7531 at 192.168.100.197:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.100.254:5060;branch=z9hG4bK06adc48b;rport From: "asterisk" <sip:asterisk at asterisk>;tag=as4ea953db To: <sip:sip:7531 at

crashing. So, as a first step to solving **that** problem, make sure asterisk is compiled with debug flags, dumps another core file, and then you do the "gdb asterisk <corefilename>", and get a stack trace. That should give us some idea of what happened. > > I have a fairly simple Followme sequence in place to see how it works > before I get into the complex scenarios.

That's the main reason I opened this thread as it surprised me a bit ... > > > Any 2-wire analog leg will be a source of echo. Many, many, many calls > do not have a 2-wire leg. Even in handset audio circuit ? I was thinking that any handset is a potential echo source due to this audio circuit ... Do you agree ? > Think cell/mobile or endpoints with PRI or T-1. > >

Hi, I'm trying to connect an HDL F10 device for a friend living in Brazil to the TDM2400 on his Asterisk server. That device should behave like a normal doorbell and it is if connected to an analog PBX. I connected to the TDM2400 and everything works fine except for one thing: when the called party hangs up his phone, the F10 HDL device does not hang up. I'm not brazilian and not

Hi List... I have an asterisk box with one TDM2400, it has 8 FXS's and 12 FXO's, and it also has the echo canceller... I'm running Asterisk 1.2.13 and Zaptel 1.2.12 with the Kernel 2.6.9-34.0.2.EL I'm using Polycom's 501 with the SIP 1.6.2.0041 The problem is when someone dials to or from the PSTN through the TDM2400, the voice quality is crappy...Instead of hearing:

sort of standard for getting media players to support dynamically mixing different tracks and also making it easy for artists to do. On Mon, Aug 18, 2008 at 7:09 PM, Andy <andycool22 at gmail.com> wrote: > i'll chime in and say that i would love to get music recorded in > separate tracks, maybe there would be some kind of settings embedded > in the files so i could hear them

sort of standard for getting media players to support dynamically mixing different tracks and also making it easy for artists to do. On Mon, Aug 18, 2008 at 7:09 PM, Andy <andycool22 at gmail.com> wrote: > i'll chime in and say that i would love to get music recorded in > separate tracks, maybe there would be some kind of settings embedded > in the files so i could hear them

sort of standard for getting media players to support dynamically mixing different tracks and also making it easy for artists to do. On Mon, Aug 18, 2008 at 7:09 PM, Andy <andycool22 at gmail.com> wrote: > i'll chime in and say that i would love to get music recorded in > separate tracks, maybe there would be some kind of settings embedded > in the files so i could hear them

Dear Mr. Baddeley   I just graduated from a Masters in Applied Mathematics on Jun19th. My thesis was about spatial distribution /a nalysis of some trees in a part of  the Congo Basic Forest. In my thesis I used your spatial package in R, and today I'm doing a more deeper study about how we choise the r distance in Function F(). I would like to be able to understand about value of rmaxdefault

1. Is it normal to see : # lsmod Module Size Used by dahdi_dummy 3236 0 Shouldn't it be used by asterisk or is this 0 value meaning something specific ? 2. How can you check dahdi is running ? Here, "ps aux | grep dahdi " replies "grep dahdi". Cheers ------=_Part_2692_19661943.1228286635399 Content-Type: text/html; charset=ISO-8859-1

<div class="socmaildefaultfont" dir="ltr" style="font-family:Arial;font-size:10.5pt" ><div dir="ltr" style="font-family:Arial;font-size:10.5pt" ><div dir="ltr" >Matthias,</div> <div dir="ltr" > </div> <div dir="ltr" >Thanks for the response.</div> <div

<div class="socmaildefaultfont" dir="ltr" style="font-family:Arial;font-size:10.5pt" ><div class="socmaildefaultfont" dir="ltr" style="font-family:Arial;font-size:10.5pt" ><div dir="ltr" > </div> <div dir="ltr" >Hi all,</div> <div dir="ltr" > </div> <div

<div class="socmaildefaultfont" dir="ltr" style="font-family:Arial, Helvetica, sans-serif;font-size:10pt" ><div dir="ltr" >&nbsp;</div> <div dir="ltr" ><div dir="ltr" >Hi,</div> <div dir="ltr" >&nbsp;</div> <div dir="ltr" >I am trying to understand libvirt

Is there a way I can change the ring cadence on FXS lines on a system using a Digium Wildcard TDM2400 card? I recently deployed a new phone system, and the customer has a few POTS phones in areas where they don't have data network services, so we're using the FXS lines to provide dialtone at those outbuildings. The old phone system would ring those phones with a short ring-short