similar to: Warning: No path to translate with SJPhone

Hi List!! Thanks for the colaboration, especially to Richard Cavanna who gave me the necessary support. I followed your indications and the comunication was better for the test users. The warning indication is no jumping anymore and the voice is not delayed. This is my sip.conf: [general] context=default ;allowguest=no ;realm=mydomain.tld bindport=5060 bindaddr=0.0.0.0 srvlookup=yes

Hola, Otro tipo de ficheros estándar y estructurados es XML, que R puede importar directamente con el paquete XML. Dependiendo de la interoperabilidad necesaria, los metadatos pueden ir en el propio fichero o utlizar ficheros DTD. Eso sí, es un poco más estricto y complicado que csv o JSON. Un saludo, Emilio L. Cano _____________________________________ Emilio L. Cano http://www.proyectum.es

Hola a todos, Tengo un fichero CSV que se genera desde una aplicación, y que me gustaría importar en R. El problema es que el fichero es "genérico", por lo que en un principio no se muy bien el tipo de datos que me voy a encontrar. Desde R se los mostraré al usuario para que decida sobre qué campos quiere hacer qué análisis. Por tanto, en el propio CSV debería contener (en la cabecera,

Hi i tray install poser 7 on wine 1.1.1 and my hardy heron 64 bits , the installation is fine , when i run it ., the screen is deform only can restart but i have a impr paint http://img147.imageshack.us/img147/2600/pantallazo2cb1.png , why deform my screen and the impr paint it's fine ? , well apply gatito at gatito:~$ WINEDEBUG=warn+all wine poser.exe for debug and this is my lines : gatito

New to this windows domain stuff, sorry ( at my age learning new stuff can take a while ). I've set up a domain and joined a couple of XP workstations to is and all is fine. What I want to do now is to ensure that the users of these PCs still have administrative rights on their PC's. Can anyone show me the basics / point me to a good guide on how to do this??? TIA, Steve -- Steve

Hi, This works for me ,you can try. After join computer to domain then log on to Windows Xp with local administrator account and go to control panel -> addusers (select account from your domain) -> Grant access level to your domain account as "Administrator". Or you can use "net" command to do this. Open a cmd shell, then execute: C:\> net localgroup

Has anyone any tips on improving samba performance with debian lenny? I've set up a raid 0 partition on a couple of new WD 1TB disks, formatting the majority of it as a single reiserfs partition. I must admit I should probably have set the blocksize to something other than the default at the time, but now it's well populated, changing stuff like that is going to be a big problem.

Does anyone have experience with the upgrade from 3.2 ( in lenny core ) to the samba provided 3.3 debian packages? Specifically, I'm looking for a list of gotchas in this process, as I'm short on machines I can use as test servers... Cheers, Steve -- Steve Holdoway <steve at greengecko.co.nz> http://www.greengecko.co.nz MSN: steve at greengecko.co.nz GPG Fingerprint = B337 828D

Hola, Vaya desastre al mandar la cabecera de la base de datos. Al enviarlo, se veía bien, pero está visto que por el camino la cosa se torció. Siguiendo el consejo de Javier, he subido un pequeño fragmento al DropBox. He quitado algunas especies para no hacerlo muy largo. Espero que ésta vez no haya problemas. https://www.dropbox.com/s/q7zla50oq7owg8k/CPUE.csv?dl=0 Un saludo y gracias Juan

Hi group I wrote 2 years ago to know if there is some workaround for PacketCable. Since then I got no answer and now I hope there's something about. Is there any chance to use Asterisk as softphone with cable modem technology using Packetcable? Thanks in advanced Carlos Bernat -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi List !! I have a lot a questions about this incredible tool but short is my time to learn it, so I apologize if my last question was too general. I got another more especific trouble. I administrating an ISP and I have my Asterisk installed on a server for testing my network performance. I followed the quick-start tutorial provided by voip-info.org (which I think it's very useful) and

Hi all, I am happily using wine (0.9.58) on Opensuse 10.2 to run a few MS packages, including Ashlar's Drawingboard. Recently I got a mini Asus eeePC (with Xandros derivative debian) on which I also installed wine from the Debian Etch archive. The latest version available was also 0.9.58. One program is running fine (Quicken) on the eeePC, but Drawingboard throws up a "Panic: Internal

I added the line "allow G723.1" in my sip.conf general config, and from a bridge connection which gives silence, I have progressed to the error message below, and the call gets rejected. help!! Dave ps. 217.168.168.49 : soft sipphone, i'm trying SJphone & Pingel Instant Expressa 723@216.52.153.207 : Go2Call SIP gateway -- Executing

En un principio, en los ldif que tenemos de los usuarios de prueba, tenemos el campo mailQuota. Sin embargo, si nos vamos a los schemas de qmail medianamente, con algo m?s de 3 a?os, se cambi? mailQuota por mailQuotaSize y mailQuoatCount. Esto se debe entre otras cosas a que en la cabecera del fichero maildirsize del buz?n, tenemos esta informaci?n, es decir, m?ximo tama?o ocupado y m?ximo n?mero

Hello. I have a codec problem to send calls from a SIP device to a H323 gateway. First I'll explain the scenario: - Asterisk 1.2.1 - The SIP phone can use any codec I want. - The H323 gateway can only use g729 (cause it's not under my administration) - SIP phone has g729 configured, so my asterisk doesn't need to "transcode" (I don't have licences for g729) - sip.conf

Hola Klaus y Jorge , gracias por responder. Intente eso mismo y me borra la columna "y", yo quiero eliminar la columna que se genera en los data frame que contiene el número de registro. En el ejemplo de abajo esa columna va de 1 a 4, cuando exporta el data frame se genera esa misma columna (que no tiene cabecera) Años Edad Dpto 1 Bueno 16,75 1 2 Bueno 16 1 3 Malo

To begin with, I am new to both asterisk and VOIP and although I've gotten pretty far with my Asterisk setup and have two different sip accounts working fine for outgoing calls I am having trouble with one issue. My problem is that I have another provider who uses IAX2 that I wish to use for calling various countries, including local (The Netherlands) calls and calls to the US to

I am having a problem with asterisk transcoding GSM and G729 codecs, the error message is below: Jun 14 09:38:12 WARNING[18292]: channel.c:2693 ast_channel_make_compatible: No path to translate from SIP/3004-fcfb(256) to SIP/3003-c1c3(2) Jun 14 09:38:12 WARNING[18292]: app_dial.c:1586 dial_exec_full: Had to drop call because I couldn't make SIP/3004-fcfb compatible with SIP/3003-c1c3 ==

Hi all, I install the last version of Asterisk and I tried to send faxes, but nothing works. Here is my configuration: Analog Fax <----> IP <----> Asterisk <----> IP <----> Patton M-ATA <----> Analog Fax 2 I tried Analog Fax 2 -> Analog Fax but nothing works!! In the Patton configuration I put G711 and no silence suppression. In asterisk I have

Hi i am trying to do the same thing: receive a call from a cisco callmanager and forward it to a SIP user. Asterisk is compiled with h323 support, and is configured as a gateway in the cisco callmanager. h323.conf: [general] port = 1720 bindaddr = 193.x.x.x ; this SHALL contain a single, valid IP address for this machine allow=all extension.conf: exten = 3298,1,Answer exten =