Hello. I install kamailio 4.1.3 and it works fine. But I neen proxy DTLS
SRTP to backend media servers through kamailio. rtpproxy no not work with
dtls srtp? so I installed rtpengine but can not find rtpengine.so file for
copiyng to kamailio modules directory.
Does anyone can install rtpengine as
10:38 GMT+02:00 Yuriy Gorlichenko ovoshl...@gmail.com:
Hello. I install kamailio 4.1.3 and it works fine. But I neen proxy DTLS
SRTP to backend media servers through kamailio. rtpproxy no not work with
dtls srtp? so I installed rtpengine but can not find rtpengine.so file for
copiyng
(that is not write).
So I have 2 questions after this:
1. What about flag? Why rtpengine does not know it?
2. How send write stun from kamailio to client and asterisk?
2014-07-03 23:46 GMT+04:00 Andrew Pogrebennyk apogreben...@sipwise.com:
On 03/07/14 16:42, Yuriy Gorlichenko wrote:
thanks
I see SDP body at Asterisk behind kamamilio, And I see that body have no
changes.
2014-07-17 1:37 GMT+04:00 Richard Fuchs rfu...@sipwise.com:
On 07/16/14 17:25, Yuriy Gorlichenko wrote:
Hello Rtpengine (rtpproxy-ng module) works fine with kamailio till today.
Without any changes
Hello. I have Kamailio running behind NAT. It lesten eth0 with ip
192.168.0.3 and
I have external IP that have domain name (for example sip.myserver.com).
Register packets from clients comes from external IP.
If I write at kamailio.cfg:
alias=sip.myserver.com
I see error at log - bad_uri
with advertised_address=sip.myserver.com.
Fred Posner
The Palner Group, Inc.
http://www.palner.com (web)
+1-503-914-0999 (direct)
+1-954-472-2896 (fax)
On 08/02/2014 07:51 AM, Yuriy Gorlichenko wrote:
Hello. I have Kamailio running behind NAT. It lesten eth0 with ip
192.168.0.3 and
I
Hello. I Installed kamailio 4.1.4, when I starting my server I see
following Warning
no fork mode and more than one listen address found (will use only the
first one)
So I see only first listening port at my netstat
At my kamailio.cfg file I write fork=yes? but it not helps me.
I found the
, Aug 5, 2014 at 12:09 PM, Yuriy Gorlichenko ovoshl...@gmail.com
wrote:
Hello. I Installed kamailio 4.1.4, when I starting my server I see
following Warning
no fork mode and more than one listen address found (will use only the
first one)
So I see only first listening port at my netstat
-l eth1 -l eth2
Try this?
On Tue, Aug 5, 2014 at 12:21 PM, Yuriy Gorlichenko ovoshl...@gmail.com
wrote:
I Already use it.
As I see my problem about to run the main kamailio process in foreground
and fork child processes as usual.
2014-08-05 23:18 GMT+04:00 Brandon Armstead bran
Hello. I try to integrate dispathcer module (with db table dispatcher) to
my Kamailio server.
I have problem with changing IP at db table, but when I calling - call goes
to old IP that alread deleted from db.
I tested my table with query that selects destination address from table,
and it
,
Kristian Høgh
On Friday 08 August 2014 01:23:43 Yuriy Gorlichenko wrote:
Hello. I try to integrate dispathcer module (with db table dispatcher)
to
my Kamailio server.
I have problem with changing IP at db table, but when I calling - call
goes
to old IP that alread deleted from db.
I
Hello I try to use kamctl dispatcher reload cmd but in gives me an error
kamctl dispatcher reload
ERROR: Error opening Kamailio's FIFO /tmp/kamailio_fifo
ERROR: Make sure you have the line 'modparam(mi_fifo, fifo_name,
/tmp/kamailio_fifo)' in your config
ERROR: and also have loaded the mi_fifo
to kamctlrc is optional,
since it is the default path.
- Deep N
On Mon, Aug 11, 2014 at 5:16 PM, Yuriy Gorlichenko ovoshl...@gmail.com
wrote:
Hello I try to use kamctl dispatcher reload cmd but in gives me an error
kamctl dispatcher reload
ERROR: Error opening Kamailio's FIFO /tmp
Kamailio running perfectly. No error messages. Errors only with kamctl
(kamcmd)
2014-08-12 13:18 GMT+04:00 Daniel-Constantin Mierla mico...@gmail.com:
Was Kamailio running at that time? Any error message in syslog file?
