[Asterisk-Users] Re:Some SIP questions AGAIN

[michelle matis litio]

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/2222 at 188.208.12.37|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-
admin at lists.digium.com');">admin at lists.digium.com');">asterisk-users-
admin at lists.digium.com 
>[mailto:asterisk-users-
admin at lists.digium.com]');">admin at lists.digium.com]');">[mailto:asterisk-
users-admin at lists.digium.com]
On Behalf Of michelle >matis litio >Sent: Wednesday, June 11, 
2003 12:12 PM >To: asterisk-
users at lists.digium.com');">users at lists.digium.com');">asterisk-
users at lists.digium.com 
>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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