[Asterisk-Users] Re:Some SIP questions AGAIN
michelle matis litio
michelleuser at mixmail.com
Thu Jun 12 03:12:01 MST 2003
Hi everybody one more time!
I also have done a SIP debug and that's an extract of what I have found:
(...)
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: <sip:sip at 188.208.12.37:5060> ;tag=0-13c4-3a5246f7-8c6b604-c3a
To: <sip:3333 at 188.208.12.237>;tag=as52ed0a6a
Call-ID: f93b00-0-13c4-3a5246f7-8c6b602-3765 at 188.208.12.37
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Contact: <sip:3333 at 188.208.12.237>
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 <sip:sip at 188.208.12.37:5060> 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: <sip:3333 at 188.208.12.237>;tag=as52ed0a6a
To: <sip:sip at 188.208.12.37:5060> ;tag=0-13c4-3a5246f7-8c6b604-c3a
Contact: <sip:3333 at 188.208.12.237>
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: <sip:3333 at 188.208.12.237>
To: <sip:sip at 188.208.12.37:5060> ;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
I can't understand why the "out of SIP" messages go to an IP so strange!!!
(229...)
Any ideas?
I've just sent my sip.conf and all in the previous message. Hope someone
can help!!
greetings
michelle
PD:188.208.12.237 is the asterisk IP
>>Michelle wrote:
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
;-----Original Message-----
>From: <A href="javascript:sendMsg('asterisk-users-
admin at lists.digium.com');">asterisk-users-admin at lists.digium.com</A>
><A href="javascript:sendMsg('[mailto:asterisk-users-
admin at lists.digium.com]');">[mailto:asterisk-users-admin at lists.digium.com]
</A> On Behalf Of michelle >matis litio >Sent: Wednesday, June 11,
2003 12:12 PM >To: <A href="javascript:sendMsg('asterisk-
users at lists.digium.com');">asterisk-users at lists.digium.com</A>
>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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