Cheers,
Daniel
On 11/08/14 23:46, Yuriy Gorlichenko wrote
Hello. I Use UAC module for register to my porviders (I have several
porviders at my server and). I successfully ring from provider to my
asterisk servers and route calls to users that registers at Kamailio.
So now I need route calls from users to my providers, But can not
understand how to send
Hello I use UAC module to register trunks to my providers. I've try to call
to my provider. I send INVITE to my porvider and then It recieve me 407
reply proxy authentication required.
Offcourse I can handle this reply manualy (by adding headers throuch
header_add) and add md5 summ, but I think
server indicates
response with 401|407 status at onreply_route and never sends to failure
route. How say to Kamailio to indicate response 401|407 as a failure?
2014-08-15 18:19 GMT+04:00 Daniel-Constantin Mierla mico...@gmail.com:
Hello,
On 15/08/14 00:24, Yuriy Gorlichenko wrote:
Hello I
the t_on_failure() is executed and not
overwritten by another t_on_failure()?
You can used debugger module with cfgtrace turned on to see what actions
are executed from kamailio.cfg
Cheers,
Daniel
On 17/08/14 13:33, Yuriy Gorlichenko wrote:
Hi, I want to realise scheme:
Asterisk-Kamailo-provider
I
Use UAC module for this
20.08.2014 7:40 пользователь Satish Patel satish@gmail.com написал:
We have setup Kamailio front and SIP Proxy, Now i want to Trunk it with
other SIP provide they gave me IP, Username/Password. How do i configure
username/password on Kamailio SIP Proxy?
using rtpproxy_ng or
mediaproxy? If yes can you provide me with some details?
Thanks in advanced!
Sent from my “contract free” BlackBerry® smartphone on the WIND network.
-Original Message-
From: Yuriy Gorlichenko ovoshl...@gmail.com
Sender: sr-users-boun...@lists.sip-router.org
Date
,
see avpops or sqlops for how to load values from database and access them
in config file via variables.
Cheers,
Daniel
On 19/08/14 22:53, Yuriy Gorlichenko wrote:
Hello. I suceesfully authenticate some accounts of providers from UAc
using DB.
When I try to Call to any provider
! I registered remote Trunk using UAC module. so now i can just use
following function to forward my call right?
rewritehost()
On Wed, Aug 20, 2014 at 12:33 AM, Yuriy Gorlichenko ovoshl...@gmail.com
wrote:
Use UAC module for this
20.08.2014 7:40 пользователь Satish Patel satish
rewritehost() sucessfully work with UAC. But As I know
1) It statless function
2) It read only string argumetns, and do not read variables
2014-08-21 14:43 GMT+04:00 Satish Patel satish@gmail.com:
I will give it a try again today, can you please make sure my t_relay()
syntax is correct?
problem to and waiting answer.
2014-08-28 16:57 GMT+04:00 Daniel-Constantin Mierla mico...@gmail.com:
On 28/08/14 14:45, Olle E. Johansson wrote:
On 28 Aug 2014, at 14:14, Yuriy Gorlichenko ovoshl...@gmail.com wrote:
Hello. I try to provide call scheme:
internal client - asterisk
:
On 28/08/14 14:45, Olle E. Johansson wrote:
On 28 Aug 2014, at 14:14, Yuriy Gorlichenko ovoshl...@gmail.com wrote:
Hello. I try to provide call scheme:
internal client - asterisk - Kamailio - provider - external endpoint
call
when I make call I see this:
asterisk kamailio
Hello All. I have kamailio with provider connection (trunk)
When I call to external number through my provider call extablished Ok. But
when i try hangup call from external number no BYE sended to me. When I
hangup call from my kamailio (internal num) I send by to exteral number and
it respond me
mandatory or not, but you can try to get the
contact there.
Cheers,
Daniel
On 05/09/14 08:37, Yuriy Gorlichenko wrote:
Hello All. I have kamailio with provider connection (trunk)
When I call to external number through my provider call extablished Ok.
But when i try hangup call from external
/09/14 12:55, Yuriy Gorlichenko wrote:
RFC not specified Contack header at ACK... So anyway I already tried it
yesterday)) Unsuccessfull...
