I was wondering if any NoSQL adapters are available for Kamailio to do
"subscriber authentication" backend?
I was looking at Mongo or CouchDB as two possible backends
Seems like these would be good candidates for Backends for Kamailio.
Any ideas on this?
Th
Using thrift/0.8.0/thrift-0.8.0.tar.gz
I get this when trying to build db_cassandra.
In file included from /usr/local/include/thrift/protocol/TProtocol.h:23,
from /usr/local/include/thrift/protocol/TBinaryProtocol.h:23,
from dbcassa_base.cpp:36:
/usr/local/include/thrift/transport/TTransport.h:3
BUG: [mem/shm_mem.c:245]: destroying the shared
memory lock
On 1/23/12 4:05 AM, Anca Vamanu wrote:
Hi David,
As mentioned in the documentation[1], the module is compatible with
thrift library version 0.6.1.
[1]
http://sip-router.org/docbook/sip-router/branch/master/modules/db
domain(string) last_modified(timestamp)
domain
On 1/23/12 7:45 AM, Anca Vamanu wrote:
Hi David,
How does you your table schema file for domain look like?
I will try myself also with domain module when I have some time.
Regards,
Anca
On 01/23/2012 02:23 PM, David wrote:
I know that it does
Yes,
I should of mentioned this. I did make that change.
I am not sure it still works.
I am also going to go to the office now.
So I will check when I get to work.
But you think this should be supported?
Thanks.
On 1/23/12 8:08 AM, Anca Vamanu wrote:
On 01/23/2012 02:58 PM, David wrote
/shm_mem.c:242]: shm_mem_destroy
0(27209) DEBUG: [mem/shm_mem.c:245]: destroying the shared
memory lock
On 1/23/12 8:08 AM, Anca Vamanu wrote:
On 01/23/2012 02:58 PM, David wrote:
domain(string) last_modified(timestamp)
domain
There is a problem in this schema: last_modified should not be of type
Amazing. I will check Thank You!
On 1/23/12 11:46 AM, Anca Vamanu wrote:
Hi David,
I have just committed an extension of the module to allow queries
without any condition (this is what domain module did and it did not
work). I have tested myself with domain module and it was ok. Update
One of my upstream gateways does not support rpid, so I want to rewrite
the from header from the rpid.
is just a matter if setting $fU = $rpid?
Thanks.
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Thank you. Got it.
On 1/25/12 12:30 PM, Alex Balashov wrote:
On 01/25/2012 12:22 PM, David wrote:
One of my upstream gateways does not support rpid, so I want to
rewrite the from header from the rpid.
is just a matter if setting $fU = $rpid?
No, $fu/$fU/$fd/etc. are not mutable.
What you
Anca,
for some strange reason I am getting "SIP/2.0 483 Too Many Hops" back
when I try to register.
I am running kamaikio -EE and I see we get authenticated correctly
but then lots of other info is spit out very quickly and I am having a
hard time trying to understand it.
I think it ma
ethods you mentioned below and report back.
Thanks
David
On 1/26/12 4:22 AM, Anca Vamanu wrote:
Hi David,
I believe it has to do more with your configuration fine than with
location and db_cassandra, but lets see exactly what happens.
First of all do a message capture on the server to see that happe
Sounds like you would really benefit from hiring a consultant.
On 1/30/12 12:00 PM, Me wrote:
Apologies if any of the questions below are a bit dumb - I don't
pretend to be an expert in SIP/VOIP - I am just an ordinary user
looking for answers.
Our current setup involves processing a small n
Please re-read my original reply.
"Sounds like you would really benefit from hiring a consultant. "
Is not "get a consultant".
Its practical advice to help you get started.
You would benefit from it. Get your project done faster and move on. It
also helps support the kamailio community. A sma
Doubt it.
You would need your "media" gateway to detect such a case.
On 2/13/12 10:44 AM, Stoyan Mihaylov wrote:
Problem - if connection drop, call can persist.
In Asterisk there is silencedetecthangup - which should cause hangup,
if there is full silence for desired period of time.
