Hello,
I'm playing with RPC on my Kamailio 4.4.7, and I getting rare behaivor
with *uac.reg_refresh* command.
When I run it thru kamcmd like docs describes, this is without *l_uuid*
parameter, I'm geting "*error: 500 - Invalid Parameters*" output. And
when I run with *l_uuid* parameter, I get no output. But in both cases,
Kamalio don't refresh data from DB.
But I run this command thru HTTP (JSONRPC-S), both call modes are
equivalent. With or without l_uuid parameter geting no output and
refresh data from DB.
It's a normal behaivor?
Regards, JV
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Javier Valencia | CTO
Centro de Negocios Martín Buendía
Camino de las Cañadas, nº 1C, Portal 1, 2ºG
29649 Mijas (Málaga)
# 951562080 (T) <tel:951562080> | 687486759 (M) <tel:687486759>
# www.voiper.es <http://www.voiper.es/>
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Hi there,
I'm using Kamailio 4.4.7 with UAC remote registrations module.
In my database have all rows with expires column value's 3600, and my
PBX (asterisk) min, max and default expires to 3600. But I can see how
Kamailio it's sending REGISTER each 150segs.
UAC module parameters values;
modparam("uac", "reg_db_url", "mysql://XXX")
modparam("uac", "reg_db_table", "uacreg")
modparam("uac", "reg_contact_addr", "XXX")
modparam("uac", "reg_timer_interval", 5)
modparam("uac", "reg_retry_interval", 60)
My uacreg table it's above 2k rows.
Regards, JV
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Logo <http://www.voiper.es/>
Javier Valencia | CTO
Centro de Negocios Martín Buendía
Camino de las Cañadas, nº 1C, Portal 1, 2ºG
29649 Mijas (Málaga)
# 951562080 (T) <tel:951562080> | 687486759 (M) <tel:687486759>
# www.voiper.es <http://www.voiper.es/>
------------------------------------------------------------------------
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-----------
Este mensaje contiene información confidencial destinada para ser leída
exclusivamente por el destinatario. Queda prohibida la reproducción,
publicación, divulgación, total o parcial del mensaje así como el uso no
autorizados por el emisor. En caso de recibir el mensaje por error, se ruega
su comunicación al remitente lo antes posible. Por favor, indique
inmediatamente si usted o su empresa no aceptan comunicaciones de este tipo
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Las opiniones, conclusiones y demás información incluida en este mensaje que
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entenderá que nunca se ha dado, ni está respaldado por el mismo.
Responsable del Tratamiento de Datos
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SMART RECARGAS , SL
Camino de las Cañadas, 1C PORTAL 1 2º G
29651 MIJAS COSTA (MALAGA)
My use case is such that I want to be able to send a SIP MESSAGE destined
for a single username to multiple users. In order to be able to map that
target username to multiple usernames, I tried using rtjson and evapi/http
(to generate the routing json), but none of them worked.
Is it the right approach? Is there any other way to programmatically
generate multiple targets for SIP request? I am familiar with dynamic
routing but I need more flexibility.
Initially, I tried sending the request to a different username but that
didn't work and got the following message in the http async result route.
kamailio | 19(25) ERROR: nathelper [nathelper.c:1461]: sdp_1918():
Unable to parse sdp body
kamailio | 19(25) NOTICE: <core> [core/kemi.c:124]:
sr_kemi_core_notice(): In evapi msg branch
kamailio | 28(34) NOTICE: {2 6661 MESSAGE q6hikd6tumsr1dh35kvj} <core>
[core/kemi.c:124]: sr_kemi_core_notice(): In evapi reply branch
The message was received on the originally requested to user and also $tu still
points to the original user instead of the new user even after executing
the following code.
KSR.rtjson.init_routes(response);
KSR.rtjson.push_routes();
KSR.nathelper.fix_nated_contact();
In http response, I receive the following and don't see any rtjson errors
when init_routes is executed on it.
{"version":"1.0","routing":"parallel","routes":[{"headers":{"from":{"display":"test_1","uri":"sip:
*test_1*@test.com:9908"},"to":{"display":"test_4","uri":"sip:*test_4*@
test.com:9908
"},"extra":""},"branch_flags":8,"fr_timer":5000,"fr_inv_timer":30000}]}
Please note that the request was sent to *test_2*(a)test.com. I have tried
other combinations of above but nothing works.
I have already searched on Google and read relevant articles and blog
posts.
Thanks
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Hi all,
I've made a post last month regarding losing MySQL connections -
https://lists.kamailio.org/pipermail/sr-users/2020-December/111389.html
At the time I thought connections were dying as a consequence of low
activity and traffic on the proxy. Meanwhile, I've migrated a great number
of equipments to the proxy with Registers being refreshed every 10minutes
and the problem still persists.