2014-09-05 12:54 GMT+04:00 Daniel-Constantin Mierla mico...@gmail.com:
Hello,
I noticed that the ACK is missing the Contact header -- not sure
Hello. I tried to use UAC module at my system for registration multiple
providers, and call through it.
When I implementing this at my cfg file I had some troubles:
-- with Authentification at INVITE (using UAC_AUTH). IT does not send
second INVITE packet with www_auth header for some trunks
Hello. I try to test with SIPp my stak of kamailio-asterisk. I run SIPp
with 200 calls/sec and see only 68 at maximum active calls at server. When
I set 500 calls/sec with limit 1000 I see 68 active connections again.
So when I try test SIPp to asterisk without Kam i see wright maximum of
active
On 26/09/14 03:42, Yuriy Gorlichenko wrote:
Hello. I try to test with SIPp my stak of kamailio-asterisk. I run SIPp
with 200 calls/sec and see only 68 at maximum active calls at server. When
I set 500 calls/sec with limit 1000 I see 68 active connections again.
So when I try test SIPp
Hello. We use 2 kamailio behind load balanser that have domain name of our
system .
uac settings like this:
modparam(uac,restore_mode,none)
modparam(uac,reg_db_url, DBURL)
modparam(uac, reg_db_table, uacreg)
modparam(uac, reg_timer_interval, 20)
modparam(uac, reg_retry_interval, 10)
Кamailio is a proxy server. Call is simple sip seesion for it. It does not
have briges or thomething else. Use asteirsk or freeswitch as backend
application server.
2014-10-02 11:23 GMT+04:00 Marino Mileti marino.mil...@alice.it:
Hi,
Is it possible to use kamailio to manage a very simple
Hello. we use UAC module for trunk registration at kamailio.
We have successfull routing and calling through our trunks to many
providers and have just one problem with ATT endpoints. When callee at
ATT endpoint hook call (I see OK from provider as response to our INVITE
from ATT endpoint) and
Hello. I have multiple endpoints registered at my kamailio with one account
(for example user1 registerd from norhway, USA, and Russia at one time), so
when I call from user2 to user1 I want to ring all endpoints registered by
user1 account. Now I can ring only one, first entry at location table.
Hello. We have multiple kamailio behind load balancer. We use UAC to send
REGISTER procedure to provider. Some providers drop wrong ip. So We have
loadbalanser with external IP and have kamailio servers with external IPs
too. So we need to send REGISTER packets to porovide with source IP of our
Thanks. I used some custom thiks to change $ru. Now I will delete it.
2014-10-07 0:33 GMT+04:00 Daniel-Constantin Mierla mico...@gmail.com:
Hello,
are you using t_relay()? Parallel forking should be default with that
function.
Cheers,
Daniel
On 06/10/14 22:02, Yuriy Gorlichenko wrote
without leading 'sip:'. It could be because of config file
operations, not parameters.
Cheers,
Daniel
On 02/10/14 01:13, Yuriy Gorlichenko wrote:
Hello. We use 2 kamailio behind load balanser that have domain name of
our system .
uac settings like this:
modparam(uac,restore_mode,none
- myst I handle responses of this branches if one of it picked
up? (for example user at canada answers, and must i handle canseling calls
to USA user for myself or kamailio breack this branch for itself without my
handling?)
Thanks.
2014-10-07 19:30 GMT+04:00 Yuriy Gorlichenko ovoshl...@gmail.com
Hello. I hawe porblem with based at websocket transport clietns. I use wss
and with chain sertificate.
When I do call with this sert signalling goes normally but sound starts
after fiev seconds of picking up. I debug RTP with tshark at my servers and
localize issue at the kamailio. Ussue
no delay can be introduced
by it.
On the other hand, webrtc uses ice to negotiate rtp relaying, and that can
take several seconds.
Cheers,
Daniel
On 12/10/14 20:48, Yuriy Gorlichenko wrote:
Hello. I hawe porblem with based at websocket transport clietns. I use wss
and with chain sertificate
Hello. I try to do parallel fork calls to endpoints that have same
username and different destination URI. Logic of my script:
checking location table for rows with needed account
get info from contact at loop
for every step
check technology (sip or ws)
append_branch with existing destination
Forget some things:
All calls going form asterisk viaUDP.
t_relay calling out of loop. At the end.
