Hi Edwin! (and everybody)
I have some questions about SIP, as I wrote in another mail. I have a SIP
Gateway and I have two phones conected to it.Also, I have two Dlink
dg102s with four phones conected to them. The main problems are two.
Calls between the phones conected to the SIP GW and the ones conected
to the MGCP GW goes OK ONLY if I call from the MGCP to the SIP. Phones
at MGCP can call without problems to the PSTN (voice quality isn't very
good, with silence times, but it can be supported!). But phones at SIP can't
do any call! The problem is that when I pick up the callee phone, I don't
hear nothing and the call goes off inbetween 4 or 5 seconds. And the
caller (SIP) doesn't realise I have picked up, because It's still hearing the
calling tone.When the call goes off, the caller hear the congestion tone. I
don't know what is the problem!!!!
I can't achive to transfer calls. When I dial #, it doesn't happen anything!!
And the callerID doesn't work either.
My sip.conf:
[general]
port = 5060
bindaddr = 0.0.0.0
context = default
transfer = yes
threewaycalling = yes
usecallerid = yes
hidecallerid = no
[sip]
type=friend
callerid="sip" <2222>
username=sip
host=188.208.12.37
accountcode=sip
My extensions.conf
exten => 2222,1,dial,SIP/[EMAIL PROTECTED]|60|rTt
exten => 2222,2,Hangup
Thanks very much for any help!!!
Bye
Michelle
>Nat=1 is so that mgcp functions properly behind a NAT gateway.
>What kind of problems are you having with your SIP? What type of SIP
>phone do you have? Can you elaborate a little more or even post you
>SIP.conf? >Here's what ours looks like so you can do a comparison:
>Sip.conf >----------- >; >; SIP Configuration for Asterisk >;
>[general] >port = 5060 ; Port to bind to >bindaddr = 0.0.0.0 ;
Address to bind to >context = sipstart ; Default for incoming calls
>tos = lowdelay >[sip_phone] >type=friend
>username=sip_phone >secret=sip_phone >host=dynamic
>nat=1 >-----Original Message----- >From: href="javascript:sendMsg
('asterisk-users-
asterisk-users-
[EMAIL PROTECTED]');">[EMAIL PROTECTED]');">asterisk-users-
[EMAIL PROTECTED]
>[mailto:asterisk-users-
[EMAIL PROTECTED]');">[EMAIL PROTECTED]');">[mailto:asterisk-
[EMAIL PROTECTED]
On Behalf Of michelle >matis litio >Sent: Wednesday, June 11,
2003 12:12 PM >To: asterisk-
[EMAIL PROTECTED]');">[EMAIL PROTECTED]');">asterisk-
[EMAIL PROTECTED]
>Subject: [Asterisk-Users] Re:Some SIP questions AGAIN >Hi Edwin
>I have my mgcp.conf almost the same as yours, except from "nat=1" ,
why >do you put it? >Anyway, DL102s now works more or less
acceptably so now I'm having a >battle with sip.conf !!!! >Thank you
for your help >Michelle >----- >Tu cuenta de correo gratuita Mixmail
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