In order to try to fix this i've added timeout_interval and ping_interval
from the db_mysql module. My SQL client on the Kamailio machine is
mysql-community-client 5.6.50-2.el7. It writes and reads in a remote
InnoDB database.
This are the logs i get from Kamailio when the problem appears :
Jan 7 09:43:27 sbc_bbt01_active
/usr/local/kamailio-5.4/sbin/kamailio[21735]: ERROR: {1 27880 REGISTER
e5f8f7bc-cbb2-40b3-9037-edacd6276a2b} db_mysql [km_dbase.c:123]:
db_mysql_submit_query(): driver error on query: Lock wait timeout exceeded;
try restarting transaction (1205)
Jan 7 09:43:27 sbc_bbt01_active
/usr/local/kamailio-5.4/sbin/kamailio[21735]: ERROR: {1 27880 REGISTER
e5f8f7bc-cbb2-40b3-9037-edacd6276a2b} <core> [db_query.c:348]:
db_do_update(): error while submitting query
Jan 7 09:43:27 sbc_bbt01_active
/usr/local/kamailio-5.4/sbin/kamailio[21735]: ERROR: {1 27880 REGISTER
e5f8f7bc-cbb2-40b3-9037-edacd6276a2b} usrloc [ucontact.c:1147]:
db_update_ucontact_ruid(): updating database failed
Jan 7 09:43:27 sbc_bbt01_active
/usr/local/kamailio-5.4/sbin/kamailio[21735]: ERROR: {1 27880 REGISTER
e5f8f7bc-cbb2-40b3-9037-edacd6276a2b} usrloc [ucontact.c:1663]:
update_contact_db(): failed to update database
Jan 7 09:43:27 sbc_bbt01_active
/usr/local/kamailio-5.4/sbin/kamailio[21735]: ERROR: {1 27880 REGISTER
e5f8f7bc-cbb2-40b3-9037-edacd6276a2b} registrar [save.c:784]:
update_contacts(): failed to update contact
Jan 7 09:43:27 sbc_bbt01_active
/usr/local/kamailio-5.4/sbin/kamailio[21735]: ERROR: {1 27880 REGISTER
e5f8f7bc-cbb2-40b3-9037-edacd6276a2b} sl [sl_funcs.c:414]:
sl_reply_error(): stateless error reply used: I'm terribly sorry, server
error occurred (1/SL)
Originally I had usrloc db_mode on mode 3 - DB-Only Scheme. In order to try
to mitigate the issue I changed it to mode 1 - Write-Through scheme but
even then I get the same log errors and an "500" error is still sent to the
client. I chose this mode since, as far as I can understand it applies
changes directly to DB but also uses cache. Please correct me if i'm wrong
on that.
Has this issue happened with anyone before? Is there a way to mitigate this
issue? My only constraint is that I need the database to be always updated
since I have an HA setup, and as such, I can't use cache only methods.
Best Regards,
Greetings,
I'm using loose_route() from the RR module and i'm having troubles making
it use the following exception from the code : "There is only one
exception: If the request is out-of-dialog (no to-tag) and there is only
one Route: header indicating the local proxy, then the Route: header is
removed and the function returns FALSE."
My example is a REGISTER without To-TAG which has a Route header with
kamailio address. If i use proxy IP on Route header , loose_route() returns
false as it should. However, if i use an hostname belonging to the proxy
in the route, loose_route() returns true.
I have hostnames and local ips defined in the "DOMAIN" table but it doesn't
seem to be working. Which other places can local hostnames and ips be
configured in order to be seen as local to loose_route() ?
Best Regards
in my scenario call coming from sbc to kamailio proxy and we route to pbx
server in that when
kamailio proxy get the call at that time i am removing Contact header using
remove_hf("Contact") and adding contact Header using
append_hf("$uac_req(hdrs)") for route to PBX server . in that signaling i
am getting modified Contact header but old contact header not removed
properly some junks are remain in the signaling
scenario :
SBC ---------> Kamailio proxy ------> PBX
before using remove_hf("Contact")
INVITE sip:user2@ <sip%3A5678(a)opensips.org>domain.org;transport=UDP SIP/2.0
Via: SIP/2.0/UDP
10.212.xxx.xxx:port;branch=z9hG4bK-524287-1---8c5ac8642f321a9e
Max-Forwards: 70
Contact: <sip:user1@10.212.xxx.xxx:39931;transport=UDP>
To: <sip:user2@ <sip%3A5678(a)opensips.org>domain.org;transport=UDP>
From: <sip:user1@ <sip%3A1234(a)opensips.org>domain.org
;transport=UDP>;tag=891fd646
Call-ID: t9Ff-fCrin_VN1IO9Xf1mA..
