18 Feb
2013
18 Feb
'13
3:58 p.m.
Hi,
Does someone know/uses a simple rule so that Kamailio only exchanges
traffic with machines in the dispatcher?
Best,
--
Benjamin Henrion <bhenrion at ffii.org>
FFII Brussels - +32-484-566109 - +32-2-3500762
"In July 2005, after several failed attempts to legalise software
patents in Europe, the patent establishment changed its strategy.
Instead of explicitly seeking to sanction the patentability of
software, they are now seeking to create a central European patent
court, which would establish and enforce patentability rules in their
favor, without any possibility of correction by competing courts or
democratically elected legislators."
18 Feb
18 Feb
4:03 p.m.
I use something like this on my boxes to keep them foreigners out:
if ( !ds_is_from_list() ) {
sl_send_reply("403","Your not in my dispatcher list");
}
Put this high up your config.
Cheers
On 18 Feb 2013, at 11:58, Benjamin Henrion wrote:
Hi,
Does someone know/uses a simple rule so that Kamailio only exchanges
traffic with machines in the dispatcher?
Best,
--
Benjamin Henrion <bhenrion at ffii.org>
FFII Brussels - +32-484-566109 - +32-2-3500762
"In July 2005, after several failed attempts to legalise software
patents in Europe, the patent establishment changed its strategy.
Instead of explicitly seeking to sanction the patentability of
software, they are now seeking to create a central European patent
court, which would establish and enforce patentability rules in their
favor, without any possibility of correction by competing courts or
democratically elected legislators."
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
4:16 p.m.
New subject: Kamailio only allowing traffic with machines in the dispatcher.cfg?
Call ds_is_from_list([groupid]) when you receive a request. If it
returns true the request came from one of the members of the group (and
you can proceed), if it returns false you can reject it and drop the
request.
Regards,
Peter
On 18/02/13 10:58, Benjamin Henrion wrote:
Hi,
Does someone know/uses a simple rule so that Kamailio only exchanges
traffic with machines in the dispatcher?
Best,
--
Benjamin Henrion <bhenrion at ffii.org>
FFII Brussels - +32-484-566109 - +32-2-3500762
"In July 2005, after several failed attempts to legalise software
patents in Europe, the patent establishment changed its strategy.
Instead of explicitly seeking to sanction the patentability of
software, they are now seeking to create a central European patent
court, which would establish and enforce patentability rules in their
favor, without any possibility of correction by competing courts or
democratically elected legislators."
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
11:28 p.m.
New subject: NAT single interface problems with Lan/Wan
I can get wan communication or lan communication working, but not both.
Setting this, allows wan communication, but then my lan tries to talk with my wan ip.
Single Interface: listen=udp:<lan_ip>:5060 advertise <nat_wan_ip>:5060
I see my wan_ip in the via header, so I assume that's causing the issues for my lan.
Using add_contact_alias for lan, with the advertise in my listen command, results in
[nathelper.c:835]: no need to add alias param
I've tried handle_ruri_alias();, but this doesn't seem to have an affect either.
Setting this, allows lan communication to work fine, but wan has issues.
Single Interface: listen=udp:<lan_ip>:5060
Where is the happy median?
Was going to try path, but I read this is not the correct work-around.
Any clues or hints, are greatly appreciated.
