search for: gimeno

Hello, I'm having difficulties to make Asterisk (1.6.0.6) interoperate correctly with a CISCO mediaGW in order to send faxes to the PSTN using T.38. When Asterisk sends the initial INVITE containing the T.38 media offer in the SDP, the CISCO answers with a 488 Not Acceptable Media. Apparently, it looks like a configuration problem in the CISCO, but I have tested the CISCO with the Zoiper

...38]V21-preamble |<-------------| |[t.38]DCN | |<-------------| |BYE | |<-------------| |200 OK | |------------->| Any idea of what might be happening? Thanks in advance. Best regards, Santiago Gimeno The relevant information in the asterisk configuration files is: extensions.conf [fax-in] exten => 99999,1,Set(INCOMING_FAXFILE=/root/santi/fax/incoming.tif) exten => 99999,n,Answer() exten => 99999,n,Wait(3) exten => 99999,n,ReceiveFax(${INCOMING_FAXFILE}) sip.conf [general] ca...

Hello, In my scenario, the asterisk machine is installed behind a CISCO mediaGW in order to be able communicate with the PSTN. Asterisk is configured to use T.38 to send and receive faxes. I'm trying to receive a fax from a fax machine located in the PSTN. Apparently everything goes well: the fax machine says the transmission was successfully completed, and the fax file is successfully

Hello, I'm experiencing a weird problem when using the VoiceMailMain application. If I use the application after dialing a Local channel, there's strange beep just after asterisk answers the call and before the first locution. The extensions.conf I'm using is: Ruido extra?o al llamar a la aplicaci?n VoiceMailMain [default] exten => _X.,1,Dial(Local/${EXTEN}@test) [test] exten

Tengo un problema que me dice: La libreria MASS de R contiene el fichero birthwt.Trabajando sobre estos datos: 1.-Recodifica las variables tal y como aparece en la ayuda sobre este fichero en R. 2.-Analiza e interpreta la posible asociacion estadistica entre las variables low y smoke. ¿Alguien me podria explicar como hacerlo? Ademas, si alguien puede, no entiendo la validacion cruzada, ni la

Hello, I have been experiencing audio problems when accessing the Voicemail application using DECT phones and the g729 codec. The issue is that whereas the vm-password is always played correctly by the DECT phone, the rest of audio files, randomly, are played or not by the DECT phone. Everything works correctly if another codec (alaw,ulaw) is used. I have noticed that asterisk doesn't send

Hello all, Not sure if this mail belongs to this users or dev list. Sorry about that. We have the following scenario: PhoneA OpenSER Asterisk PhoneB PhoneC | | | | | | | | | | | | | |

Hi, I'm having problems when the callerid of a user defined in the sip.conf contains special characters such as: ?, ?, ?, ?, ? , etc. The strange thing is that these characters are displayed correctly in the dialplan by using the sip show peer command, but if this user makes a call, these characters are not displayed correcly in the SIP message. Any ideas of what might be happening? Thank