24 Jul
2009
24 Jul
'09
4:33 p.m.
Ok here it goes,
I have a Mera Server that I'm trying to connect my Kamailio to change my old
SER server to a new level for my Retail clients, my ser will not route to
any router my Mera Server will do it
What I'm trying to make is a SIP:xxx.xxx.xxx.xxx:5060 on my mera server.
When I register it goes ok
U 200.196.28.25:63042 -> 200.196.28.124:5060
REGISTER sip:sps0.matrix.net.br SIP/2.0..Via: SIP/2.0/UDP
192.168.5.38:63042;branch=z9hG4bK-d8754z-2209c854c438c347-1---d8754z-;rport.
.Max-Forwards: 70..Contact: <sip:rafael
@192.168.5.38:63042;rinstance=bcdb08a4009ee386>..To:
"SPS0"<sip:rafael@sps0.matrix.net.br>..From:
"SPS0"<sip:rafael@sps0.matrix.net.br>;tag=43659358..Call-ID:
ZWVkMTRiNjQ1ZT
VkZTNlNmMyOGE5M2MzOTM2NTM4YmY...CSeq: 2 REGISTER..Expires: 3600..Allow:
INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE,
INFO..User-Agent: X-Lite releas
e 1103d stamp 53117..Authorization: Digest
username="rafael",realm="sps0.matrix.net.br",nonce="4a68f31e000000008e5e591d
36c39beb4f5939faad6d9d29",uri="sip:sps0.matrix.net.br"
,response="386a15e662550d45f24bccb6cbb25494",algorithm=MD5..Content-Length:
0....
U 200.196.28.124:5060 -> 200.196.28.25:63042
SIP/2.0 200 OK..Via: SIP/2.0/UDP
192.168.5.38:63042;branch=z9hG4bK-d8754z-2209c854c438c347-1---d8754z-;rport=
63042;received=200.196.28.25..To: "SPS0"<sip:rafael@sps0.matrix.
net.br>;tag=f8f2ab2c1295e90ed7dbb499b30f44b2.d55b..From:
"SPS0"<sip:rafael@sps0.matrix.net.br>;tag=43659358..Call-ID:
ZWVkMTRiNjQ1ZTVkZTNlNmMyOGE5M2MzOTM2NTM4YmY...CSeq: 2 R
EGISTER..Contact:
<sip:rafael@192.168.5.38:63042;rinstance=bcdb08a4009ee386>;expires=3600..Ser
ver: Kamailio (1.5.1-notls (i386/linux))..Content-Length: 0....
But when I try to make calls
U 200.196.28.25:63042 -> 200.196.28.124:5060
INVITE sip:551133239212@sps0.matrix.net.br SIP/2.0..Via: SIP/2.0/UDP
192.168.5.38:63042;branch=z9hG4bK-d8754z-a957a2771c264219-1---d8754z-;rport.
.Max-Forwards: 70..Contact:
<sip:rafael@200.196.28.25:63042>..To:
"551133239212"<sip:551133239212@sps0.matrix.net.br>..From:
"SPS0"<sip:rafael@sps0.matrix.net.br>;tag=b45cd830..Call-ID:
NzNiNzc0YzhkNTJ
mYWE1ZWNmYzcwZGNlMzQ3NDc5ZDU...CSeq: 1 INVITE..Allow: INVITE, ACK, CANCEL,
OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO..Content-Type:
application/sdp..User-Agent:
X-Lite release 1103d stamp 53117..Content-Length: 361....v=0..o=- 7 2 IN IP4
192.168.5.38..s=CounterPath X-Lite 3.0..c=IN IP4 192.168.5.38..t=0
0..m=audio 8934 RTP/AVP 107 0
8 101..a=alt:1 3 : b2cv/jrI CAiY+XUC 192.168.5.38 8934..a=alt:2 2 : oUZIn8S0
Uq5tdU9E 192.168.52.1 8934..a=alt:3 1 : Z8/DE0ry qIvXPCKF 192.168.228.1
8934..a=fmtp:101 0-15..
a=rtpmap:107 BV32/16000..a=rtpmap:101 telephone-event/8000..a=sendrecv..