U <carrier_wan_ip>:5060 -> <lan_ip>:5060
INVITE sip:+1<dialed_number>@<lan_ip>:5060 SIP/2.0..Via: SIP/2.0/UDP
<carrier_wan_ip>:5060;branch=z9h
G4bK04B0f1624ef34479ee3..From:
<sip:+1<caller_id>@<carrier_wan_ip:5060;isup-oli=0>;tag=gK0413c49b..To
: <sip:+1<dialed_number>@<lan_ip>:5060>..Call-ID:
184873373_18184370@<carrier_wan_ip..CSeq: 27475 IN
VITE..Max-Forwards: 63..Allow: INVITE,ACK,CANCEL,BYE,PRACK,UPDATE..Accept:
application/sdp, appl
ication/isup, application/dtmf, application/dtmf-relay, multipart/mixed..Contact:
<sip:+1720276
3205@<carrier_wan_ip:5060>..P-Asserted-Identity:
<sip:+1<caller_id>@<carrier_wan_ip:5060>..Supported: 100
rel..Content-Length: 305..Content-Disposition: session;
handling=required..Content-Type: applic
ation/sdp....v=0..o=Sonus_UAC 25113 26131 IN IP4 <carrier_wan_ip..s=SIP Media
Capabilities..c=IN IP4
4.55.10.130..t=0 0..m=audio 15746 RTP/AVP 0 8 18 101..a=rtpmap:0 PCMU/8000..a=rtpmap:8
PCMA/800
0..a=rtpmap:18 G729/8000..a=fmtp:18 annexb=no..a=rtpmap:101
telephone-event/8000..a=fmtp:101 0-1
5..a=sendrecv..a=maxptime:20..
U <lan_ip>:5060 -> <carrier_wan_ip>:5060
SIP/2.0 100 trying -- your call is important to us..Via: SIP/2.0/UDP
<carrier_wan_ip:5060;branch=z9h
G4bK04B0f1624ef34479ee3..From:
<sip:+1<caller_id>@<carrier_wan_ip:5060;isup-oli=0>;tag=gK0413c49b..To
: <sip:+1<dialed_number>@<lan_ip>:5060>..Call-ID:
184873373_18184370@<carrier_wan_ip..CSeq: 27475 IN
VITE..Server: kamailio (3.3.2 (x86_64/linux))..Content-Length: 0....
U <lan_ip>:5060 -> <asterisk_ip>:5060
INVITE sip:+1<dialed_number>@<lan_ip>:5060 SIP/2.0..Via: SIP/2.0/UDP
<wan_ip>:5060;branch=
z9hG4bK5d66.7cb4b7a1.0..Via: SIP/2.0/UDP
<carrier_wan_ip>:5060;branch=z9hG4bK04B0f1624ef34479ee3..Fro
m:
<sip:+1<caller_id>@<carrier_wan_ip:5060;isup-oli=0>;tag=gK0413c49b..To:
<sip:+1<dialed_number>@192.168.
9.130:5060>..Call-ID: 184873373_18184370@<carrier_wan_ip>..CSeq: 27475
INVITE..Max-Forwards: 62..Allo
w: INVITE,ACK,CANCEL,BYE,PRACK,UPDATE..Accept: application/sdp, application/isup,
application/dt
mf, application/dtmf-relay, multipart/mixed..Contact:
<sip:+1<caller_id>@<carrier_wan_ip:5060>..P-As
serted-Identity: <sip:+1<caller_id>@<carrier_wan_ip:5060>..Supported:
100rel..Content-Length: 305..C
ontent-Disposition: session; handling=required..Content-Type:
application/sdp....v=0..o=Sonus_UA
C 25113 26131 IN IP4 <carrier_wan_ip..s=SIP Media Capabilities..c=IN IP4
4.55.10.130..t=0 0..m=audio
15746 RTP/AVP 0 8 18 101..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:18
G729/8000..a=
fmtp:18 annexb=no..a=rtpmap:101 telephone-event/8000..a=fmtp:101
0-15..a=sendrecv..a=maxptime:20
..
