Dear all,
I am trying to make a call between 2 sipml5 client over WSS. Everything
works fine and as expected when using WS. While using WSS, the clients are
registered successfully, but when I make a call, the called party is
disconnected from the network and the caller gets "User offline" message.
There a wireshark capture here:
https://drive.google.com/file/d/0B6LMw8kmoAMYMGhrYXNUVXJySEk/view?usp=shari…
The call is made from Alice to Bob, the port Nrs: 4060: PCSCF, 5060:ICSCF,
and 6060 SCSCF. WWS is running on 4443.
The error message that is shown in Firefox console: "The connection to
wss://192.168.0.11:4443/ was interrupted while the page was loading."
Thanks in advance,
Serhat
Hi Guys,
Anybody knows how to get the ip of the new contact of the redirect? I need
it to update one xavp variable that contains the INVITE destination.
6 2017-01-20 12:44:55.138473 172.16.213.38 *172.16.208.121* SIP/SDP 85 Request:
INVITE sip:01152780776@172.16.208.121:5060 |
9 2017-01-20 12:44:55.142296 172.16.208.121 172.16.213.38 SIP 661 Status:
302 Moved temporarily |
10 2017-01-20 12:44:55.156268 172.16.213.38 172.16.208.121 SIP 423 Request:
ACK sip:01152780776@172.16.208.121:5060 |
16 2017-01-20 12:44:55.293349 172.16.213.38 *172.16.208.111* SIP/SDP
95 Request:
INVITE sip:1152780776@172.16.208.111:5060;user=phone |
My cfg:
# Manage failure routing cases
failure_route[MANAGE_FAILURE] {
....
if (t_check_status("302")) {
xlog("Got a 302 - redirecting");
get_redirects("*:1");
* $xavp(ip_b[$avp(current_route)]=>name) =
"172.16.208.111";#Hardcoded for testing*
xlog("Got a 302 - redirecting $hdr(Contact)");
route(RELAY);
#t_relay();
}
Thanks in advance.
Diego.
Hi,
I think Siremis not work correctly in a multi domain scenario if the
extension number are the same for two domains.
I explain a little more. I have two extension:
1 200(a)sip1.domain.org
2 200(a)sip2.domain.org
On the Siremis I create the user 200 with the SIPuser role. When I login
with this user, and go to subscriber data i can see bots extensions.
I think is relate to this select on kamailio database:
.....SELECT T0.id, T0.username, T0.domain, T0.password,
T0.email_address, T0.ha1, T0.ha1b, T0.rpid FROM `subscriber` T0 WHERE
T0.username='200' LIMIT 10
Any hint?
Regards
can someone please guide me where to find the usage of auc module, thanks
in advance.
I am registering the uac in one end then i want to forward that
registration to an external registrar, if i use $du some domain, some of
the ipv6 info is going to be forwarded, i need to make it look more like an
a b2bua than a outbound proxy therefore i need to use the auc module(also i
need to cover all ipv6 info since i am proxying from ipv4 to ipv6).
i am able to register the client to kamailio, then kamailio send the
registration to the external registrar, but when the registrar sends back
the challenge uac does not reply, any ideas how can achieve this.
uac(linphone etc) --registration--->kamailio(registrar
module)------register(uac)--->SBC
sbc----invite---->kamailio(lookup)---------invite---->uac(linphone etc)
this is the code i am using.
# Handle SIP registrations
route[REGISTRAR] {
xlog("REGISTERING ");
if (is_method("REGISTER"))
{
if(isflagset(FLT_NATS))
{
setbflag(FLB_NATB);
# uncomment next line to do SIP NAT pinging
## setbflag(FLB_NATSIPPING);
}
if(af==INET6)
setbflag(FLB_IPV6);
xlog("IT IS IPV6");
if (!save("location"))
sl_reply_error();
route(REGSBCFWD);
exit;
}
}
route[REGSBCFWD]{
xlog("CREATING OTHER LEG TO THE SBC");
if(!is_method("REGISTER"))
{
return;
}
xlog("CREATING PACKAGE $si -- $fU-- $au--$du-- $tU --$ru");
$var(rip) = "sip.somedomain.com";
$uac_req(method)="REGISTER";
$uac_req(ruri)="sip:" + $var(rip) + ":" + "5060";
$uac_req(furi)="sip:" + $tU + "@" + $var(rip);
$uac_req(turi)="sip:" + $tU + "@" + $var(rip);
$uac_req(hdrs)="Contact: <sip:" + $tU + "@"
+ "10.0.0.222"
+ ":" "5060" + ">\r\n";
if($sel(contact.expires) != $null)
$uac_req(hdrs)= $uac_req(hdrs) + "Expires: " +
$sel(contact.expires) + "\r\n";
else
$uac_req(hdrs)= $uac_req(hdrs) + "Expires: " +
$hdr(Expires) + "\r\n";
uac_req_send();
}
REGISTER sip:sip.somedomain.com:5060 SIP/2.0.
