[Asterisk-Users] The same SIP problems...SORRY!

[michelle matis litio]

Hi eveybody again!

I don't want to be annoying, but if nobody can help me with this, I'll have to 
desist of working with SIP.I have some questions about SIP, as I wrote in 
another mail. I have a SIP Gateway and I have two phones (an analog one 
and a DECT one) 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!!!! 

The other problem is that 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


I also have done a SIP debug and I'm sneding an extract of what I have 
found. I can't understand why the "out of SIP" messages go to an IP so 
strange!!! (229...) I can't find this IP anywhre in my system...Any ideas? 
Hope someone can help!!
Thanks in advance!
michelle
PD:188.208.12.237 is the asterisk IP

(...) 
s=session
c=IN IP4 188.208.12.237
t=0 0
=audio 13532 RTP/AVP 0
a=rtpmap:0 PCMU/8000

to 229.159.241.112:5060
Retransmitting #5 (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 229.159.241.112:5060 ;branch=z9hG4bK-3a5246f7-
8c6b606-10eb
From: ;tag=0-13c4-3a5246f7-8c6b604-c3a
To: ;tag=as52ed0a6a
Call-ID: f93b00-0-13c4-3a5246f7-8c6b602-3765 at 188.208.12.37
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Contact: 
Content-Type: application/sdp
Content-Length: 135

v=0
o=root 11673 11673 IN IP4 188.208.12.237
s=session
c=IN IP4 188.208.12.237
t=0 0
=audio 13532 RTP/AVP 0
a=rtpmap:0 PCMU/8000

to 229.159.241.112:5060
-- Hungup 'IAX2[test]/1'
== Spawn extension (default, 3333, 1) exited non-zero 
on 'SIP/229.159.241.112:5
060'
set_destination: Parsing for address/port to 
send t
o
set_destination: set destination to 188.208.12.37, port 5060
Reliably Transmitting:
BYE sip:sip at 188.208.12.37:5060 SIP/2.0
Via: SIP/2.0/UDP 188.208.12.237:5060;branch=z9hG4bK6723148d
From: ;tag=as52ed0a6a
To: ;tag=0-13c4-3a5246f7-8c6b604-c3a
Contact: 
Call-ID: f93b00-0-13c4-3a5246f7-8c6b602-3765 at 188.208.12.37
CSeq: 102 BYE
User-Agent: Asterisk PBX
Content-Length: 0

(no NAT) to 188.208.12.37:5060
Sip read:
SIP/2.0 200 OK
From: 
To: ;tag=0-13c4-3a5246f7-8c6b604-c3a
Call-ID: f93b00-0-13c4-3a5246f7-8c6b602-3765 at 188.208.12.37
CSeq: 102 BYE
Via: SIP/2.0/UDP 
188.208.12.237:5060 ;received=188.208.12.237 ;branch=z9hG4bK67231
48d
Content-Length:0


7 headers, 0 lines
Message is BYE



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