When t_relay at loop first packet goes great but INVITE to second
destinationd not going and ended with ERROR can't generate 200 reply when
a final 200 was sent out
2014-10-17 20:33 GMT+04:00 Yuriy
me to solve this issue, or give me an advice
about this issue? Thks
2014-10-17 20:41 GMT+04:00 Yuriy Gorlichenko ovoshl...@gmail.com:
Forget some things:
All calls going form asterisk viaUDP.
t_relay calling out of loop. At the end.
When t_relay at loop first packet goes great but INVITE
What you mean under full set of flags? At reply I use mirror (+/-) flags
off course. More, it work without branches fine ( i select only one
endpoint). I have issue only with branches.
23.10.2014 18:56 пользователь Richard Fuchs rfu...@sipwise.com написал:
On 10/23/14 06:03, Yuriy Gorlichenko
Still have same error...
Now rtpproxy_manage(co-sp) for classic call. At log I see that rtpproxy
wirked gine. For each step it generate write body, but t_Relay still send
strange compinated packet to UDP with SDP for WS...
2014-10-23 20:42 GMT+04:00 Yuriy Gorlichenko ovoshl...@gmail.com:
Oh. Ok
15:06, Yuriy Gorlichenko wrote:
Still have same error...
Now rtpproxy_manage(co-sp) for classic call. At log I see that
rtpproxy wirked gine. For each step it generate write body, but t_Relay
still send strange compinated packet to UDP with SDP for WS...
Do you mean that the outgoing packet
) and body sets as body of first step (for WS packet)
2014-10-23 23:36 GMT+04:00 Yuriy Gorlichenko ovoshl...@gmail.com:
No SDP body only one. but packet like this
INVITE
sip:device-200@sip:1.21.10.2:45437;rinstance=07f88c423145358e;transport=UDP
SIP/2.0
Record-Route: sip:sip.myservice.com
Hello. I use kamailio with last rtpengine and
I have 5-7 Seconds voice delay. This happened only for from webphone. But
it is not client issue as i see. Wireshark at client side shows that RTP
starts as soon I pick up call. So rtp leaves rtpengine and goes to the
destination with delay... I use
Hello. I use kamailio for calling to porvider. My providr seccefuully
registered from UAC module, but when I try to call through it? it back 401
Unauthorised. I send second try with Digest Auth header at INVITE and it
receive me 401 too...
I register this provider from asterisk and call
Does it possible increase cSeq manually (for example remove and then
append headers?) for UAC module when send INVITE messages with Auth, or
kamailio have pseudovar for this header?
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
As I understand UAC module can not be used at production as module
foroutgoing calls from kamailio to provider with this limitations?
2014-10-30 18:24 GMT+04:00 Pavel Eremin eremina@gmail.com:
No way. Use sems or b2b.
30.10.2014 19:59 пользователь Yuriy Gorlichenko ovoshl...@gmail.com
at:
-
http://by-miconda.blogspot.de/2014/10/kamailio-42-tips-7-increment-cseq-for.html
Let me know if works ok for you, as I did not test it yet extensively.
Cheers,
Daniel
On 30/10/14 16:11, Yuriy Gorlichenko wrote:
As I understand UAC module can not be used at production as module
:26 GMT+04:00 Yuriy Gorlichenko ovoshl...@gmail.com:
Thanks for answer. Now will insttall it for tests.
2014-10-30 20:01 GMT+04:00 Daniel-Constantin Mierla mico...@gmail.com:
This feature (increasing/decreasing cseq for calls authenticated to the
next hop by kamailio) is available with 4.2.0
,
Contact:sip:vebinar-...@sip.myservice.com:5068
Can you verify is a valid one.
On Wed, Oct 29, 2014 at 3:56 PM, Yuriy Gorlichenko ovoshl...@gmail.com
wrote:
Hello. I use kamailio for calling to porvider. My providr seccefuully
registered from UAC module, but when I try to call through it? it back
Does I need to use $dlg_var(cseq_diff) before UAC_AUTH()?
If yes - How. Documentation say only that this var stores Difference
between CSeq...
2014-10-31 1:58 GMT+04:00 Yuriy Gorlichenko ovoshl...@gmail.com:
Daniel. I installed new Kamailio 4.2.
