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...my voice to the person I've called (extension), after doing this tests below I always got this error "error: this FQDN or IP is not valid: voicegw". This could cause problems (namely audio problems)? Best regards, Helder voicegw:~# sipsak -C empty -a password -s sip:505@voicegw.microrede.com error: this FQDN or IP is not valid: voicegw voicegw:~# sipsak -C empty -a password -s sip:505@192.168.1.50 error: this FQDN or IP is not valid: voicegw -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachmen...

Hi! Are you also getting in trouble while trying to register in Broadvoice? Cumprimentos / Best regards, Helder Rog?rio __________________________________________ Microrede - Tecnologias de Informa??o, Ltd. http://www.microrede.pt *** ? There are only two types of people in the world, those who have lost data and those who will. ? -- Richard Nixon

...he authentication to a IAX provider and "transfer" the call to it, avoiding using my own bandwidth? I've tested it with SER with some results, I was wondering if it is possible with Asterisk. Cumprimentos / Best regards, Helder Rog?rio __________________________________________ Microrede - Tecnologias de Informa??o, Ltd. http://www.microrede.pt Sede / Headoffice Rua S. da Gl?ria, 66 1170-353 Lisbon Portugal Tel. 21 887 13 21 Fax. 21 8127158 *** Filial / Branch Office Rua Lopes, 55 - C/V E 1900-297 Lisbon Portugal Tel. 21 814 83 72 Web: http://www.microrede.pt *** ? There...

Hi, I've been loosing my mind with NAT and read that IAX doesn't have problems about nat. Does anyone knows about hadware (routers and etc) support IAX? Best regards helder -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041230/f689bdde/attachment.htm

...t; 200,3,Playback(demo-echodone) ; demonstra??o de qualidade exten => 200,4,Hangup ; demonstra??o de qualidade exten => 210887677,1,macro(stdexten,SIP/210887677) ; daniel - teste exten => 502,1,macro(stdexten,SIP/502) ; Jo?o Portatil exten => 505,1,macro(stdexten,SIP/505) ; Central PBX Microrede exten => 506,1,macro(stdexten,SIP/506) ; Semortrade Router exten => 510,1,macro(stdexten,SIP/510) ; PC Tiago escritorio This is generated by the perl scripts on the addons. Cumprimentos / Best regards, Helder Rog?rio

Hello, Is there an utility for asterisk for codec conversion? I tried google but i haven' got anything. I am trying to initiate a call with G711 codec to asterisk and i would like asterisk to call a gateway with an g729 codec, therefore making a codec conversion from g711 to g729. I know chan_oh323 does it by specifying the OUT_CODEC variable, but chan_h323 does not. And i was wondering is

Hi, This is a thank you message for all that helped me including Max from www.asterisk-support.ru with whishes of a Happy New Year. Althought I still have a problem I'm happier I've 50% of my task complete. I'm using two TA from Draytek (router 2600V / router 2500V) 3 ADSL lines (2 for TA 1 for ASTERISK with a Draytek 2500 no voice model). If I call from one to another (using

I'm sorry as Patrick said I've made a post 4 times but didn't realize it until I've started received the messages back. The mail client hung (apparently) but it was in fact sending the msgs. I know probably some/all of you (and the owners of the list at Digium) were not very pleased for receiving such a bunch of copies of the same email, however it was sent in error and apologies

Hi, Is there some parameter that I should pay attention to when using externip parameter on sip.conf? I ask this because after using sipsak I've noticed maybe the reason that i don't get voice in one direction could be because at the SDP/SIP messages are references to the 192.168.1.50 (server internal ip) and not the ip from the provider. Is there a way to alter SDP messages to correct

Hi! For those who use Xonox Asterisk distribution based on Debian, I've installed it and got a strange situation. the server used to hang at 6:25am every day. I've found there is a cron job task that hangs the server while executing. Disabling will solve the problem. best regards, Helder -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi! There's any way to set up a call using G726 (sip peer) receive it on Asterisk convert it to G711Mu to send it to PSTN broadband termination? I've put the following in sip.conf: disalow=all allow=gsm allow=g726 (my TAs use G726 32K) best regards, Helder -------------- next part -------------- An HTML attachment was scrubbed... URL: