similar to: about trasncoders

Hello, I have transcoding card TC400P installed in server running Debian with Asterisk 1.4.23. Everything seams to be fine and after I boot up server I see in dmesg: 7.590966] Zapata Telephony Interface Registered on major 196 [ 7.590966] Zaptel Version: 1.4.12.1 [ 7.590966] Zaptel Echo Canceller: MG2 [ 7.610963] zttranscode: Loaded. [ 7.618969] wctc4xxp: tc400b0: Attached to

Hello, I need help to install a digium card TC400P. I compiled the dahdi source, but dahdi don?t find the card! debian:/usr/src/dahdi-linux-complete-2.2.0.2+2.2.0# lspci 11:03.0 Ethernet controller: Digium, Inc. Wildcard TC400P transcoder base card (rev 11) debian:/usr/src/dahdi-linux-complete-2.2.0.2+2.2.0# /etc/init.d/dahdi restart Unloading DAHDI hardware modules:

Hello all, I have a TC400B Digium card in order to deal with transcoding and I have some trouble using it, I have a timer synchronisation problem! I would be very grateful if you have any idea to help me? It seems that the card is not correctly synchronised to the system because when I speak to one side, the sound takes 5 seconds to go to the other side, and increasing, after 30 seconds of call,

Hi, I just read a pressrelease from VON that Digium will soon be releaseing a couple of new cards. What got me interested was: "The TE420P and TE415P support 128ms of G.168 (2002)-compliant echo cancellation across their entire 128 channels." Does anyone know when thease will be released and what they will cost when released? Thanks!

Hi All, I have recently installed a TC400B card into a system and am trying to get it to work. As far as I ca tell from the docco on Digiums website, there is no config as such unless you want to enable / disable only 1 codec, otherwise by default it runs as 92 channels of either. I have tried asterisk 1.4.9, 1.4.10 and 1.4.10.1 along with zaptel 1.4.4 and addons 1.4.2. The zaptel modules

Hi: How to install and set up my asterisk server with G723 codec to send and receive calls using it. Thanks in advance; Wassim _________________________________________________________________ Explore the seven wonders of the world http://search.msn.com/results.aspx?q=7+wonders+world&mkt=en-US&form=QBRE

hello friends, i have to configures asterisk n my hardware details are O.S - Ubuntu 8.04 Lts Memory - 1 GB Proccessor- core 2 duo is any one having a good link or how to related asterisk. any help,support will be higly appreciated thx -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi All, I need to replace old analog PBX with Asteriskl and X-Lise SIP SoftPhones as client phones. First: I have problems with implementation of PBX functions. I need and unsuccesfully tried theese functions (took info at http://voip-info.org/wiki-Asterisk+PBX+functions) Call Pickup: Supported in the standard installation (*8 - defined in res_parking.c +54) - Just don't understand how to

Hi, In voicemail.conf, you can read ; Look in /usr/share/zoneinfo/ for names of timezones. ; Look at the manual page for strftime for a quick tutorial on how the ; variable substitution is done on the values below. Where can this manual page for strftime be found ? man strftime and apt-cache search strftime don't reply much ... Regards -------------- next part -------------- An HTML

Hi I have On func_odbc [EXEC] readhandle=ressqlserver writehandle=ressqlserver readsql=${ARG1} writesql=${ARG1} I'm trying an update on dialplan: exten=> 141,3,Set(dummy=${ODBC_EXEC(UPDATE Tabla set campo = ${EXTEN})}) On Cli: WARNING[3579]: func_odbc.c:353 acf_odbc_read: Error -1 in FETCH [UPDATE Tabla set campo = 4356] Any idea why is this?? The query

Hi, Say I wanted to know what context a SIP registration is using to dial out in my dialplan, what would I do? For example, I have phones on a "local-calls-only" context (as defined in sip.conf), others in "unrestricted-calls". In my dialplan, I`d like to act on that knowledge. Mike -------------- next part -------------- An HTML attachment was

Hi, Here is the next iteration of the patch. The only comment not addressed is this one: > It would be better to implement a target-independent check for > overflow for the "Legal" case (like how SADDO does). Hacker's > Delight > has some hints on how to do this. It's not easy for the signed case, > but is do-able. It can be lowered to a division + a branch,

[Chris asked me to bring this up on the mailing list some time ago, but I couldn't get to it. Sorry for that.] Since the beginning, I used ReST [1] for documenting llvmc, instead of plain HTML that was used historically. In my opinion, ReST is much easier to write and read (in the text editor or on terminal); it can also be used to produce PDFs, man pages or HTML that looks exactly the same

Have a nice day, Scott Berry E-mail: N7zib at northlc.com I am studying out of the book Asterisk: The Future of Telephony on Chapter 4, and right now for practicing using the built in Debian version of Asterisk for Ubuntu. I am however having some problem where I cannot do "asterisk -r" and hook up to the asterisk CLI. I have checked to see that

When I attempt to create a 46 disk pool with 5 and 6 disk raidz vdevs, I get the following message: mismatched replication level: both 5-way and 6-way raidz vdevs are present mismatched replication level: both 6-way and 5-way raidz vdevs are present I expect this is correct.[1] But what does it mean for performance or other issues? Why am I being warned? The factory config for x4500s had the a

Hi, In voicemail.conf: ; Supported values: ; 'filename' filename of a soundfile (single ticks around the filename ; required) ; ${VAR} variable substitution ; A or a Day of week (Saturday, Sunday, ...) ; B or b or h Month name (January, February, ...) ; d or e numeric day of month (first, second, ..., thirty-first) ; Y Year ; I or l

Hello everybody, while developing an application for the company I work for, I noticed that very often libtheora reports division by zero errors. I fixed those errors by adding a few additional checks, but I'm not 100% sure that these checks are correct. Could you please check out the attached patch file and tell me what you think? Best regards, Marco Giovannini -- /*

Hi, I've read that the new dahdi channel supporting B410P doesn't support PtmP when in NT mode. Does it support PtmP when in TE mode ? Cheers -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081209/f422b5d0/attachment.htm

On Tue, Dec 9, 2008 at 6:11 AM, Zoltan Varga <vargaz at gmail.com> wrote: > Hi, > > Here is the next iteration of the patch. The only comment not > addressed is this one: > Thanks! It's looking good. >> It would be better to implement a target-independent check for >> overflow for the "Legal" case (like how SADDO does). Hacker's > Delight

2008/12/9 Olivier <oza-4h07 at myamail.com> > > > 2008/12/9 Tilghman Lesher <tilghman at mail.jeffandtilghman.com> > > On Tuesday 09 December 2008 09:14:11 Olivier wrote: >> > Hi, >> > >> > In voicemail.conf: >> > ; Supported values: >> > ; 'filename' filename of a soundfile (single ticks around the >>