putting it as ^(00|\+)([1-9][0-9]+) and ^([5-9][0-9]{8}) to
designate start of the string.
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.
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module:
http://www.mail-archive.com/sr-...@lists.sip-router.org/msg00804.html
But I'm not sure what was the outcome.
Same for load_attrs() and user_attrs column - it looks like there's no
direct replacement for that and I have to migrate it to usr_preferences.
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Andrew Pogrebennyk
scenarios too (and I have no relation to the authors whatsoever:)).
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= [ / ] / / / : / / + / $
No .-* characters are allowed in the paramchar. But at least that's not
causing me any problems.
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On 03.02.2011 10:26, Andrew Pogrebennyk wrote:
I think the topoh module should force the angle brackets.
BTW it seems that parameter needs to be urlencoded, see rule
'other-param' in RFC 3261 section 25.1:
From what I understand the valid form is:
Contact:
sip:192.168.0.107;line=sr
into it.
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it for now due
to traveling. If you can test it and report back would be great. When
all is working fine, I will backport to 3.x branches.
Thanks,
Daniel
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, presence, dialog.
Server: kamailio (3.1.3 (i386/linux)).
Content-Length: 0.
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:5060;transport=udp;+sip.instance=urn:uuid:--1000-8000-00085D10A1A6.
Event: message-summary.
Expires: 86400.
Supported: path.
User-Agent: Aastra 57i/3.2.1.43 http://3.2.1.43.
Content-Length: 0.
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,$avp(s:cnt));
xlog(L_INFO, - currently, the carrier $rd has
$avp(s:cnt) active outgoing calls\n);
set_dlg_profile(carrierout,$rd);
}
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,
Andrew Pogrebennyk
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interface?
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Mokhtar, could you please make sure that you are calling route(RTPPROXY)
in reply route as well, as Alex suggested?
On 03.06.2011 16:33, Mokhtar Bengana wrote:
This is how I configured rtpproxy. Not sure why rtpproxy is not
engaged both ways. Thanks for your help.
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QUERY=select * FROM $DIALPLAN_TABLE ORDER BY
$DIALPLAN_DPID_COLUMN, $DIALPLAN_PR_COLUMN;
fi
$DBROCMD $QUERY
;;
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that
calls env -i (start with an empty environment), maybe this is the reason
it works differently for the root and kamailio users.
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sr-users
on Kamailio how could i go about doing it?
thanking you in advance
Phillip
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On 28.06.2011 11:18, Raúl Alexis Betancor Santana wrote:
This is not fully true on the DTMF side, if you use SIP-INFO as DMTF
transport, DTMF's will go throught kamailio (if configured to do so).
Right - but then this is achieved through the configuration script.
Andrew
advertised_port = sip
both define after the line of listen=public_ip
please advice.
thanks.
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(,$avp(s:allowed_cli));
xlog(L_INFO, $ci : replaced from to $avp(s:allowed_cli)\n);
}
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, no luck..
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() before
the call to t_set_fr(), still no luck.
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On 09/02/2011 08:58 PM, David Zambrano wrote:
After doing some traces
on the network I realized that the transcoder is trying to reach the
router in front of the softphone and skipping the Kamailio
loadbalancer. The call never reaches the softphone so the phonecall
never completes. In UDP
On 09/02/2011 10:33 PM, David Zambrano wrote:
Hi andrew. Thanks for your help. What module or config should I use to
make sure the connection goes back through the loadbalancer?
That's simply the task for record-route like:
if (is_method(INVITE))
record_route();
But you also need the
On 09/26/2011 02:16 PM, Mino Haluz wrote:
Ok, the problem is the original feeradius package (squeeze) does not
support Update radius messages (it gives Unsupported Acct-Status-Type =
15).
It's not related to kamailio.
.. however anybody knows how to patch it?
The patches are here:
Jeff,
On 10/03/2011 08:45 PM, Jeff Anderson wrote:
I am trying to get kamailio sip-capture up and running for use with homer. I
can get the service to start but i receive the following error.
[root@homer kamailio]# /usr/local/sbin/kamailio -c
loading modules under
Austin,
Actually you could share your config to the mailing list, I will tell
you if there is something blatantly wrong ;-)
Regards,
Andrew
On 11/03/2011 06:15 AM, Jason Penton wrote:
Hi Austin,
Have a look at the TM module docs. You will find the appropriate
commands there.
HTH
On
Austin,
the block beginning with dlg_manage() should be placed between
route(AUTH) and route(PSTN). It is commented out and put outside the
main routing block so I'm not sure if that was the case..
If the problem persists please get the dialogs list before calling,
while the callee's phone is
Hi Bruno,
What I have done is an explicit check
if (uri_param(transport,tls) || uri_param(transport,TLS))
to call force_send_socket with either udp or tls port.