Unfortu
What SIP stack does this use?
Is it open source?
Thanks.
On 2/16/12 6:22 PM, Daniel Pocock wrote:
Another strategy is to modularise the app: e.g. divide Lumicall into 3
apps, each with less permissions, and they collaborate using
inter-process communication (IPC)
Not sure it will help much, f
C3261, it says that
the CSEQ number has to increase and that's what it has done. Can anyone
tell me where the errorlies ?
I tried changeing the check sanity to not check that CSeq is valid, but
it screws up authentication as it mis detects the method type.
Please advise,
David
inter
I was testing the head version in GIT;
I ran across a strange issue.
The new config includes a "WITH_VOICEMAIL" definition
I am not sure exactly how it is designed to work, but by default if the
"usrloc" fails it creates a branch and goes to voicemail
That part work correctly.
It seems howe
his query coming from ( source code/ config file )? Why is
this subscription being deleted almost immediatly after it is setup ?
Thanks,
David
Here is a sample request ( not the same as the one above, I did not
capture the SIP at the same time as the SQL ) :
U PROXYIP:5060 -> P
Hello,
Should I be requiring users to authenticate before letting
them into loose_route(); ? What about anonymous calls from E164, how do
I authenticate these calls after they have started?
Thanks,
David
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call, the phone rings for a
long time ( or endlessly on certain devices ). I tried adding an
Expires: 10, and it did not make any difference.
Is there a way to cancel this call from the ctd script ?
David
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, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 4MB
poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
id: unknown
compiled on 14:19:23 Apr 19 2012 with gcc 4.4.3
David
On 2012-06-20 11:47, David wrote:
Hello,
I am using the example
Do the death stars you are selling have a patch for the small exhaust
port vulnerability (BBY0) ?
David
On 2012-06-20 15:12, Fred Posner wrote:
On Jun 20, 2012, at 2:40 PM, Alex Balashov wrote:
On 06/20/2012 02:38 PM, Fred Posner wrote:
I know a great bakery that can offer dessert
To chime in here,
On behalf of the professional developers on the list, it seems "a guy
giving a try" may get you stated but someone with experience will get
you there faster with a more predictable result.
I would think the going rate today for an experienced kamailio developer
is about $15
to have it purged sooner.
Thanks,
David
version: kamailio 3.2.3 (x86_64/linux)
flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS,
DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC,
DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE,
US
per child or global for kamailio ?
Is there an MI command I can send to check what kamailio's health ? (
load, free memory, etc... ) ?
Thanks,
David
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Hello,
Thanks for the clarification.
I do not see get_statistics on the "micommands" page on the wiki. Where
in the documentation is this command discussed?
Are the stats for all processes or only for the process the responds to
the mi command ?
Thanks,
David
On 2012-09-10 12:
e();
};
If the server does not have any status information, it will send out a
blank notify with a 0 content length. Is there anyway to tell Kamailio
to send an xml doc with status "terminated" if the server does not know
the status that was requested by the SUBSCRIBE?
Than
r is shorter than a round trip to the
presence server.
In the second scenario, Kamailio never updates the presence information
using the first SIP-ETag, so the light on my phone stays stuck at "ringing".
Where do I go from here ?
Thanks,
David
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rification on how the matching is done. Given that I
have realtime table setup, does that mean that dialogs that are in the
table will return true when I call is_known_dlg() ?
Does that mean that if the dialog is NOT in the table, is_known_dlg will
retu
. I do not want to change
the timers in the tm module for the entire proxy, so my question is if
there is a way to set the timeout for the dialog created by the
t_uac_dlg command, and can I set it on a per dialog basis ?
David
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hat future users don't have the same
issue I did.
Thanks,
David
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You can adapt pretty easily.
On 9/21/11 3:46 PM, Henrik Aagaard Sørensen wrote:
Does anyone know if there somewhere exists a tutorial about Kamailio
and FreeSWITCH realtime integration?
I have Googled a lot and found:
http://kb.asipto.com/kamailio:index and
http://kb.asipto.com/freeswitch:i
This is really awesome.
