[Asterisk-Users] some sip questions AGAIN

[Edwin A. Silva]

Hi Michelle,

For the d-link VoIP gateways you need to configure your mgcp.conf the
following is my configuration for a dg104s.

Mgcp.conf
--------------
;
; MGCP Configuration for Asterisk
;
[general]
port = 2427
bindaddr = 0.0.0.0

[mgcp01]
threewaycalling=yes
transfer=yes
callwaiting=yes
callwaitingcallerid=yes
host = dynamic
context = sipstart
callerid = Edwin Silva <6014>
line => aaln/4
callerid = Edwin Silva <6013>
line => aaln/3
callerid = Edwin Silva <6012>
line => aaln/2
callerid = Edwin Silva <6011>
line => aaln/1
nat=1

And the following is my entry in extensions.conf

Extensions.conf
-----------------
;MGCP Phones (DG-104S)
exten => _60XX,1,Dial,MGCP/aaln/${EXTEN:3}@mgcp0${EXTEN:2:1}|20
exten => _60XX,2,Congestion
exten => _60XX,102,Congestion


This should let you call any device on the dg102s without any problems
just make sure that you configure your dg-104s to send out mgcp01 or
whatever you decide to use as the entry in your mgcp.conf and change the
pattern matching in extensions.conf to reflect this as well.


Edwin Silva
WW Works Inc.
3060 Mainway Dr. Unit 104
Burlington, ON
L7M 1A3
(905) 332-5844 ext. 517

-----Original Message-----
From: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com] On Behalf Of michelle
matis litio
Sent: Wednesday, June 11, 2003 3:37 AM
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] some sip questions AGAIN



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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