It would be cool for kamailio to select the proper socket automatically,
I think there was a discussion on that previously but I can't
Bruno,
the address from contact header is put into R-URI on outgoing request to
that user. This is where I catch that parameter. I think we should debug
why kamailio sends the request using UDP, it is not clear, as Daniel
pointed out it should work automatically. I think I had to do these
Gnani,
since version 3.2.0 kamailio accepts several arguments to
record_route_preset() function:
http://www.kamailio.org/docs/modules/3.2.x/modules_k/rr.html#id2542169
In 3.1.0 you can call insert_hf directly, but IMO it's worth upgrading.
Andrew
On 12/08/2011 08:50 PM, Gnaneshwar Gatla wrote:
On 12/23/2011 08:18 PM, Stefan Sayer wrote:
shouldn't the db layer and driver be smart enough to do insert ... on
duplicate key update at least where it's supported?
my fear is that such first insert then update policy will affect the
performance. can create noise in the log on some db backends
On 12/29/2011 03:36 PM, Robert R wrote:
Thank a lot. $T_reply_code works.
I tried all variables in pv doc ($rc, $err.rcode, $rs ... ) and none of
them works,
Actually the $rs pseudo-variable should also work as described here:
On 12/29/2011 03:21 PM, Robert R wrote:
What is the Pseudo-variables for Status-Line filed of SIP response
messages (2xx, 3xx,4xx,5xx,6xx)? i.e., is there a Pseudo-variables to
display the SIP response code?
You can test it with tm function t_check_status(...) or get it in a
config variable
Hello,
assuming that I want to use contact from 302 response as outbound proxy
but keep the original Request-URI, what should I do? Calling the
revert_uri() after get_redirects() in failure_route doesn't do the trick.
___
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Daniel, thank you - method (2) works like a charm.
On 01/25/2012 03:16 PM, Daniel-Constantin Mierla wrote:
an option - get the contact header from 302 in failure route via
$T_rpl($ct). Use its uri to set $du.
I was getting an error for some reason:
0(7860) ERROR: core [pvapi.c:516]: error
On 01/25/2012 03:18 PM, Alex Balashov wrote:
Put the 302 Contact URI in the destination set instead, i.e. $du, not
$ru. That will cause it to be relayed to the redirect destination on
the network and transport level, but the logical target will remain the
same.
Ok. I was thinking that
On 01/26/2012 08:21 PM, Krishna Kurapati wrote:
Is there a configuration option to let kamailio use Public IP when
setting record-route in 200 OK?
Of course, you need record_route_preset() - see
http://kamailio.org/docs/modules/3.1.x/modules_k/rr.html#id2667590
Also I would expect you need to
Since this thread will probably end up in Google I''ll share my
experience. I ended up with this
if(t_check_status(301|302))
{
#NOTE: must assign to $du to keep R-URI intact
$var(contact) = $T_rpl($ct);
$var(contact) =
Hi,
Was reading through the xmlrpc docs, could anybody with an access check
why Figure RPC Example is missing from
http://kamailio.org/docs/modules/devel/modules/xmlrpc.html#fig.rpc_example
as well as 3.1 and 3.2 module docs?
Thanks,
Andrew
___
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Hi Daniel,
On 02/17/2012 10:47 AM, Daniel-Constantin Mierla wrote:
I made a patch for server reconnect -- I had no access to a computer
with redis lib installed for the moment, hopefully it compiles. If you
can try and tell the result, it would be great, I can commit then.
I may be able to
Hello,
we have a Kamailio proxy which gets the call from PSTN gw and does some
call forwarding (serial forking) to several destinations through our sbc
The call flow I am looking at is:
- Kamailio sends INVITE to branch_1.
- branch_1 sends 180 with to-tag*, proxy relays it to the gw
* 180 meets
On 03/07/2012 07:37 PM, Lucas Alvarez wrote:
I want rewrite $tU but I'm not being able, I'm doing the following:
remove_hf(To);
insert_hf(To: sip:$rU@$rd\r\n, From);
Then I'm printing $tU and it is still having the previous value, any
help will be appreciated.
Thanks in advance.
Did
On 03/07/2012 08:46 PM, Lucas Alvarez wrote:.
