asterisk users - Jun 2003 - The same SIP problems...SORRY!

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@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@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@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@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@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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On Mon, 16 Jun 2003, michelle matis litio wrote:
> 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@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
Hi,

Its being sent to that IP address, because that is that the
originating SIP device put in its Via header.

Also, your SIP device didn't put any From or To in its INVITE.

Perhaps you could send a sip debug from the start of a SIP call
attempt.

But I'm sure that the trouble is with your SIP Gateway device's
setup.

Steve

Hi!
I thought it was the SIP device too, but I have looked for avery litle comand 
of this device and I can't find this Ip address, and I see that its Ip is
Ok,
and I have configurated the REGISTRAR section too... I don't know what's
happening, and I don't understand that, if the IP is wrong, why can I hear 
the callee phone ringing and the call only goes off when I pick it up?

it's so strange...I think!

Michelle




>On Mon, 16 Jun 2003, michelle matis litio wrote: >> 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 &gt;&gt; To: ;tag=as52ed0a6a &gt;&gt; Call-ID: <A
href="javascript:sendMsg('f93b00-0-13c4-3a5246f7-8c6b602-
3765@188.208.12.37');">f93b00-0-13c4-3a5246f7-8c6b602-
3765@188.208.12.37</A> &gt;&gt; CSeq: 1 INVITE &gt;&gt;
User-Agent:
Asterisk PBX &gt;&gt; Contact: &gt;&gt; Content-Type:
application/sdp
&gt;&gt; Content-Length: 135 &gt;&gt; &gt;&gt; v=0
&gt;&gt; o=root 11673
11673 IN IP4 188.208.12.237 &gt;&gt; s=session &gt;&gt; c=IN IP4
188.208.12.237 &gt;&gt; t=0 0 &gt;&gt; =audio 13532 RTP/AVP 0
&gt;&gt;
a=rtpmap:0 PCMU/8000 &gt;Hi, &gt;Its being sent to that IP address, 
because that is that the &gt;originating SIP device put in its Via header. 
&gt;Also, your SIP device didn't put any From or To in its INVITE. 
&gt;Perhaps you could send a sip debug from the start of a SIP call 
&gt;attempt. &gt;But I'm sure that the trouble is with your SIP
Gateway
device's &gt;setup. &gt;Steve 
&gt;_______________________________________________ &gt;Asterisk-
Users mailing list &gt;<A href="javascript:sendMsg('Asterisk-
Users@lists.digium.com>http://lists.digium.com/mailman/listinfo/asterisk-
users');">Asterisk-Users@lists.digium.com 
&gt;http://lists.digium.com/mailman/listinfo/asterisk-users</A>

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