CSeq: 2 INVITE
Content-Type: application/sdp
Proxy-Authorization: Digest username="user1",realm="domain.org
<http://opensips.org/>
",nonce="5ff5a660afc72502212b4a611bb0d689efadafab",uri="sip:user2@domain.org
<sip%3A5678(a)opensips.org>;transport=UDP",response="f926e3742cbdcb8c68ea9b5
82ac2dc",cnonce="9b3276d6d37aaa6107a835df8e5b3a87",nc=00000001,qop=auth,algorithm=MD5
User-Agent: Z 3.15.40006 rv2.8.20
Allow-Events: presence, kpml, talk
Content-Length: 243
v=0
o=Z 0 0 IN IP4 10.212.xxx.xxx
s=Z
c=IN IP4 10.212.xxx.xxx
t=0 0
m=audio 8000 RTP/AVP 3 110 8 0 97 101
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
after using remove-hf("Contact") function removing contact header but some
junks are left and that eats next To header line and breaks Invite packet
INVITE sip:user2@domain.org:;transport=UDP SIP/2.0
Record-Route:
<sip:172.xx.xx.xxx;lr=on;ftag=891fd646;vst=AAAAAFFTQVkuCENVXUEpHwNLAQEOSwdcDhwUcG9ydD1VRFA-;vsf=AAAAAFVCQ1U0ChwlAQMMHgBHHwFJVAYVYW5zcG9ydD1VRFA->
Via: SIP/2.0/UDP
172.xx.xx.xxx;branch=z9hG4bKbe4c.9743b885ad3452287530994a2cad6e50.0
Via: SIP/2.0/UDP
10.212.xxx.xxx:39931;rport=39931;received=172.xx.xx.x;branch=z9hG4bK-524287-1---8c5ac8642f321a9e
Max-Forwards: 69
sip:user1@172.xx.xx.x:39931;transport=UDPTo: <sip:user2@domain.org
<sip%3Adevang3032(a)opensips.org>>
From: <sip:user1@domain.org <sip%3Adppatel(a)opensips.org>>;tag=891fd646
Call-ID: t9Ff-fCrin_VN1IO9Xf1mA..
CSeq: 2 INVITE
Content-Type: application/sdp
User-Agent: Z 3.15.40006 rv2.8.20
Allow-Events: presence, kpml, talk
Content-Length: 534
Contact: <sip:user2@172.xx.xx.xxx:5060
<http://sip:dppatel@172.16.16.163:5060/>>
v=0
o=Z 0 0 IN IP4 10.212.xxx.xxx
s=Z
c=IN IP4 172.xx.xx.xxx
t=0 0
m=audio 11210 RTP/AVP 3 110 8 0 97 101
a=rtpmap:3 GSM/8000
a=rtpmap:110 speex/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:97 mode=30
a=fmtp:101 0-16
a=sendrecv
a=rtcp:11211
a=ice-ufrag:lzjregPe
a=ice-pwd:RtA9x3jUWk4yNpJlzPL60nsRck
a=candidate:LhtmUSbPt9BJs4ZC 1 UDP 2130706431 172.xx.xx.xxx 11210 typ host
a=candidate:LhtmUSbPt9BJs4ZC 2 UDP 2130706430 172.xx.xx.xxx 11211 typ host
As you can see Contact header is gone but
sip:1234@172.xx.xx.x:39931;transport=UDP
part left before To header . and its breake invite packet .
Any suggestion will be highly appreciated.
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arising from any third party taking any action, or refraining from taking
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in my requirement call coming from sbc to kamailio proxy and we route to
pbx server in that when
kamailio proxy get the call at that time i am removing Contact header using
remove_hf("Contact") and adding contact Header using
append_hf("$uac_req(hdrs)") for route to PBX server . in that signaling i
am getting modified Contact header but old contact header not removed
properly some junks are remain in the signaling
scenario :
SBC ---------> Kamailio proxy ------> PBX
before using remove_hf("Contact")
INVITE sip:user2@ <sip%3A5678(a)opensips.org>domain.org;transport=UDP SIP/2.0
Via: SIP/2.0/UDP
10.212.xxx.xxx:port;branch=z9hG4bK-524287-1---8c5ac8642f321a9e
Max-Forwards: 70
Contact: <sip:user1@10.212.xxx.xxx:39931;transport=UDP>
To: <sip:user2@ <sip%3A5678(a)opensips.org>domain.org;transport=UDP>
From: <sip:user1@ <sip%3A1234(a)opensips.org>domain.org
;transport=UDP>;tag=891fd646
Call-ID: t9Ff-fCrin_VN1IO9Xf1mA..