I set dialog module params like
Hello. I need to increment CSeq value for INVITE with Auth params when use
UAC_AUTH for outgoing calls to provider.
Kamailio 4.2 may increment this using dialog module
http://by-miconda.blogspot.de/2014/10/kamailio-42-tips-7-increment-cseq-for.html
Now I experements with this and var
internal flag to tell dialog to increment cseq.
Cheers,
Daniel
On 01/11/14 16:29, Yuriy Gorlichenko wrote:
Hello. I need to increment CSeq value for INVITE with Auth params when use
UAC_AUTH for outgoing calls to provider.
Kamailio 4.2 may increment this using dialog module
http
ERROR: t_should_relay_response: status rewrite by UAS: stored: 491,
received: 200
Hello. I use UAC module for trunks form 4.2 master branch. I have an
error when endpoit at the trunk picked up. OK reply never replied to
caller because I see next error:
ERROR: t_should_relay_response: status
Hello. I use UAC module for trunks and have trouble when call to mobile
endpoint of ATT provider.
When endpoint pickup call ans 200 OK reply come into kamailio, kamailio
send CANCEL. so at the endpoint I see than call is dropped and at kamailio
client I hear voicemail speech form ATT endpoint.
Hello. We use kamailio 4.3 and dispatcher with 9 algorithm. We use db
instanse for dispatcher and at attrs column for our backend servers set
WEIGHT=40 and 60 for first and second server, but packets sended only at
first server ignoring weight
___
SIP
We solved this issue with another way because enpoints can registar at only
one kam (at location table we see socket field). So we need not register at
both servers one endpoint- wee ned call all servers for calling endpoints
with same username from both servers. This philosophy right for load
Hello I use dipatcher algorithm 8 that works with weight. I added 2
Asterisks and try to call its with my kam.We use 4.3 version.
Tthis config select needed dst from database with my scenario.
if(!ds_select_dst($var(setid), 8))
$var(setid)- is variable for setting setid that i get from
Hello. I try to use NDB_REDIS with remote REDIS DB and can not to connect
because remote DB use password, but Kamailio module have no any variable or
attr of modparam that implenets password for DB.
How I can connect to my REDIS?
___
SIP Express Router
}
}
}
}
}
}
I hope that helps.
All the best.
Will
On Tue, Jan 27, 2015 at 3:12 AM, Yuriy Gorlichenko ovoshl...@gmail.com
wrote:
Hello I use dipatcher algorithm 8 that works
Hello. I need parallel forking calls with the same username. (Call to all
contacts with name for example User123), my endpoints may be WebSocket
based and standart UDP endpoints. And I use rtpengine_manage for nmanaging
calls wor webphones and standart softh/hard phones.
I get all contacts
Hello. We use 2 kamailio server 4.3 master brancher for load balansing
cluster. We have some problems with this deplooiment
fist of all we have REgstering issue: Witth one server all works fine (we
have some endpoints with same creditians) all endpoints reinging well. But
at 2 servers we hawe
Hello. I need parallel forking calls with the same username. (Call to all
contacts with name for example User123), my endpoints may be WebSocket
based and standart UDP endpoints. And I use rtpengine_manage for nmanaging
calls wor webphones and standart softh/hard phones.
I get all contacts
:
You probably look for priority based routing -- see the readme of
dispatcher module.
Cheers,
Daniel
On 09/01/15 17:52, Yuriy Gorlichenko wrote:
I as wrote before - we find dispatcher algorithm than can do mechanism
something like this:
Try call to fist server with max priority or weight
.
2015-01-09 20:23 GMT+03:00 Daniel-Constantin Mierla mico...@gmail.com:
You probably look for priority based routing -- see the readme of
dispatcher module.
Cheers,
Daniel
On 09/01/15 17:52, Yuriy Gorlichenko wrote:
I as wrote before - we find dispatcher algorithm than can do mechanism
Hello. We use 2 kamailio servers cluster and we have porblems with db.
Database failed pecause of error:
Could not execute Write_rows_v1 event on table production.location;
Duplicate entry 'uloc-54aae947-86d-a67' for key 'ruid_idx', Error_code:
1062; handler error HA_ERR_FOUND_DUPP_KEY; the
I as wrote before - we find dispatcher algorithm than can do mechanism
something like this:
Try call to fist server with max priority or weight. OIf this server
unavailible then call second server with less weight and etc.