Asterisk box:
Retransmitting #4 (no NAT) to <wan_ip>:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP <wan_ip>:5060;branch=z9hG4bK5d66.7cb4b7a1.0;received=192.168.9.130
Via: SIP/2.0/UDP <carrier_wan_ip>:5060;branch=z9hG4bK04B0f1624ef34479ee3
From:
<sip:+1<caller_id>@<carrier_wan_ip>:5060;isup-oli=0>;tag=gK0413c49b
To: <sip:+1<dialed_number>@<lan_ip>:5060>;tag=as1265f4d0
Call-ID: 184873373_18184370@<carrier_wan_ip>
CSeq: 27475 INVITE
Server: Asterisk PBX 1.6.2.14
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:+1<dialed_number>@<asterisk_ip>>
Content-Type: application/sdp
Content-Length: 239
Matt Scott
19 Feb
19 Feb
2:20 p.m.
New subject: NAT single interface problems with Lan/Wan
Hi,
When kamailio is listening on more than one interface is is sometimes
necessary to use force_send_socket() to tell kamailio to use one socket
or the other. This means it will put the correct values into Via and
Record-Route headers. In the example below, I am choosing the socket
based on if the request has come from a dispatcher list, but any other
logic can be used.
#!substdef "!WAN_IP!x.x.x.x!g"
#!substdef "!LAN_IP!y.y.y.y!g"
route[RELAY]{
if (ds_is_from_list("CLUSTER_ID")) {
xlog("L_INFO", "Using WAN_IP for sending\n");
force_send_socket("WAN_IP":5060);
} else {
xlog("L_INFO", "Using LAN_IP for sending\n");
force_send_socket("LAN_IP":5060);
}
t_relay();
}
Regards,
Hugh
On 18/02/2013 18:28, Scott, Matt wrote:
I can get wan communication or lan communication
working, but not both.
Setting this, allows wan communication, but then my lan tries to talk with my wan ip.
Single Interface: listen=udp:<lan_ip>:5060 advertise <nat_wan_ip>:5060
I see my wan_ip in the via header, so I assume that's causing the issues for my lan.
Using add_contact_alias for lan, with the advertise in my listen command, results in
[nathelper.c:835]: no need to add alias param
I've tried handle_ruri_alias();, but this doesn't seem to have an affect either.
Setting this, allows lan communication to work fine, but wan has issues.
Single Interface: listen=udp:<lan_ip>:5060
Where is the happy median?
Was going to try path, but I read this is not the correct work-around.
Any clues or hints, are greatly appreciated.
U <carrier_wan_ip>:5060 -> <lan_ip>:5060
INVITE sip:+1<dialed_number>@<lan_ip>:5060 SIP/2.0..Via: SIP/2.0/UDP
<carrier_wan_ip>:5060;branch=z9h
G4bK04B0f1624ef34479ee3..From:
<sip:+1<caller_id>@<carrier_wan_ip:5060;isup-oli=0>;tag=gK0413c49b..To
: <sip:+1<dialed_number>@<lan_ip>:5060>..Call-ID:
184873373_18184370@<carrier_wan_ip..CSeq: 27475 IN
VITE..Max-Forwards: 63..Allow: INVITE,ACK,CANCEL,BYE,PRACK,UPDATE..Accept:
application/sdp, appl
ication/isup, application/dtmf, application/dtmf-relay, multipart/mixed..Contact:
<sip:+1720276
3205@<carrier_wan_ip:5060>..P-Asserted-Identity:
<sip:+1<caller_id>@<carrier_wan_ip:5060>..Supported: 100
rel..Content-Length: 305..Content-Disposition: session;
handling=required..Content-Type: applic
ation/sdp....v=0..o=Sonus_UAC 25113 26131 IN IP4 <carrier_wan_ip..s=SIP Media
Capabilities..c=IN IP4
4.55.10.130..t=0 0..m=audio 15746 RTP/AVP 0 8 18 101..a=rtpmap:0
PCMU/8000..a=rtpmap:8 PCMA/800
0..a=rtpmap:18 G729/8000..a=fmtp:18 annexb=no..a=rtpmap:101
telephone-event/8000..a=fmtp:101 0-1
5..a=sendrecv..a=maxptime:20..
