search for: amejia

...onf has "disallow=all & allow=g729" The problem: [SIPphone] [sip.conf] [h323.conf] [H323gw] g729 ---> allow=g729 ---> allow=g729 ---> g729 When I dial to the gateway from the SIPphone using g729 as my sip phone's default codec I get: -- Executing Dial("SIP/amejia-8be1", "H323/######@H323gw") in new stack Apr 19 15:02:14 WARNING[68595]: channel.c:2504 ast_request: No translator path exists for channel type H323 (native 4) to 256 Apr 19 15:02:14 NOTICE[68595]: app_dial.c:1010 dial_exec_full: Unable to create channel of type 'H323' (caus...

Hello everyone. I'm using Asterisk 1.4.31 and A2Billing 1.7.0 to manage a small wholesale operation, so I configured A2Billing for not to answer the call nor play any greetings or balance notifications to the caller. I'm authenticating each customer by it's IP address, and each customer has it's own context, in which I set the following: ;=====in extensions.conf======

I'm running: OS: FreeBSD 6.0 Asterisk: 1.2.4 Installing OH323: 0.7.3 I have this error when compiling chan_oh323.c: In function `reload_config': chan_oh323.c:4677: warning: implicit declaration of function `sscanf' chan_oh323.c: At top level: chan_oh323.c:3244: warning: 'update_call_ids' defined but not used gcc -shared -Xlinker -x -g -o chan_oh323.so chan_oh323.o

Hello, I am trying to port a package that was built for R 2.15 over to R 3.0. This package has an Initialize method that uses dyn.load to load a dll that was built separately, and then uses .C() to make calls on the functions in that dll. This worked fine in 2.15, however, I see that now for 3.0 .C() will only search in the namespace of your current package. This seems to make it impossible to use

Hello everyone. I'm using Asterisk 1.4.31 and A2Billing 1.7.0 to manage a small wholesale operation, so I configured A2Billing for not to answer the call nor play any greetings or balance notifications to the caller. I'm authenticating each customer by it's IP address, and each customer has it's own context, in which I set the following: ;=====in extensions.conf======

Hi everybody: I configured my Asterisk to register to my VoIP provider, and I can make outgoing calls, but I can't receive any calls with it. I used Ethereal to sniff the activity of it, and I found something that might be causing the problem: When my provider's gateway does the "Request: INVITE mynumber@my-voip-provider.tld ..." my Asterisk asks for "Status: 407 Proxy