[Asterisk-Users] some sip questions AGAIN

[michelle matis litio]

I write the email again, the third time!!, cause the other two ones, I have 
had problems while sending them. I hope this time it works. Here is the 
email again:

Hi (and sorry) everybody

I'm starting with SIP and I wanted to ask some questions, perhaps silly 
ones, but I hope people can answer me! 

1) Which codecs may I use? I want the SIP phones to call to the PSTN 
above all, but I have two dlink dg102s (MGCP) and I'd like to can call them 
too. The problem is that when I use g723 I can call MGCP but I can't call 
PSTN (call goes off when I pick the phone up). What can I do?

2)What is  [EMAIL PROTECTED] ? For what is used?

3)Can I transfer calls? I guess that if I do transfer = yes in the general 
section of sip.conf, it should work... but it doesn't!!

4)And finally, the caller ID. I have done usecallerid=yes in the general 
section of sip.conf and the I put callerid="SIP" <2222> in the [sip] section 
(the one that I have created for my devide). But it doesn't work either! Any 
ideas?

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

Thanks you all!!!

Michelle

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