Does anyone know what ling of algorithm we can use for this?
12:31 GMT+03:00 Daniel-Constantin Mierla mico...@gmail.com:
On 13/01/15 16:52, Yuriy Gorlichenko wrote:
Daniel. I added 8 algorithm to our server and it works with 2 asterisk
now but it works strange because:
While works server with priority 1 - all ok. When this server goes down
with lowes priority until
this server not goes down.
2015-01-12 15:18 GMT+03:00 Yuriy Gorlichenko ovoshl...@gmail.com:
Daniel. Hello. I see changes at documentation about algorithms at 4.3
documentation for dispatcher. Now I see than 8 algo use priority. Not I set
this algorithm to my servers
Daniel. I added 8 algorithm to our server and it works with 2 asterisk now
but it works strange because:
While works server with priority 1 - all ok. When this server goes down
dispatcher choose next server with lowes priority. But when server with
highest priority waking up dispatcher use server
Hello I use this version of kamailio
kamailio -v
version: kamailio 4.3.0-dev3 (x86_64/linux) 8cdbe7
flags: STATS: Off, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS,
DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC,
F_MALLOC, DBG_F_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT,
as extra
branches.
You may want to set a branch flag when processing the REGISTER to know
that it is a websocket. Then, you can test the same branch flag in
branch_route to discover if the destination is over websocket or not.
Cheers,
Daniel
On 10/02/15 12:01, Yuriy Gorlichenko wrote:
Hello
Hello. How I must use this function for dynamic reload dispatcher without
restarting me server?
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
are you using? Can you paste here the records you have for
the destination set (you can replace the ip addresses, I am interested in
attributes) and the ds_select_dst() or ds_select_domain() lines from your
config?
Cheers,
Daniel
On 08/01/15 04:07, Yuriy Gorlichenko wrote:
Hello I use
);
}
}
route[FINAL_RELAY]
{
if (!t_relay()) {
sl_reply_error();
}
return;
}
2015-03-16 15:46 GMT+03:00 Yuriy Gorlichenko ovoshl...@gmail.com:
request_route
2015-03-16 15:43 GMT+03:00 Daniel-Constantin Mierla mico...@gmail.com:
Is this used in request_route or in failure_route or other
: + $tU + @ + $(du{s.select,1,:});
Cheers,
Daniel
On 16/03/15 05:44, Yuriy Gorlichenko wrote:
Now. when I use
seturi(sip:$tU@$(du{s.select,1,:}));
I see error at my log
ERROR: tm [t_lookup.c:1264]: new_t(): ERROR: new_t: uri invalid
ERROR: tm [t_lookup.c:1411]: t_newtran(): ERROR
request_route
2015-03-16 15:43 GMT+03:00 Daniel-Constantin Mierla mico...@gmail.com:
Is this used in request_route or in failure_route or other routing block?
Cheers,
Daniel
On 16/03/15 12:44, Yuriy Gorlichenko wrote:
If I use
$ru=sip:+$tU+@+$(du{s.select,1,:});
if (!t_relay
Hello. I try to call multi[ple endpoints from my server using
append_branch. It works fine but when I have only one endpoint - kamailio
generate 2 INVITE requests to it.
As I understand it is original request and the next one is branch.
I used seturi() before for sending original reqest to
to process the URI (479/SL)
As i see error generate twice maby because I ure t_on branch() route
2015-03-16 7:18 GMT+03:00 Yuriy Gorlichenko ovoshl...@gmail.com:
Hello. I try to call multi[ple endpoints from my server using
append_branch. It works fine but when I have only one endpoint - kamailio
I need to remove all header line witht tags but remove_hf() removes only
Header:value
All tags after ; stay at the packet as garbage and next Header moves up
As there
ACK sip:12345678...@phone.provider.com SIP/2.0
Via: SIP/2.0/UDP sip.server.com:5068
#why_changes_made_to_headers_or
Cheers,
Daniel
On 03/03/15 01:28, Yuriy Gorlichenko wrote:
I need to remove all header line witht tags but remove_hf() removes only
Header:value
All tags after ; stay at the packet as garbage and next Header moves up
As there
ACK sip:12345678
What type of info can I provide for deeper analys of this situation?
