similar to: Asterisk on Zonbu, impact of transcoding

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I have 12 T1's going into 3 servers, 4 in each into "Digium, Inc. Wildcard TE410P Quad-Span togglable E1/T1/J1 card 3.3v (rev 02)" cards. Each "group" of T1's have the primary D on 24 and the secondary D on 96. The first server (ts20) and the last server (ts22) can playback "demo-congrats" fine. The "middle" server (ts21) cannot -- just dead air.

I have a problem where SIP calls from some providers are dropping at 15 minutes. The topology is: Client sends calls to a host running OpenSIPS, OpenSIPS sends calls to an Asterisk server. Below, 'Client' is the IP address of the client's host (running FPBX-2.8.1(1.8.20.0) 'OpenSIPS' is the IP address of my host running OpenSIPS 1.7.2-tls 'Asterisk' is the IP

Hi, I'm working with panel data from the Swiss Houshold Panel (SHP). The data i got came in the following way: 1.) 12 *different* /individual/ files - one for each year . 2.) 12 *different* /houshold/ files - again: one for each year Each file came in the SPSS format (.sav). I implemented all the files in R an managed (via rename, cbind, rbind, merge etc.) to get *two* files. The first file

I am configuring a test Asterisk server (1.8.9.2) to practice setting a single codec globally, to avoid transcoding as much as possible. Since all of my recordings are in gsm format, I am trying to make the SIP clients use gsm everywhere. I am using Ekiga on Fedora 16 x86_64 for my tests. [root at elx2 asterisk]# cat /etc/asterisk/sip_general_additional.conf

Hi the list, unlist respect the all the atomic type except orderd (it change of ordered into factor) : ### integer class(unlist(list(1:5,1:3))) #[1] "integer" ### numeric class(unlist(list(1.2,3.5))) #[1] "numeric" ### character class(unlist(list("e","e"))) #[1] "character" ### factor

Dear List, I ran into some problems with time-series-Data. Imagine a data-structure where observations (x) of test attendants (i) are made a four times (q) a year (y). The data is orderd the following way: I y q x 1 2006 1 1 1 2006 3 1 1 2006 4 1 1 2007 1 1 1 2007 2 1 1 2007 3 1 1 2007 4 1 2 2006 1 1 3 2007 1 1 3 2007 2 1 I am looking for a way to count the attendants that at least have

Hi. I am using a large storage which is ext2 file system of 1T byte. It has many file types, doc, mail data, zip, etc. It doesn't have a database file. I want to change the file system from ext2 to ext3. It also can't divide to any partitions. How to decide journal mode and journal size? Please advice to me. Recommend a journal mode. (orderd or journal) Recommend a journal

Does anyone know how to get CID working in AEL2 ? In extensions.conf you can do: exten => 111/666,1,PlayBack(demo-congrats) exten => 111/666,2,Hangup() exten => 111,1,PlayBack(demo-moreinfo) exten => 111,2,Hangup() and if callerid 666 dialed 111, they would get demo-congrats, everyone else gets demo-moreinfo. In AEL: 111 => { Playback(demo-moreinfo);

Hi! I just installed Asterisk 0.4.0 with all the default options, and the configuration samples it has. When I try to dial from an h323 client (gnomemeeting) I get this message on the messages file: Jun 19 11:48:45 WARNING[15375]: File file.c, Line 410 (ast_openstream): File demo-congrats does not exist in any format Jun 19 11:48:45 WARNING[15375]: File file.c, Line 553 (ast_streamfile):

actually, i do have a workaround which bypasses the exec command entirely: system("asterisk -r -x 'add extension s,3,Playback(demo-congrats) into local'"); but it's ugly. seems like it should be possible to do this with exec. .t ---------- Forwarded message ---------- Date: Sun, 23 Nov 2003 21:17:50 -0500 (EST) From: tad <tad@media.mit.edu> To:

Hello, I have data in the following form age sex Int.Prev.Est.1 Int.Prev.Est.2 Int.Prev.Est.3 Int.Prev.Est.4 Int.Prev.Est.5 93110 93 0 23.75482 57.86592 9.755003 4.343534 4.280714 93610 93 1 53.36475 39.47247 4.381618 1.622119 1.159044 94110 94 0 23.47514 58.23936 10.789339 3.690415 3.805741 94610 94 1

Hi, I have cvs updated all my modules (zapata, libpri, zaptel and asterisk). I have also read in the archives & seems that no-one has run into this problem. What I'm trying to do is simple. Just make and outbound call using the /var/spool/asterisk/outgoing directory. I copied /usr/src/asterisk/sample.call and only changed the context & extension. I configured my Zap1 to the same

Guys. I just unpacked on of the new spa841 I orderd and I was changing the ringtone (and listening to the options) when suddently the phone stopped playing back the tones and now the phone doesn't ring, speaker doesn't work and no ringtone play can be heard. Has anybody had this kind of problems?

as i said, right now i'm just getting my feet wet. but, i will be needing to build dialplans on the fly. 'add extension' seems like the right call to make. .t > What is the goal of this? It doesn't make much sense to me. Care to > share some insite into what your goal is? > > bkw > > On Sun, 23 Nov 2003, tad wrote: > > > actually, i do have a

Hello, The problem is that the factors are not orderd by the row number. If you want to check their order, use str(sortdata) and you'll see Santa-Rosa was attributed factor level 4 (in the output, first variable, the 3rd and 4th). Try the following. sortdata <- read.table(text=" county year x1 x2 x3 x4 x5 x6 x7 rank 141 Escambia 2002 6.50

I am recieving inbound calls to my asterisk box over IAX. And getting "spawn extension....exited non-zero" errors. Im not entirely sure what this means, and would appreciate any help in fixing my problem. FYI: ********** is the inbound phone number x.x.x.x is a remote asterisk box calling my own asterisk box. When I choose it to dial an internal extension using this dialplan: exten

I am using asterisk 1.4.18 with a polycom phone. sip.conf has: [532] type=friend username=532 secret=XXX dtmfmode=RFC2833 host=dynamic context=smvoice-sip callerid=532 qualify=no nat=no disallow=all allow=ulaw allow=alaw allow=gsm canreinvite=no I call into the dialplan and try to play demo-congrats and I hear nothing. Firewall is disabled. Everything is on the 192.168.1.X network for this

My first Asterisk install: Debian sarge with the 2.6 kernel, and two X-Lite soft-phones. I followed the online how-to documents and was calling between the two soft-phones and calling the demo system with no problems and had full audio. I then went on to configure the TDM400P's two FXS modules. I got into that a ways and was having some success, but no dial-tone when I was off the

Hello list, this is my first post on this list. I have a server with Asterisk and a Linksys SPA3102 with 3 SIP phones. I have configured the SPA PSTN line as trunk to receive and send calls. I can call outside from SIP phone throw the PSTN line and all is OK, the problem is when I receive a call from the PSTN, on the out caller phone there is a demo playback. I want to redirect the call to a