CSeq: 2 INVITE
Content-Type: application/sdp
Proxy-Authorization: Digest username="user1",realm="domain.org
<http://opensips.org>
",nonce="5ff5a660afc72502212b4a611bb0d689efadafab",uri="sip:user2@domain.org
<sip%3A5678(a)opensips.org>;transport=UDP",response="f926e3742cbdcb8c68ea9b5
82ac2dc",cnonce="9b3276d6d37aaa6107a835df8e5b3a87",nc=00000001,qop=auth,algorithm=MD5
User-Agent: Z 3.15.40006 rv2.8.20
Allow-Events: presence, kpml, talk
Content-Length: 243
v=0
o=Z 0 0 IN IP4 10.212.xxx.xxx
s=Z
c=IN IP4 10.212.xxx.xxx
t=0 0
m=audio 8000 RTP/AVP 3 110 8 0 97 101
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
after using remove-hf("Contact") function removing contact header but some
junks are left and that eats next To header line and breaks Invite packet
INVITE sip:user2@domain.org:;transport=UDP SIP/2.0
Record-Route:
<sip:172.xx.xx.xxx;lr=on;ftag=891fd646;vst=AAAAAFFTQVkuCENVXUEpHwNLAQEOSwdcDhwUcG9ydD1VRFA-;vsf=AAAAAFVCQ1U0ChwlAQMMHgBHHwFJVAYVYW5zcG9ydD1VRFA->
Via: SIP/2.0/UDP
172.xx.xx.xxx;branch=z9hG4bKbe4c.9743b885ad3452287530994a2cad6e50.0
Via: SIP/2.0/UDP
10.212.xxx.xxx:39931;rport=39931;received=172.xx.xx.x;branch=z9hG4bK-524287-1---8c5ac8642f321a9e
Max-Forwards: 69
sip:user1@172.xx.xx.x:39931;transport=UDPTo: <sip:user2@domain.org
<sip%3Adevang3032(a)opensips.org>>
From: <sip:user1@domain.org <sip%3Adppatel(a)opensips.org>>;tag=891fd646
Call-ID: t9Ff-fCrin_VN1IO9Xf1mA..
CSeq: 2 INVITE
Content-Type: application/sdp
User-Agent: Z 3.15.40006 rv2.8.20
Allow-Events: presence, kpml, talk
Content-Length: 534
Contact: <sip:user2@172.xx.xx.xxx:5060
<http://sip:dppatel@172.16.16.163:5060>>
v=0
o=Z 0 0 IN IP4 10.212.xxx.xxx
s=Z
c=IN IP4 172.xx.xx.xxx
t=0 0
m=audio 11210 RTP/AVP 3 110 8 0 97 101
a=rtpmap:3 GSM/8000
a=rtpmap:110 speex/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:97 mode=30
a=fmtp:101 0-16
a=sendrecv
a=rtcp:11211
a=ice-ufrag:lzjregPe
a=ice-pwd:RtA9x3jUWk4yNpJlzPL60nsRck
a=candidate:LhtmUSbPt9BJs4ZC 1 UDP 2130706431 172.16.16.163 11210 typ host
a=candidate:LhtmUSbPt9BJs4ZC 2 UDP 2130706430 172.16.16.163 11211 typ host
As you can see Contact header is gone but
sip:1234@172.xx.xx.x:39931;transport=UDP
part left before To header . and its breake invite packet .
Any suggestion will be highly appreciated.
--
*Disclaimer*
In addition to generic Disclaimer which you have agreed on our
website, any views or opinions presented in this email are solely those of
the originator and do not necessarily represent those of the Company or its
sister concerns. Any liability (in negligence, contract or otherwise)
arising from any third party taking any action, or refraining from taking
any action on the basis of any of the information contained in this email
is hereby excluded.
*Confidentiality*
This communication (including any
attachment/s) is intended only for the use of the addressee(s) and contains
information that is PRIVILEGED AND CONFIDENTIAL. Unauthorized reading,
dissemination, distribution, or copying of this communication is
prohibited. Please inform originator if you have received it in error.
*Caution for viruses, malware etc.*
This communication, including any
attachments, may not be free of viruses, trojans, similar or new
contaminants/malware, interceptions or interference, and may not be
compatible with your systems. You shall carry out virus/malware scanning on
your own before opening any attachment to this e-mail. The sender of this
e-mail and Company including its sister concerns shall not be liable for
any damage that may incur to you as a result of viruses, incompleteness of
this message, a delay in receipt of this message or any other computer
problems.