Is any of that stuff on the video open to play with?
(I know it does not support audio but the SIP chat aspect is really great.
Thanks.
On 10/11/11 5:26 AM, Iñaki Baz Castillo wrote:
2011/10/11 Olle E. Johansson:
I think this is awsome. If I understand this right, t
Cool; I will wait until the code is available.
If I understand what I saw correctly you have implemented a JS SIP stack
which runs native in the browser over web-sockets.
On 10/11/11 8:11 AM, Iñaki Baz Castillo wrote:
2011/10/11 David:
This is really awesome.
Is any of that stuff on the
/usr/sbin/kamailio[4668]: ERROR: dialog
[dlg_hash.c:315]: destroying dialog 0x7f74db910e60 (ref 0)
You see it falls to 0 and the dialog is destroyed.
So, how do I debug this ? Should I maybe be posting to sr-dev ?
How do I determine who is incrementing the counter without decrementing it?
Wha
Hey,
Not sure if it matters, but the column 'state' stays at '4' even if the
log says state 5.
David
On 13-02-26 01:08 PM, David wrote:
Hey all,
I am dealing with a possible race condition in the dialog module.
About one call in 50 isn't cleared from the databa
Hello,
Look on Wikipedia and read the articles for SIP, RTP, NAT and STUN.
The answer to your question can be found in the above articles.
David
On 13-02-27 10:26 PM, Khoa Pham wrote:
Hi, I have Kamailio as SIP server and RTP server. Client is PJSIP.
I read that STUN is for non-symmetric
Hello,
Can someone tell me which is executed first ?
the on reply route setup with t_on_reply() or the callbacks in the code
that allows the module dialog to insert the confirmed dialog into the
dialog table ?
Thanks,
David
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;t capture the mi udp packets, but I did get the SIP, see below
I think the problem has to do with expires_offset. I set this to reduce
SUBSCRIBE expiration by 30 seconds, but it is also reducing PUBLISH
expiration by 30 seconds. Do you think this might be the source of my
problem?
What else mig
ader has the server's domain
name and the response has the IP address that set by fix_contact().
So I think that nat_keepalive() should not be comparing the Contact from
the original request but using the contact header from the modified request.
Any ideas what to do next ?
Thanks,
D
e nat keep alive option on
a server that does NAT before t_relaying it to another server that has
usrloc on it?
Thanks,
David
On 13-05-22 04:06 AM, Klaus Darilion wrote:
On 21.05.2013 18:37, David wrote:
Hello,
So I want to setup a Kamailio SIP Proxy( version 4 ) that will do NAT
signa
I can
figure out why and post back my results.
David
On 13-05-23 04:04 AM, Daniel-Constantin Mierla wrote:
Hello,
if you have registrar behind a proxy, then you have to use path module.
Cheers,
Daniel
On 5/22/13 3:17 PM, David wrote:
Hello,
fix_nated_register() doesn't seem to appl
,
David
On 13-05-23 10:27 AM, David wrote:
Hello,
I was using add_path() to add the path, in this case it was the wrong
method.
To anyone who runs into the same problem as me, the solution is that
the proxy doing nat should have add_path_received() instead of
add_path() and the proxy should
Hello,
Thank you for the clarification.
David
On 13-07-15 09:53 AM, Alex Balashov wrote:
On 07/15/2013 09:52 AM, David K wrote:
Are you saying that avp_db_query is obsolete?
It's not obsolete in the sense of deprecated, it's just not the most
flexible, modern way to access SQL
Hello,
I may have misunderstood the thread, but why not use the
override_lifetime parameter ?
http://kamailio.org/docs/modules/stable/modules/pua_dialoginfo.html#idp15277344
Would this not be easier than patching the code?
David
On 13-10-23 09:54 AM, Kristian F. Høgh wrote:
Hi,
I have
can do in Kamailio to resolve this issue ? Is
there an option that I can set that will cause Kamailio to relay the
CANCEL only to devices that have already returned a 100 Trying or 180
Trying ?
