search for: enable_vad

...lnet_level: "2" ; 0-Disabled (default), 1-Enabled, 2-Privileged # Phone prompt/password for telnet/console session phone_prompt: "123" ; Telnet/Console Prompt phone_password: "123" ; Telnet/Console Password # Enable_VAD (1-enabled, 0-disabled) enable_vad: "0" # Network Media Type (auto, full100, full10, half100, half10) network_media_type: "auto" user_info: phone tftp_cfg_dir: "/" sync: 1 # URL for external Directory location logo_url: http://www.domain.com/asterisk/tux.bmp #directo...

...et_level: "2" ; 0-Disabled (default), 1-Enabled, 2-Privileged # Phone prompt/password for telnet/console session phone_prompt: "Get Out" ; Telnet/Console Prompt phone_password: "a" ; Telnet/Console Password # Enable_VAD (1-enabled, 0-disabled) enable_vad: "0" # Network Media Type (auto, full100, full10, half100, half10) network_media_type: "auto" user_info: "none" # URL for external Directory location #logo_url: "http://10.0.1.3/10-20logo.bmp" ; URL for...

When I make a call using sip, I get the line NOTICE[327696]: File rtp.c, Line 292 (ast_rtp_read): Unknown RTP codec 19 received Repeated many times on the console ; SIP Configuration for Asterisk ; [general] port = 5060 ; Port to bind to ;bindaddr = 0.0.0.0 ; Address to bind to context = outgoing ; Default for incoming calls allow=gsm allow=ulaw

...evel: "2" ; 0-Disabled (default), 1-Enabled, 2-Privileged # Phone prompt/password for telnet/console session phone_prompt: "admin" ; Telnet/Console Prompt phone_password: "admin" ; Telnet/Console Password # Enable_VAD (1-enabled, 0-disabled) enable_vad: "0" # Network Media Type (auto, full100, full10, half100, half10) network_media_type: "auto" user_info: phone tftp_cfg_dir: "/" sync: 1 # URL for external Directory location #logo_url: http://www.domain.com/asterisk/tux.bmp #direct...

...ration [1-3932100 sec] (Default - 3600) timer_register_expires: "3600" # Codec for media stream (g711ulaw (default), g711alaw, g729) preferred_codec: "none" # TOS bits in media stream [0-5] (Default - 5) tos_media: "5" # Enable VAD (0-disable (default), 1-enable) enable_vad: "0" # Allow for the bridge on a 3way call to join remaining parties upon hangup cnf_join_enable: "1" ; 0-Disabled, 1-Enabled (default) # Allow Transfer to be completed while target phone is still ringing semi_attended_transfer: "0" ; 0-Disabled, 1-Enabled (d...

How to disable silence suppression (or Voice activity detection - VAD) on Cisco 7905 phone? On Cisco 7940 I use "enable_vad: 0", but I can't find anything similar for 7905. -- Tomislav Parcina tparcina#lama.hr

I have an *ipphone 7965G* which has to be connected to Asterisk. It has been flashed with SIP firmware but the config file doesn't seem to work maybe I am missing something in it. I appreciate it if you can share your working sample config file with me. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

Anyone have experience with the 3-08-2 release of Cisco's SIP firmware? Are there any new features in the SIPDefault.cnf? Thanks, Ron

Hi Everyone.... Got a newbie type question regarding MOH & Cisco phones. I'm still new to Asterisk (very new in fact) & built up a AsteriskNOW box just to get something going. My simple test system has just 3 Cisco phones a 7905, 7940 & 7960. - Everything's running SIP. The 3 phones can call each other fine. - Can even leave (and retreive) voicemail messages. - No problems.

...piration [1-3932100 sec] (Default - 3600) timer_register_expires: "120" # Codec for media stream (g711ulaw (default), g711alaw, g729) preferred_codec: "none" # TOS bits in media stream [0-5] (Default - 5) tos_media: "5" # Enable VAD (0-disable (default), 1-enable) enable_vad: "0" # Allow for the bridge on a 3way call to join remaining parties upon hangup cnf_join_enable: "1" ; 0-Disabled, 1-Enabled (default) # Allow Transfer to be completed while target phone is still ringing semi_attended_transfer: "0" ; 0-Disabled, 1-Enabled (d...

...sync: 1 ; Default 1 proxy_backup: "" ; Dotted IP of Backup Proxy proxy_backup_port: 5060 ; Backup Proxy port (default is 5060) proxy_emergency: "" ; Dotted IP of Emergency Proxy proxy_emergency_port: 5060 ; Emergency Proxy port (default is 5060) # Configurable VAD option enable_vad: 0 ; VAD setting 0-disable (Default), 1-enable nat_enable: 0 ; 0-Disabled (default), 1-Enabled nat_address: "" ; WAN IP address of NAT box (dotted IP or DNS A record only) voip_control_port: 5060 ; UDP port used for SIP messages (default - 5060) start_media_port: 16384 ; Start RTP ran...

...; ; Dotted IP of Backup Proxy proxy_backup_port: 5060 ; Backup Proxy port (default is 5060) # Emergency Proxy Support proxy_emergency: "" ; Dotted IP of Emergency Proxy proxy_emergency_port: 5060 ; Emergency Proxy port (default is 5060) # Configurable VAD option enable_vad: 0 ; VAD setting 0-disable (Default), 1-enable ####### New Parameters added in Release 2.2 ###### # NAT/Firewall Traversal nat_enable: 0 ; 0-Disabled (default), 1-Enabled nat_address: "" ; WAN IP address of NAT box (dotted IP or DNS A record...

Hi everyone, I've been toying with * for quite some time now. I've got two Cisco 7940's with the SIP firmware playing nice with *. I can also make outbound calls via IAXTel (toll-free calls only) and all other calls I have routed out my X100P-clone adapter. Here's my question... Is there a way to capture the inbound callerid from my phone line (coming in on the X100P) and have

Thanks But how about the config files (SIP...) that needs to be inside the tftp server, where can I get a sample of that? That's where the images for the firmwares of the ip phones come from, on boot right? -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Alen Salamun Sent: Wednesday, January 26, 2005 5:47