Hi All
Is it possible to explicitly define the tls cert to be used per destination by dispatcher?
I'm attempting to integrate with a service that requires domain name presented to match that of the cert. Where the use of dispatcher is probably not as intended anyway as it targets the same set of destinations multiple times (primary and secondary endpoints for the service im integrating with), with multiple distinct set id's (basically a set per customer for multi tenancy) which expects a sip ping per account to verify that connection is alive and healthy.
#Example Dispatcher list
#Account 1
1 sip:a.cloudservice.com:5061;transport=tls 8 1 socket=tls:10.0.0.2:5061;ping_sni=sip.account1.com;ping_from=sip:sip.account1.com:5061
1 sip:b.cloudservice.com:5061;transport=tls 8 1 socket=tls:10.0.0.2:5061;ping_sni=sip.account1.com;ping_from=sip:sip.account1.com:5061;
#Account 2
2 sip:a.cloudservice.com:5061;transport=tls 8 1 socket=tls:10.0.0.2:5061;ping_sni=sip.account2.com;ping_from=sip:sip.account2.com:5061;
2 sip:b.cloudservice.com:5061;transport=tls 8 1 socket=tls:10.0.0.2:5061;ping_sni=sip.account2.com;ping_from=sip:sip.account2.com:5061;
Currently I'm running a solution based around certs with multiple SAN (subject alternative name) defined, but this is a pain to administrate, and not as scalable as I would like. I want to be able to define multiple client:any tls profiles and explicitly send via that profile. Its easy enough to do that in config using the tls xavps to define a server name and/or id. But for the sip options ping by dispatcher to work you need to specify the server name or id before the tm:local-request event route fire's.
I've resorted to hacking away at dispatcher to see what would happen if I set the tls server name or id before the OPTIONS message is sent, and it works great for the first tls client profile matched, but any others you define re-use the initial tcp connection so reuses the first connections tls config, thus presenting the wrong cert (I'm probably wrong but logs show only one tls complete_init() related to dispatcher pings and the second set's pings skips all the tls init logging so suggests some thing is cached/reused, and logging on the service indicates cert name mismatch with set1s cert being offered)
So I've hit a bit of a dead end, is there a way to force a new tcp connection per server name/id
I fear outbound SNI might be a bit of a can of worms, for it to work nicely with dispatcher as is, would be an extension of the proto:host:port format to include a name or id, maybe something like:
tls(name=sub.domain.com):10.0.0.5:5061
But that looks like it would have a lot of wide reaching knock-on effects, so setting it as an attribute like ping_from works fine as I'm currently doing, if I could force a new connection from dispatch.c....
The other thing that's occurred to me is that it might be better to stop trying to adapt a small percentage of dispatcher's functionality and trying to hammer a round peg into a square hole and instead try and create a new module similar to dispatcher in so much that it sends pings (that's all I really need in this case) and use dispatcher as intended with a single set of non-repeated destinations for routing. This would also allow much simpler integration with MS teams (if added ability to set contact address on the background pings) as well as the service im trying to integrate, and a more natural data set to be used as in my example you can see the only difference is the customers domain between each set. How I would imagine it to work would be to create a tls connection per customer domain, then iterate through the connections per destination in pseudo json the data would look like this:
{
Destinations:[
[1,["a.cloudservice.com","b.cloudservice.com"]],
[2,["sip.pstnhub.microsoft.com", "sip2.pstnhub.microsoft.com", "sip3.pstnhub.microsoft.com"]
],
Sources:[
{uri:"sip.account1.com",destinationMapping:[1]},
{uri:"sip.account2.com",destinationMapping:[1,2]}
]
}
And would just run in the background sending pings with SNI support, and if based on dispatcher should make it easy enough adapt things like the destination up and down event routes, and would allow me to use Kamailio much like I would use nginx for web stuff as a reverse proxy and handle TLS offload on my edge gateway proxy.
Any thoughts?, im frustratingly close to having something that will do if I could get a new connection to be created, but I fear my use case might be sufficiently different from dispatchers to make a new module a sensible approach..
Regards
Tim.
Greetings,
I'm doing a migration from a opensips system to kamailio one and i'm trying
to replicate its functionalities.
The system is a Registrar with a pstn gateway. I noticed that on the
opensips version, loose_route() is called on initial requests and it seems
to be used to protect against preloaded routes.
However, on the default Kamailio configuration file loose_route() is only
called on requests in-dialog and the verification mentioned above is not
used.
Why doesn't Kamailio includes this verification? Is there are any security
concerns not using it?
Best Regards,