Via: SIP/2.0/UDP 10.0.0.222:5060
;branch=z9hG4bK285f.13a7f0c0000000000000000000000000.0.
To: <sip:14052423@sip.somedomain.com>.
From: <sip:14052423@sip.somedomain.com
>;tag=a82eaf71666096790d1397845fbd3254-ebaa.
CSeq: 10 REGISTER.
Call-ID: 357e21ae51d8416d-26605(a)10.0.0.222.
Max-Forwards: 70.
Content-Length: 0.
User-Agent: kamailio (4.4.4 (x86_64/linux)).
Contact: <sip:140524@10.0.0.222:5060>.
Expires: 3600.
.
SIP/2.0 401 Unauthorized.
Via: SIP/2.0/UDP 10.0.0.222:5060
;branch=z9hG4bK4b53.ad842757000000000000000000000000.0.
To: <sip:14052423@sip.somedomain.com>.
From: <sip:14052423@sip.somedomain.com
>;tag=a82eaf71666096790d1397845fbd3254-37da.
Call-ID: 357e21ae51d8416d-26603(a)10.0.0.222.
CSeq: 10 REGISTER.
WWW-Authenticate: Digest realm="sip.somedomain.com",
nonce="f5831435-3e69-4b78-ab6d-dcb0897e477b", algorithm=MD5, qop="auth".
User-Agent: somedomain.com Host:some proxy.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE.
Content-Length: 0.
.
Hello,
We are using Kamailio's perl module and want our custom library to set the
body on a SIP message.
Is it possible to set a SIP message body from within Perl, or can we return
a string to the Kamailio configuration to then use with set_body()?
Thanks for any advice.
--
David Cunningham, Voisonics
http://voisonics.com/
USA: +1 213 221 1092
Australia: +61 (0) 2 8063 9019
root@homer02:~# netstat -altp
Active Internet connections (servers and established)
Proto Recv-Q Send-Q Local Address Foreign Address
State PID/Program name
tcp 0 0 homer02.xxxxx.xxx:sip *:*
LISTEN 16018/kamailio
tcp 0 0 homer02.xxxxx.xxx:5066 *:*
LISTEN 16018/kamailio
Thanks.
JR
> If you use netstat, what is in the recv queue for tcp packets on sip ports?
>
> netstat -altp
>
> Cheers,
> Daniel
>
>
> On 18/01/2017 16:43, JR Richardson wrote:
>> Yes, this is a sipcapture node. I'm listening on a switch port that is
>> set to mirror my VoIP traffic. I see all SIP UDP/TCP packets on the
>> mirror port and the Ethernet port of the host node. Just don't see any
>> TCP packets process in kamailio, debug 3. UDP packets are processed as
>> expected.
>>
>> My config is using port mirror for the capture parameters:
>>
>> modparam("sipcapture", "capture_on", 1)
>> modparam("sipcapture", "hep_capture_on", 0)
>> modparam("sipcapture", "raw_ipip_capture_on", 0)
>> modparam("sipcapture", "raw_moni_capture_on", 1)
>>
>> modparam("sipcapture", "raw_sock_children", 4)
>> modparam("sipcapture", "raw_interface", "eth1")
>> modparam("sipcapture", "raw_socket_listen", "10.99.99.99:5060-5070")
>> modparam("sipcapture", "promiscious_on", 1)
>> modparam("sipcapture", "raw_moni_bpf_on", 1)
>>
>> Is there a method I could diagnose if the SIP TCP Packets are getting
>> from the kernel network process and the kamailio process?
>>
>> # ngrep -d eth1 -W byline host x.x.x.x | /var/run/kamailio/kamailio.pid
>>
>> Or pipe to kamailio local unix socket?
>>
>> I don't know, I'm just guessing.
>>
>> Thanks.
>>
>> JR
>>
>>
>>> Somehow is not clear for me how you have the configuration there ...
>>> before commenting further, this needs to be clarified.
>>>
>>> The node you presented the config is a sipcapture instance, right? What
>>> is sending traffic to it? Is another kamailio with siptrace module? Or
>>> the sipcature agent? Or you have a port mirroring in the router?
>>>
>>> Cheers,
>>> Daniel
>>>
>>>
>>> On 17/01/2017 16:37, JR Richardson wrote:
>>>>> On Mon, Jan 16, 2017 at 10:29:39AM -0600, JR Richardson wrote:
>>>>>> Yes, I'm familiar with the methods sipcapture uses, I don't use HEP,
>>>>>> using raw socket capture, I think this may be a sipcapture issue,
>>>>>> debuging kamailio shows normal startup and processing of UDP SIP
>>>>>> packets, but does not show any activity with TCP packets.