U 200.196.28.124:5060 -> 200.196.28.25:63042
SIP/2.0 407 Proxy Authentication Required..Via: SIP/2.0/UDP
192.168.5.38:63042;branch=z9hG4bK-d8754z-a957a2771c264219-1---d8754z-;rport=
63042;received=200.196.28.25..To: "55
1133239212"<sip:551133239212@sps0.matrix.net.br>;tag=f8f2ab2c1295e90ed7dbb49
9b30f44b2.e355..From:
"SPS0"<sip:rafael@sps0.matrix.net.br>;tag=b45cd830..Call-ID:
NzNiNzc0YzhkNT
JmYWE1ZWNmYzcwZGNlMzQ3NDc5ZDU...CSeq: 1 INVITE..Proxy-Authenticate: Digest
realm="sps0.matrix.net.br",
nonce="4a68f32800000002a304ed8610abc47da56bdc29957a394d"..Server: Kama
ilio (1.5.1-notls (i386/linux))..Content-Length: 0....
U 200.196.28.25:63042 -> 200.196.28.124:5060
ACK sip:551133239212@sps0.matrix.net.br SIP/2.0..Via: SIP/2.0/UDP
192.168.5.38:63042;branch=z9hG4bK-d8754z-a957a2771c264219-1---d8754z-;rport.
.To: "551133239212"<sip:5511332
39212 [!at] sps0.matrix.net.br (replace the [!at] with a
@)>;tag=f8f2ab2c1295e90ed7dbb499b30f44b2.e355..From:
"SPS0"<sip:rafael@sps0.matrix.net.br>;tag=b45cd830..Call-ID:
NzNiNzc0YzhkNTJmYWE1ZWNmYzcwZGNlMzQ3N
Dc5ZDU...CSeq: 1 ACK..Content-Length: 0....
U 200.196.28.25:63042 -> 200.196.28.124:5060
INVITE sip:551133239212@sps0.matrix.net.br SIP/2.0..Via: SIP/2.0/UDP
192.168.5.38:63042;branch=z9hG4bK-d8754z-267c082e8463a77a-1---d8754z-;rport.
.Max-Forwards: 70..Contact:
<sip:rafael@200.196.28.25:63042>..To:
"551133239212"<sip:551133239212@sps0.matrix.net.br>..From:
"SPS0"<sip:rafael@sps0.matrix.net.br>;tag=b45cd830..Call-ID:
NzNiNzc0YzhkNTJ
mYWE1ZWNmYzcwZGNlMzQ3NDc5ZDU...CSeq: 2 INVITE..Allow: INVITE, ACK, CANCEL,
OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO..Content-Type:
application/sdp..Proxy-Author
ization: Digest
username="rafael",realm="sps0.matrix.net.br",nonce="4a68f32800000002a304ed86
10abc47da56bdc29957a394d",uri="sip:551133239212@sps0.matrix.net.br",response
="bd0
2426b09957f78a347036f020e4c48",algorithm=MD5..User-Agent: X-Lite release
1103d stamp 53117..Content-Length: 361....v=0..o=- 7 2 IN IP4
192.168.5.38..s=CounterPath X-Lite 3.0
..c=IN IP4 192.168.5.38..t=0 0..m=audio 8934 RTP/AVP 107 0 8 101..a=alt:1 3
: b2cv/jrI CAiY+XUC 192.168.5.38 8934..a=alt:2 2 : oUZIn8S0 Uq5tdU9E
192.168.52.1 8934..a=alt:3 1
: Z8/DE0ry qIvXPCKF 192.168.228.1 8934..a=fmtp:101 0-15..a=rtpmap:107
BV32/16000..a=rtpmap:101 telephone-event/8000..a=sendrecv..
U 200.196.28.124:5060 -> 200.196.28.25:63042
SIP/2.0 404 Not Found..Via: SIP/2.0/UDP
192.168.5.38:63042;branch=z9hG4bK-d8754z-267c082e8463a77a-1---d8754z-;rport=
63042;received=200.196.28.25..To: "551133239212"<sip:5511
33239212 [!at] sps0.matrix.net.br (replace the [!at] with a
@)>;tag=bbd8f31c58bc6a3266ec5509a254d55b-2513..From:
"SPS0"<sip:rafael@sps0.matrix.net.br>;tag=b45cd830..Call-ID:
NzNiNzc0YzhkNTJmYWE1ZWNmYzcwZGNlMz
Q3NDc5ZDU...CSeq: 2 INVITE..Server: Kamailio (1.5.1-notls
(i386/linux))..Content-Length: 0....
Now may someone help me :D
Thanks guys!
-------------------------------
Message: 1
Date: Fri, 24 Jul 2009 10:56:34 +0200
From: I?aki Baz Castillo <ibc(a)aliax.net>
Subject: Re: [Kamailio-Users] Can't make calls
To: users(a)lists.kamailio.org
Message-ID: <200907241056.34301.ibc(a)aliax.net>
Content-Type: Text/Plain; charset="utf-8"
El Viernes, 24 de Julio de 2009, Rafael Moreno Gomes escribi?:
Can Someone help me?