Something like this:
if(($rU=~^(box02)[0-9]{2,15}$)) {
$rU = $(rU{s.substr,5,0});
$ru = sip: + $rU + @ + $sel(cfg_get.box02.gw_ip) + :
+ $sel(cfg_get.box02.gw_port);
}
BTW alternatively you can use dialplan the
Hello Klaus,
please check this thread:
http://www.mail-archive.com/sip-implementors@lists.cs.columbia.edu/msg10537.html
On 03/14/2012 05:11 PM, Klaus Darilion wrote:
I wonder if a presence server may send a NOTIFY if the previous NOTIFY
in the dialog did not received an answer yet. I greped the
Dan,
Well, it looks like kamailio recognizes 127.0.0.1 as its own URI that's
why it does rewrite. Do you have by chance such alias in your config (or
auto_aliases=yes)?
On 03/26/2012 03:47 PM, Dan-Cristian Bogos wrote:
Hey Guys,
I have noticed some unexpected behavior (at least by me) during
I've been pondering an issue with Route header parameters being not
mirrored by kamailio proxy into Record-Route field on in-dialog requests
over the last few hours so I thought I'd just whether I'm missing
something obvious.
The call scenario in UA - lb - proxy - sems. Everything is fine with
Hi,
despite my initial doubt it works well :-) Thank you Daniel.
On 04/04/2012 05:51 PM, Daniel-Constantin Mierla wrote:
Hello,
is record_route() executed as well as add_rr_param() for reinvites?
Cannot spot in the logs. You can load debugger module and enable
cfgtrace to see what actions
Hi Daniel,
See RFC 3261 section 12.2.2:
If the remote sequence number was not empty, but the sequence number
of the request is lower than the remote sequence number, the request
is out of order and MUST be rejected with a 500 (Server Internal
Error) response.
However, 400 or some 4xx
Hello,
It is already there, see
http://www.kamailio.org/dokuwiki/doku.php/core-cookbook:3.1.x
On 05/09/2012 06:04 PM, Konstantin M. wrote:
Hi,
I would like (and a many people here I believe) to have a functional of
including a multiple config files like (foe example asterisk's
#include
Konstantin,
You should put the include_file directive after loadmodule and modparam
directives. So it can be either before main route block or at the bottom
of your main kamailio.cfg.
On 05/09/2012 06:48 PM, Konstantin M. wrote:
After including a part of main config to included file -- I got a
Hi,
On 05/30/2012 12:22 PM, Aft nix wrote:
So i'm interested if RFC 3261 provides any mechanism by which we can
differentiate a BYE whether its from caller or callee.
Check out is_direction() function:
http://www.kamailio.org/docs/modules/3.2.x/modules_k/rr.html#id2527009
The papers talk about transport protocol for signaling, not media/RTP.
I didn't hear of anyone who does RTP over TCP neither. I doubt even that
the performance is a primary reason behind that, for media over TCP the
client link must be virtually packet-loss free (due to TCP
retransmissions), while
On 06/19/2012 02:06 PM, Uri Shacked wrote:
I am testing kamailio replies when an INVITE or another request arrives
with lets say, VIA header missing
The core drops the request. But, there is no reply for the originator
(so it keep on resending the request...)
Why?
If there was no Via
Hi Gary,
Yes, that's an old/known issue. You can convert the IP to common format
with a simple command:
perl -MSocket -le 'print inet_ntoa(pack(N, 1273060816))'
Maybe somebody from the kamailio team will take time to fix this.
On 07/24/2012 09:39 PM, Gary Chen wrote:
Kamailio 3.2.0
When
Hi Gary,
It was fixed already by Richard in 3.3 branch:
http://git.sip-router.org/cgi-bin/gitweb.cgi/sip-router/?a=commit;h=28be16549831df46dd1b8312da223b02359d8a9c
(and master)
Thank you for the report.
On 07/24/2012 09:44 PM, Gary Chen wrote:
Sorry, it should be Kamailio 3.3.0 not 3.2.0.
At least some links at http://www.kamailio.org/wiki/ don't work at the
moment:
Install Kamailio v3.3.x From GIT
Upgrade Kamailio v3.1.x to v3.2.0
Upgrade Kamailio v3.2.x to v3.3.0
you are redirected to some instruction from DokuWiki Installer when
trying to visit them. Could somebody
Mino, I am not sure, but you could try the following:
set failure_reply_mode 3
(http://kamailio.org/docs/modules/stable/modules/tm.html#failure_reply_mode),
then handle 302 redirect in the proxy and use the contact as a new
branch like this:
if(status == 302)
{
$var(contact) = $ct;
Same here, also you can use protoshoot provided by kamailio, see
utils/protoshoot.