What information do you need to know about my config? What parts of the
is gone. Is this the correct solution ?
David
On 2010-06-17 15:49, David wrote:
Hey,
I am using a Cisco WIP310 wifi phone. Seeing as wifi is very battery
demanding, the phone goes into a standby mode. When it's in the
standby mode, it takes a few seconds to come back on.
So I send an INVITE
Hey,
With my Cisco WIP310, what is happening is it is replying 481 Call
Leg/transaction unknown. After that it processes the INVITE
retransmission at which point it rings for 240 seconds. Which is a wee
bit annoying.
David
On 2010-06-17 16:43, Alex Balashov wrote:
David,
You can suppress
hat I could read that would help me better
understand the TM module, please say so.
Thanks,
David
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L request ever be routed using loose_route().
Thanks,
David
On 10-06-18 11:40 AM, Henning Westerholt wrote:
On Friday 18 June 2010, David wrote:
I am using Kamailio 1.5.4.
I read RFC3261 section 17 and the TM doc, I had a few questions :
Does t_relay() manage both the client and server tra
a Proxy to fully support
crossing NATs.
David
On 10-06-20 06:50 AM, JinKevin wrote:
> Hi,
>
> My kamailio and application server are with Public IPs so the system
> uses nathelper without RTPProxy. It works with the eyebeam behind the
> ADSL router, but I cannot hear any voice ( e
Hello,
Still using Kamailio 1.5.4.
What behaviour should the tm module be doing? Is it doing the RFC
behaviour where the CANCEL is only sent after a provisional reply is
received or is it doing some other non standard action ?
David
On 2010-06-17 16:59, Andrei Pelinescu-Onciul wrote:
On
Hey,
Looks like they took the horn people have been buying here in quebec
city for 20 years and renamed it
http://a33.idata.over-blog.com/600x450/1/52/06/88/Carnaval-de-Quebec/Carnaval--364-.JPG
David
On 2010-06-25 08:49, Isamar Maia wrote:
> EhehehehhEHHEhehe.
>
> Vuvuzela
;hnear=Canada&ll=46.812707,-71.213832&spn=0.001301,0.003484&t=h&z=19
David
On 2010-06-25 08:53, David wrote:
Hey,
Looks like they took the horn people have been buying here in quebec
city for 20 years and renamed it
http://a33.idata.over-blog.com/600x450/1/52/06/88/Carnaval-de-Quebec/Ca
t set module parameter
ERROR: bad config file (1 errors)
0(9334) ERROR: dialog [dlg_db_handler.c:177]: invalid database handle
Funny thing is I have not set the module parameter for pua_dialoginfo,
looks like kamailio is getting confused with my pipes, if I name each
param
Hey,
Is it possible that Kamailio 1.5.x silently discarded modparams that
were not valid and Kamailio 3.0 doesn't ?
David
On 2010-06-25 10:11, Klaus Darilion wrote:
IIRC the problem is that the string is interpreted as pattern, thus
"pua" matches "pua_dialoginfo" an
consuming all my server resources cause the timer is completed
instantly, so the message repeats hundreds of times per second.
Thank,
David
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http
be between two devices on the same network.
Is there a command I can execute as the INVITE comes back to cancel the
changes made by RTP Proxy ?
( The config I am using is a slightly modified version of alg.cfg ).
Thanks!
David
___
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one please shed some light on my situation?
Thanks,
David
On 2010-06-17 16:59, Andrei Pelinescu-Onciul wrote:
On Jun 17, 2010 at 16:39, David wrote:
Hello,
I had a look at RFC 3261, section 9.1. The trouble is that Kamailio
is answering "100 Trying" to the caller, so the it
Hello,
Here are the corrected attachments, I sent the email from Window's
wordpad which set it to RTF by default.
Here are proper txt logs.
David
On 2010-07-05 17:47, David wrote:
Hello,
I upgraded to Kamailio 3.0 and the CANCEL is being sent after it
receives a 100 Tryin
Hey,
It's sent when I hit the red button on the phone ( hang up ). It's not
automated.