>>>>> I never used HOMER sofar but when I saw your first message my thoughts
>>>>> was that this can't work in a simple way since for TCP you need to
>>>>> complete a 4 way handshake before you can start to send data.
>>>>>
>>>> Interesting. Are you referring to handshaking on the network stack or
>>>> SIP TCP TLS handshaking? I guess I can see it two ways.
>>>>
>>>> 1) if your talking about TCP/IP handshake, even though the SIP packet
>>>> comes into the mirror port on the host node, the kernel processing the
>>>> TCP packet is not establishing a valid connection due to no TCP
>>>> handshake because its only a monitor port, no transmit back, then the
>>>> kernel network stack does not pass the SIP TCP packet to the kamailio
>>>> process for capture because it drops the packet due to no valid
>>>> handshake?
>>>>
>>>> 2) the kernel network stack is passing the SIP TCP packet to the
>>>> kamailio process, but since kamailio cannot handshake back it drops
>>>> the packet and does not process through the sipcapture module. This
>>>> kinda breaks the whole capture ability for homer with SIP TCP. Using
>>>> ngrep, I see all SIP TCP packets, invite -->, trying <--, session
>>>> progress <--, request timeout <--, ack -->, etc...
>>>>
>>>> So how would I diagnose if the network stack is the culprit? Debugging
>>>> kamailio is pretty straight forward, setup and listening for SIP TCP,
>>>> but never see any processing of any TCP packets.
>>>>
>>>> Thanks.
>>>>
>>>> JR
>
Hello,
slightly off-topic, but related to voip -- does anyone have a link that
can quickly share to a site with up to date details about phone
numbering plans for as many country as possible?
Ideally to include the toll and premium number prefixes, the split
between mobile and fixed lines if it is the case.
Cheers,
Daniel
--
Daniel-Constantin Mierla
www.twitter.com/miconda -- www.linkedin.com/in/miconda
Kamailio World Conference - May 8-10, 2017 - www.kamailioworld.com
Hello,
I’m trying to resolve an issue with the ds_is_from_list function from the dispatcher module.
I’m calling it as follows:
ds_is_from_list(”42”, "3", "sip:sip.foo.bar”)
This makes Kamailio log an error:
ERROR: dispatcher [dispatch.c:2327]: ds_is_addr_from_list(): could not resolve sip.foo.bar
sip.foo.bar doesn’t have any A records but there are NAPTR and SRV records:
$ host -t naptr sip.foo.bar
sip.foo.bar has NAPTR record 20 0 "s" "SIP+D2T" "" _sip._tcp.sip.foo.bar.
sip.foo.bar has NAPTR record 10 0 "s" "SIP+D2U" "" _sip._udp.sip.foo.bar.
$ host -t srv _sip._tcp.sip.foo.bar
_sip._tcp.sip.foo.bar has SRV record 10 0 5060 sip1.foo.bar.
_sip._tcp.sip.foo.bar has SRV record 10 0 5060 sip2.foo.bar.
$ host -t srv _sip._udp.sip.foo.bar
_sip._udp.sip.foo.bar has SRV record 10 0 5060 sip2.foo.bar.
_sip._udp.sip.foo.bar has SRV record 10 0 5060 sip1.foo.bar.
$ host sip1.foo.bar.
sip1.foo.bar has address 10.47.11.1
$ host sip2.foo.bar.
sip2.foo.bar has address 10.47.11.2
The following DNS options are set:
dns_srv_lb = yes
dns_try_naptr = yes
dns_naptr_ignore_rfc = no
The Kamailio version is 4.2.8 (installed from the Debian binary package) on Debian Wheezy.
Shouldn’t ds_is_from_list be able to resolve the NAPTR and SRV records? Can I change anything to be able to use ds_is_from_list with the specified domain?
Thanks,
Björn Bylander
Dear list,
I'm planning the upgrade fromn 4.3.6 tree to 4.4.x version on Debian
distribution.
Can i run the apt-get update and upgrade without problem.. or is
possible that i will found some error during upgrade ?
Is the db schema upgraded or not from 4.3 up to 4.4 ?
Regards
Laura
hello all,
I want to use kamailio like as a proxy SIP (outbound proxy) to transcode
tls to udp like us:
call flow is:
Softphone --> tls --> Kamailio server (kamailio listen on 443 port) -->
udp --> Asterisk server (asterisk listen on 5060 port)
I know Kamailio can convert tls to udp.
According tm documentation
http://kamailio.org/docs/modules/devel/modules/tm.html, I think I must use
t_relay_to_udp().
Do you confirm I must replace t_relay() function by t_relay_to_udp()
function?
Thank for your help.
Regards
Abdoul.