No.
First: Don't send the same mail twice.
Second: Describe your problem in a detailed way (nobody will spend his time
trying to guess which problem you have by inspecting your full configuration
and SIP traces).
--
I?aki Baz Castillo <ibc(a)aliax.net>
------------------------------
Message: 2
Date: Fri, 24 Jul 2009 11:43:01 +0200
From: Daniel-Constantin Mierla <miconda(a)gmail.com>
Subject: Re: [Kamailio-Users] Can't make calls
To: I?aki Baz Castillo <ibc(a)aliax.net>
Cc: users(a)lists.kamailio.org
Message-ID: <4A698225.7080309(a)gmail.com>
Content-Type: text/plain; charset=UTF-8; format=flowed
On 24.07.2009 10:56 Uhr, I?aki Baz Castillo wrote:
> El Viernes, 24 de Julio de 2009, Rafael Moreno Gomes escribi?:
>
> Can Someone help me?
>>
>
> No.
>
> First: Don't send the same mail twice.
>
> Second: Describe your problem in a detailed way (nobody will spend his
time
trying to guess which problem you have by inspecting
your full
configuration
and SIP traces).
I tried to put together some info about reporting issues. It can be used
to help new people in such situation:
http://www.kamailio.org/dokuwiki/doku.php/troubleshooting:report-an-issue
Please add content there, so reporting and troubleshooting becomes
easier for everybody.
Cheers,
Daniel
--
Daniel-Constantin Mierla
http://www.asipto.com/index.php/sip-router-bootcamp/
------------------------------
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End of Users Digest, Vol 50, Issue 49
*************************************
24 Jul
24 Jul
4:43 p.m.
On Friday 24 July 2009 15:33:10 Rafael Moreno Gomes wrote:
U 200.196.28.124:5060 -> 200.196.28.25:63042
SIP/2.0 404 Not Found..Via: SIP/2.0/UDP
Just "User not found" ... what's wrong with this ?
Now may someone help me :D
Not with just a SIP trace, you need to explain a little your enviroment and
what are your goals.
--
Raúl Alexis Betancor Santana
Dimensión Virtual
27 Jul
27 Jul
3:45 p.m.
New subject: [Kamailio-Users] no inbound sound even with RTPproxy
Hello
I have Kamailio with Nathelper and RTPproxy running this way:
rtpproxy -l xxx.xxx.xxx.xxx -u mysql -s udp:localhost:7722
I think RTPproxy is working fine with NAThelper, because Kamailio logs say:
"RTPproxy found - support enabled"
X-Lite and SNOM phones behind NAT work fine.
But when I try to connect with an Asterisk behind NAT, the sound doesnt
pass through the Asterisk client (the Asterisk client doesn't receive
sound).
I already tried in 2 different NATs, with no firewalls.
Kamailio logs show no errors.
This is my Asterisk config:
[kamailio]
type=peer
host=xxx.xxx.xxx.xxx
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=g726
qualify=1000
username=my_username
fromuser=my_username
secret=password
sip*CLI> sip show peers
Name/username Host Dyn Nat ACL Port Status
kamailio/my_username xxx.xxx.xxx.xxx 5060 OK
(890 ms)
Im not sure if the problem is in Kamailio or Asterisk config.
How can I be sure?
How can I debug the RTP stream in Kamailio and RTPproxy?
Thanks
Regards
Joao Pereira
--
StarTel - A Rede Livre
Joao Gomes Pereira
www.startel.pt
+351 304500650
sip: gomespereira(a)startel.pt
3:51 p.m.
New subject: [Kamailio-Users] no inbound sound even with RTPproxy
2009/7/27 Joao Gomes Pereira <gomespereira(a)startel.pt>pt>:
Im not sure if the problem is in Kamailio or Asterisk
config.
How can I be sure?
How can I debug the RTP stream in Kamailio and RTPproxy?
Use ngrep to inspect the SIP messages and check the address in the SDP.
--
Iñaki Baz Castillo
<ibc(a)aliax.net>
5:03 p.m.
New subject: [Kamailio-Users] no inbound sound even with RTPproxy
You can also check if RTP is ariving to Asterisk by doing 'rtp set
debug on' in the CLI.
--
Saúl -- "Nunca subestimes el ancho de banda de un camión lleno de disketes."
----------------------------------------------------------------
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5 comments
6 participants
participants (6)
-
Iñaki Baz Castillo -
Joao Gomes Pereira -
Rafael Moreno Gomes -
Raúl Alexis Betancor Santana -
Ricardo Martins -
Saúl Ibarra