Hi Wagner!
Thanks for the interest. I'm from Colombia and I'm trying to develop VoIP as you know on *. So thanks again for the offering in Brazil, althought you can help me with some idea by this way.
To make the call I'm using SJphone (softphones) to make the tests. I'm not using IP phones because we don't have a lot of investment as a said before.
This is my [general]sip.conf format:
I omitted other parts which were on comments because it is example from the web site
[general]
context=default
;allowguest=no
;realm=mydomain.tld
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
;domain=mydomain.tld
;************** Cambio de lineas
disallow=all
;allow=g729
allow=gsm
allow=ulaw
jitterbuffer=yes
maxjitterbuffer=800
;allow=ilbc
;musicclass=default
;language=en
;relaxdtmf=yes
rtptimeout=60
;rtpholdtimeout=300
;trustrpid = no
;sendrpid = yes
;progressinband=never
;useragent=Asterisk PBX
;promiscredir = no
;usereqphone = no
;*********** Cambio de lineas DTMFMODE estaba en comentarios ********************
dtmfmode = rfc2833
;compactheaders = yes
;sipdebug = yes
;subscribecontext = default
;notifyringing = yes
;******************** Usuario 1 ************************
[usuario1]
type=friend
host=dynamic
dtmfmode=rfc2833
username=usuario1
secret=usuario1
;******************** Usuario 2 ************************
[usuario2]
type=friend
host=dynamic
dtmfmode=rfc2833
username=usuario2
secret=usuario2
Thanks again for the interest and if you have and idea I would apreciate a lot!
Carlos Bernat
2006/7/27, Wagner Nunes <
[EMAIL PROTECTED]>:
Hi Carlos!!!Let me ask one thing... ... r u brazilian??? Becouse I work with * projects and if u r in brazil maybe i can help u.But about your problem, What are u using to call thru *? IP Phone, softphone? What is your sip.conf settings?
Carlos Alberto Bernat Orozco <[EMAIL PROTECTED] > escreveu:Hi Group!
Still I'm concern about my problem with echo on the voice and I want to ask some advice to developing VoIP. Maybe I'm very ambiciuos or maybe not because I want to give VoIP to near 500 users.
We got an small ISP and we have the project to give telephony (for now) to our users between them. Our resources are limited and I have installed * as a hope to give this service to our users. We have a good network (with small problems) but I believed that is possible to give this service. Our HFC network is very well calibrated and works fine. The users have cable modems to connect to the internet and we give private adresses to some users.
I'm searching for someone who has the same problem in the past with similar things, to know how solve it and if is possible to give VoIP calls with a server with a public address and the softphones (for the costs) with extensions registered on our * box. I configured * four months ago and between two extensions and works very well and but later I did the same test on this week and unfortunaly the voice goes out with echo. So I have the feeling that maybe there's something wrong with the codecs and wich codecs do I need to give the service.
Thanks for any help you can give me
Carlos Bernat_______________________________________________
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