2015-02-27 14:34 GMT+03:00 Javi Gallart jgall...@systemonenoc.com:
Hello
On 27/02/15 11:58, Yuriy Gorlichenko wrote:
at the monitor I see nothing about this request
It's difficult to say without further information
Hello I try to get some replies from redis. Time after time redis request
give me null result. But redis bs not disconnected.
This happens only with websocket endpoints. My queries is:
redis_cmd(srv1, EXISTS $si, s);
So at xLOG i see that $si correctly sended, but result is null. At db I
keep
I will try. I new at redis. Does cli monitor get resul of kamailio request
at the cli?
2015-02-27 11:38 GMT+03:00 Javi Gallart jgall...@systemonenoc.com:
Hello
can you check with redis-cli monitor what is the command sent to Redis
in that case?
Javi
On 27/02/15 09:15, Yuriy Gorlichenko
at the monitor I see nothing about this request
2015-02-27 13:21 GMT+03:00 Yuriy Gorlichenko ovoshl...@gmail.com:
Now I see that null values recieved after I see this at kamailio log
redisc_exec(): Redis error: Server closed the connection
2015-02-27 12:45 GMT+03:00 Javi Gallart jgall
that kamailio is
delivering to the server when you execute the redis_cmd(...) function
inside the script.
For a non null reply, yo will need a key in redis with the same value as
$si.
Javi
On 27/02/15 10:04, Yuriy Gorlichenko wrote:
I will try. I new at redis. Does cli monitor get resul
Hello. We try to use redis for maximum features of kamailio.
We already realise dispatcher (not as module, but I want to do it at the
future), and now we want to relocate usrloc to redis. Does anyone do this
with Redis?
___
SIP Express Router (SER) and
to redis
Javi
On 27/02/15 14:30, Yuriy Gorlichenko wrote:
What type of info can I provide for deeper analys of this situation?
2015-02-27 14:34 GMT+03:00 Javi Gallart jgall...@systemonenoc.com:
Hello
On 27/02/15 11:58, Yuriy Gorlichenko wrote:
at the monitor I see nothing about
. For instance, is your redis instance in the same machine
as Kamailio? Are you using unix sockets or tcp sockets? Can you run
kamailio in debig mode to see any potentially helpful message?
Javi
On 06/03/15 08:04, Yuriy Gorlichenko wrote:
arrives kamailio stil disconnects from redis. Haw can I debug
:
Hello,
does that happen in all cases or just for some records? Can you rung with
debug=3 and check the syslog messages for what happens at that moment when
401 is processed?
Cheers,
Daniel
On 30/04/15 11:37, Yuriy Gorlichenko wrote:
Hello. We have an issue with REGISTER to Provider
Hello. We have an issue with REGISTER to Provider. When Provider answers
401 Kamailio don't send any REGISTER with digest auth
IP ourservice.com.5068 provider.dev.5060: UDP, length 468
E...U...@.3'
..AREGISTER sip:provider.dev SIP/2.0
Via: SIP/2.0/UDP ourservice.com:5068
One more thing may be useful for you. If you will get an error with cseq
numder when provider send 401/407 message- usedialog module. It resole an
issuevwith cseq( read documentation)
30.04.2015 18:23 пользователь SamyGo govoi...@gmail.com написал:
I'd like you to google around, there is a
Hello. I thry to integrate redis for location module and first at all that
I do - dublicate location to redis.
First At all I create analog of lookup procedure that use location but from
redis. I take values from location and create branches by mannualy. All
works good but branch route create
Try to start kamailio on Ubuntu 14.04.02
Get this errors
ERROR: ctl [init_socks.c:115]: init_unix_sock(): ERROR: init_unix_sock:
bind: No such file or directory [2]
Jun 23 08:54:29 kamailio-test kamailio[3706]: ERROR: ctl [ctl.c:273]:
mod_init(): ERROR: ctl: mod_init: init ctrl. sockets failed
)
fd_no=5 called
2015-06-23 17:00 GMT+03:00 Roberto Fichera ker...@tekno-soft.it:
On 06/23/2015 03:57 PM, Yuriy Gorlichenko wrote:
Hi Yuriy,
done this.
now kamailio fails when try to fork...
Jun 23 09:56:22 kamailio-test kamailio[4122]: ALERT: core [main.c:725]:
handle_sigs(): child process
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