U <lan_ip>:5060 -> <carrier_wan_ip>:5060
SIP/2.0 100 trying -- your call is important to us..Via: SIP/2.0/UDP
<carrier_wan_ip:5060;branch=z9h
G4bK04B0f1624ef34479ee3..From:
<sip:+1<caller_id>@<carrier_wan_ip:5060;isup-oli=0>;tag=gK0413c49b..To
: <sip:+1<dialed_number>@<lan_ip>:5060>..Call-ID:
184873373_18184370@<carrier_wan_ip..CSeq: 27475 IN
VITE..Server: kamailio (3.3.2 (x86_64/linux))..Content-Length: 0....
U <lan_ip>:5060 -> <asterisk_ip>:5060
INVITE sip:+1<dialed_number>@<lan_ip>:5060 SIP/2.0..Via: SIP/2.0/UDP
<wan_ip>:5060;branch=
z9hG4bK5d66.7cb4b7a1.0..Via: SIP/2.0/UDP
<carrier_wan_ip>:5060;branch=z9hG4bK04B0f1624ef34479ee3..Fro
m:
<sip:+1<caller_id>@<carrier_wan_ip:5060;isup-oli=0>;tag=gK0413c49b..To:
<sip:+1<dialed_number>@192.168.
9.130:5060>..Call-ID: 184873373_18184370@<carrier_wan_ip>..CSeq: 27475
INVITE..Max-Forwards: 62..Allo
w: INVITE,ACK,CANCEL,BYE,PRACK,UPDATE..Accept: application/sdp, application/isup,
application/dt
mf, application/dtmf-relay, multipart/mixed..Contact:
<sip:+1<caller_id>@<carrier_wan_ip:5060>..P-As
serted-Identity: <sip:+1<caller_id>@<carrier_wan_ip:5060>..Supported:
100rel..Content-Length: 305..C
ontent-Disposition: session; handling=required..Content-Type:
application/sdp....v=0..o=Sonus_UA
C 25113 26131 IN IP4 <carrier_wan_ip..s=SIP Media Capabilities..c=IN IP4
4.55.10.130..t=0 0..m=audio
15746 RTP/AVP 0 8 18 101..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:18
G729/8000..a=
fmtp:18 annexb=no..a=rtpmap:101 telephone-event/8000..a=fmtp:101
0-15..a=sendrecv..a=maxptime:20
..
Asterisk box:
Retransmitting #4 (no NAT) to <wan_ip>:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP
<wan_ip>:5060;branch=z9hG4bK5d66.7cb4b7a1.0;received=192.168.9.130
Via: SIP/2.0/UDP <carrier_wan_ip>:5060;branch=z9hG4bK04B0f1624ef34479ee3
From:
<sip:+1<caller_id>@<carrier_wan_ip>:5060;isup-oli=0>;tag=gK0413c49b
To: <sip:+1<dialed_number>@<lan_ip>:5060>;tag=as1265f4d0
Call-ID: 184873373_18184370@<carrier_wan_ip>
CSeq: 27475 INVITE
Server: Asterisk PBX 1.6.2.14
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:+1<dialed_number>@<asterisk_ip>>
Content-Type: application/sdp
Content-Length: 239
Matt Scott
_______________________________________________
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users(a)lists.sip-router.org
http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
--
Hugh Waite
Principal Design Engineer
Crocodile RCS Ltd.
16 Mar
16 Mar
5:06 p.m.
New subject: cant register
hi
i am so nubie about kamailio
i follow the tutorial kamailio skype in one hour.
i install on vps and give the public ip
before i change the kamailio.cfg from the tutorial i run ps -fC kamailio have result and
can register.but when i change kamailio.cfg from the tutorial run ps -fC kamailio no
result and can't register.
any help please need private sip server urgently
thanks
4078
days inactive
4104
days old
5 comments
6 participants
participants (6)
-
Benjamin Henrion -
davy van de moere -
Hugh Waite -
Peter Dunkley -
Scott, Matt -
siraya