David
On 2010-07-06 11:39, Klaus Darilion wrote:
Am 06.07.2010 17:04, schrieb Andrei Pelinescu-Onciul:
On Jul 06, 2010 at 16:55, Klaus
Darilion wrote:
Am 06.07.2010 16:36, schrieb Andrei
Hey,
It's sent when I hit the red button on the phone ( hang up ). It's not
automated.
David
On 2010-07-06 11:39, Klaus Darilion wrote:
Am 06.07.2010 17:04, schrieb Andrei Pelinescu-Onciul:
On Jul 06, 2010 at 16:55, Klaus
Darilion wrote:
Am 06.07.2010 16:36, schrieb Andrei
between the first time through and the second time is the
port 5061 instead of 5060.
How can I tell tm that these are not the same transactions ? Would it be
easier to have a seperate instance of Kamailio for 5061 ?
Thanks,
David
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Hey,
I do not do anything IP level forwarding. All my forwarding is done
using Kamailio. It looks like what I am doing is called hairpin routing.
Thanks,
David
On 2010-07-07 18:48, Timo Reimann wrote:
Hi,
David wrote:
I have setup two Kamailio servers, my INVITEs will go through
orbed.
Is the custom header I used a good solution?
David
On 10-07-08 07:04 AM, Klaus Darilion wrote:
Am 08.07.2010 00:53, schrieb David:
Hey,
I do not do anything IP level forwarding. All my forwarding is done
using Kamailio. It looks like what I am doing is called hairpin routing.
I think t
ed ( or dropped ). ( I check the IPs using "ds_is_from_list()" )
The NAT related header is checked like this :
if ( nat_uac_test('19') && !is_present_hf('X-Cool-Header') )
{
fix_nated_contact();
# Set flags for NAT fixing such as RTP Proxy.
}
I am pretty su
.3 setup,
but it would seem in migrating to 3.0, something broke, any ideas ?
Thanks,
David
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hello Daniel
thanks for the explanation.
then i understand the "quick connect" message is also normal? seen in
version 4.2.2 or 4.4?
best regards
david
El mar, 26-01-2016 a las 12:45 +0100, Daniel-Constantin Mierla escribió:
> Hello,
>
> the pending write message is due
ok thanks Daniel
i was quite confused by seeing different things on different versions
(which had sligthly configuration differences) and i though i was doing
something wrong somewhere
best regards
david
El mié, 27-01-2016 a las 14:55 +0100, Daniel-Constantin Mierla escribió:
> He
oot@test:/var/log/rtpengine_conf# cat rtpproxy
set_id(int) url(str) weight(int) disable(int)
1:udp\:x.x.x.x\:x:100:0
2:udp\:x.x.x.x\:x:100:0
is there something i missed here? do i must use mysql or postgres
drivers for this? (i use a test machine to old to install a driver of
those with kam 4.4
hello all
i just noticed it was missing on 4.4.0, but it's present on 4.4.1
:)
regards
david
El mié, 25-05-2016 a las 14:58 +0200, david escribió:
> dear all
>
> i'm using kamailio 4.4.1 and i'm not seeing nh_reload_rtpp as a fifo
> exported function
> i'
[26575]: INFO: test
Call 64431 / Call-ID 04a7c19a-34ad-477e-81bd-659dd4295ae8: Destroying
rtpproxy session on BYE
Jun 28 11:20:05 /usr/local/kamailio/sbin/kamailio[26575]: ERROR:
rtpengine [rtpengine.c:2627]: send_rtpp_command(): can't send command
"delete" to RTP proxy
Jun 28 1
d
before a request retransmission arrives, i guess it should be enough
(¿?)
best regards
david escartin
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sorry with the mess i did with this issue, i had the digest configured
in my subscription and i was not getting the response, so i created 3
mails for the same
:(
El mié, 29-06-2016 a las 13:58 +0200, david escribió:
> hello all
>
> i have been taking a look to the rtpengine.c code an
hello
those variables are for PFI, not PAI
for PAI you have $ai pseudo variable and you can retrieve parts like
$(ai{uri.user}) for username
regards
david
El jue, 30-06-2016 a las 15:34 +0100, José Seabra escribió:
> Hello there,
>
>
> The Pseudo-Variablest hat store informa
sipt_destination($rU, 31, 4);
sipt_set_calling($fU, 4, 0, 3);
msg_apply_changes();
before the record_route command, and when using sipt_destination($rU,
31, 4); i'm getting asegfault
here you have the logs
could you please give me any clue about something i'm missing or doing
forward call indicators,
so i hope that does give us many problems
i will let you know anything
thanks and best regards
david
El lun, 01-08-2016 a las 12:09 +0200, Daniel-Constantin Mierla escribió:
> Hello,
>
> I am not familiar with isup and no testbed around at this moment. That&
/sbin/kamailio[4190]: WARNING:
[msg_translator.c:1958]: build_req_buf_from_sip_req(): check_boundaries
error
but i think there is already another query with this so i will check
there
best regards
david
El mar, 02-08-2016 a las 09:22 +0200, Daniel-Constantin Mierla escribió:
> Hello,
>
&
on/sdp.
Content-Disposition: session.
Content-Length: 104.
P-Asserted-Identity: "10707334" .
.
v=0.
o=user1 53655765 23536 IN IP4 79.170.68.171.
s=-.
c=IN IP4 2.2.2.2.
t=0 0.
m=audio 6001 RTP/AVP 8.
do you have any idea why these messages a
ge...
> Without it you cant generate correct ISUP part..
>
> Look in wireshark. It will inform you about any error in ISUP.
>
> --
> Best regards,
> Sergey Basov e-mail: sergey.v.ba...@gmail.com
>
> tel: (+38067) 403-62-54
>
>
> 2016-08-
07]
could it be posible that having manye ESL sockets opened in the kamailio
server makes this to happen?
how many async workers are needed? i have 8 right now
is there a way where i can know or correlate the log to a $ci or
something known?
best reg
conn:bgapi it
waits to receive an event) and having issue to do the t_suspend and
t_continue.
best regards
david
El jue, 06-10-2016 a las 12:01 +0200, Daniel-Constantin Mierla escribió:
> Hello,
>
>
> On 04/10/16 13:57, david wrote:
> > Hello all
> >
> > i'
msg_apply_changes();
sipt_destination($rU, 31, 4);
sipt_set_calling($fU, 4, 0, 3);
msg_apply_changes();
}
am i doing something wrong? how can i set this to end the SDP with last
boundary?
thanks a lot and regards
david
___
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(.*)/\1--unique-boundary-1--\2/sg});
set_body("$var()","multipart/mixed;boundary=\"unique-boundary-1
\"");
}
this way at least we have a base ISUP part for a 1XX response
best regards
david
El lun, 07-11-2016 a las 23:08 +0200, Sergey Basov escribió:
here, and we dont know what to look for :)
could you please give us some guidance or clue to at least start to
searching for something that can cause this issue?
thanks alot and regards
david
mem_dump_empty.gz
Description: application/gzip
mem_dump_load.gz
Description: applic
hello DAniel
we have 4.4.1 sorry
we are only handling SIP calls without REGISTER messages
best regards
david
El vie, 17-02-2017 a las 13:53 +0100, Daniel-Constantin Mierla escribió:
> Hello,
>
> what kamailio version?
>
> Are registrations handled by kamailio, or is just
pleased to provide if possible
best regards
david
El vie, 17-02-2017 a las 14:17 +0100, david escribió:
> hello DAniel
>
> we have 4.4.1 sorry
> we are only handling SIP calls without REGISTER messages
>
> best regards
> david
>
> El vie, 17-02-2017 a las 13:53 +010
variable);
seems we are not suffering this leak, or at least is much much more slow
we have tested this with kamailio 4.4.1, 4.4.5 and 5.0.0-rc1 versions,
and seems the same.
do you think this has any sense or could be explained?
thanks a lot and regards
david
El vie, 17-02-2017 a las 15:49
Yes set max-contacts in usrloc module
On Mar 2, 2012 5:45 AM, "Reda Aouad" wrote:
> Hi,
>
> Is there a way to ensure single-registration per user-agent for a user,
> which overwrites previous registration ?
> Or is there a way to limit the number of registrations per user, but
> overwriting the e
You can use IP auth its simple and works.
On Mar 8, 2012 4:19 PM, "romon.zaman" wrote:
> hello room,
>
> i was trying to add sip provider(like. flowroute,vitelity) with user-pass
> authentication in kamailio to accept inbound calls.
>
> any help?
>
> thanks
>
> _
e to use a
seasoned developer on the list.
I am sure they can respect your budget but equally respect their time and
contribution.
On Jun 20, 2012 6:13 PM, "copycall" wrote:
> david,
>
> invariably, this happens on most open source projects and lists.
>
> a few of developer
Sorry Daniel. I didn't see your message until I replied.
Understood.
On to 3.3...
On Jun 20, 2012 6:27 PM, "David J" wrote:
> Dave.
>
> Understandably. But my point was missed if you think that anyone here is
> trying to monopolize on the list please do u
In a UAC-Kamailio-UAS scenario, we've found a case where the ACK coming
from uac is not relayed by our proxy to the uas. This is the log for the
ACK message:
Jul 27 10:04:59 theseus-test /usr/local/kamailio/sbin/kamailio[17358]:
DEBUG: [parser/msg_parser.c:624]: SIP Request:
Jul 27 10:04
thing is wrong with
> it.
>
> Cheers,
> Daniel
>
>
> On 7/30/12 8:53 AM, David Notivol wrote:
>
> In a UAC-Kamailio-UAS scenario, we've found a case where the ACK coming
> from uac is not relayed by our proxy to the uas. This is the log for the
> ACK message:
&
This is the code that's being executed:
route[WITHINDLG] {
if (has_totag()) {
# sequential request withing a dialog should
# take the path determined by record-routing
xlog("ESTAMOS EN WITHIN\n");
if (loose_route()){
xlog("LOOSE ROUTE DETECTED\n");
:630]: method:
Jul 30 09:15:00 theseus-test /usr/local/kamailio/sbin/kamailio[1577]:
DEBUG: [parser/msg_parser.c:632]: uri:
Regards,
David.
2012/7/30 Daniel-Constantin Mierla
> Hello,
>
> the log message shows that the transaction is not found. Is the ACK coming
> late
Hello,
We found out the problem, it was out fault. The ACK was dropped because the
was a previous (and incorrect) evaluation of $rU. Being $null for these
calls, the ACK wasn't relayed. We already fixed it; thanks for the
assistance.
Regards,
David.
2012/7/30 Daniel-Constantin Mierla
>
invalid;branch=z9hG4bK8cie6bGSG0eOGh7Ne08Ro4CS1hI0oJ;rport=50906;received=10.1.2.229
From: sip:101@10.1.20.40;tag=3U0osO4h3h2bgCrvnCqO
To: sip:102@10.1.20.40;tag=9de7b31b15b69da019f867d4866ff286.b1f7
Call-ID: 9KoKnFdh285k2jg4
CSeq: 3 INVITE
Proxy-Authenticate: Digest realm="10.1.20.40",
se to check?
I'm blocked since last week.
Thanks a lot.
Kind Regards,
--
--DAVID--
2012/8/21 Peter Dunkley
>
> Hello,
>
> This does look like an issue with authentication rather than WebSockets.
> Have you tried using an ordinary SIP client (for example, Ji
2 params: INVITE:sip:10.1.20.40
HA2: 793dd6fa2e181e25a226cc09efc6dc2c
Sol params:
1d578520f175df632f0850415e603029:UDOU+1Azk89QyTccXdYGLoCTgF7+rIGv:793dd6fa2e181e25a226cc09efc6dc2c
Sol: 362206a07a3782c6e2557699abe5bf7b
Are those values correct?
Thank you very much.
Kind Regards,
--
--DAVID--
2012/8/21 Klaus
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