On 08/13/2012 07:45 AM, Mark Anthony Delfin wrote:
Hi Anton,
Previously I used the following.
sipsak
http://sipsak.org/
or
sipp
http://sipp.sourceforge.net/
Regards,
Mark
You might need to upgrade to 3.3.1 to use sdpops module:
http://kamailio.org/docs/modules/stable/modules/sdpops.html
On 08/13/2012 09:52 AM, phillman25 wrote:
Dear List
I am trying to remove specific lines from the following original SDP body:
Content-Type: application/sdp
Hello Spencer,
actually double quotes are not allowed in URI parameter. In the BNF
grammar the allowed chars in the unreserved definition are alphanum
and mark, where mark is only - / _ / . / ! / ~
/ * / ' / ( / ) ).
This is already fixed in 3.3.0 if I am not mistaken, please check
On 09/18/2012 12:15 PM, Gary Shergill wrote:
Note that I am testing this with one computer connected by Bria and
another computer connected via Blink. I am able to log on to a user on
each (test1 and test2) and they are able to call each other. The issue
is, with presence enabled, they are
Hi,
I have found recently that in order to detect retransmits I have to
create a transaction explicitly when the request comes in:
force_rport();
if(!t_check_trans())
t_newtran();
sl_send_reply(100, Trying);
xlog(L_INFO, New request - $ci\n);
it
/docs/modules/3.3.x/modules_k/tmx.html#id2543767
Cheers,
Daniel
On 9/26/12 3:30 PM, Andrew Pogrebennyk wrote:
Hi,
I have found recently that in order to detect retransmits I have to
create a transaction explicitly when the request comes in:
force_rport
, Daniel-Constantin Mierla wrote:
Hello,
you can try after you set the flag you wanted to be in transaction, to
be sure it gets there.
Cheers,
Daniel
On 9/26/12 7:43 PM, Andrew Pogrebennyk wrote:
Hi Daniel,
No, I don't. Thanks for the tip. Could you advice where t_flush_flags()
should
Kamal,
perhaps RFC 5763 provides you some of the answers?
On 10/16/2012 11:06 AM, Kamal Palei wrote:
Hi Johansson, All
Sincier regards and thanks for input.
As I understand, all media packets pass through RTP Proxy. The RTP
Proxy will receive simple UDP media packets from endpoints. Next
Richard,
well, the RURI of remote ACK has proxy IP address 10.200.70.100 so proxy
thinks that previous hop was a strict router. I can't think of any
workaround that would not be an ugly hack at the moment, though.
On 11/23/2012 03:24 AM, Richard Brady wrote:
Hi guys
I have a multihomed
Hello,
I'm getting some errors - if I restart kamailio on a live system, the
first call after restart fails with max nr of branches exceeded.
However, we are not doing append_branch() or anything fancy for that
call at all:
Jan 3 15:26:54 sp1 /usr/sbin/kamailio[8078]: ERROR: core
[dset.c:306]:
Some Fritzbox devices don't accept the OPTIONS ping generated by
kamailio with RURI: sip:11.22.3.4:5060. In the location table we have:
received: sip:11.22.3.4:5060
contact: sip:user@11.22.3.4:5060;uniq=6633BC1386F4D4CC4EBD64DC7E967
path: sip:lb@127.0.0.1;lr;received='sip:11.22.3.4:5060'
Kamailio
On 01/07/2013 01:29 PM, Andrew Pogrebennyk wrote:
What I don't understand is why kamailio sets RURI of the OPTIONS message
to value of received instead of the contact. I suspect a bug in the
parser somewhere along these lines:
rval =
ul.get_all_ucontacts(buf,cblen
Hello,
try printing the $ru with xlog statement when the request comes into the
server and in the beginning of route[LCR].
On 11/01/2013 01:50, Douglas Ugalde wrote:
Hi,
Im trying to configure LCR in Kamailio 3.3.3 but I dont Know how can I
do to fix this error:
ERROR: lcr
Hello,
I see no problem with an ACK in the trace. The t38modem ACK's the 407,
sends the new INVITE with authorization but it's not accepted by
kamailio. I'm not sure if t38modem can even perform the authorization.
Maybe you need to accept the calls from it without authorization. But if
it can and
Carsten,
On 01/29/2013 08:25 PM, Carsten Maass wrote:
Or does it mean, the authentication is rejected because the local part
030123456789 in URI does not match the subscriber 979?
Indeed, this is what happens when you call auth_check with flag 1:
if (!auth_check($fd, subscriber, 1)) {..}
Hello,
Usually this happens when the RURI has the IP address/ domain which
points to this server or which is present in the domain table. There are
some clients which may put even server IP address into Contact of INVITE
or 200 OK instead of its own so the further requests will be mis-routed.
You
Hi Juha,
On 04/07/2013 01:51 PM, Juha Heinanen wrote:
i escaped them, but it didn't help. path header now looks like:
Path:
sip:192.98.102.10;transport=tcp;lr;received='sip:192.98.102.10:58156%3Btransport%3Dtcp'.
and i still get the same error:
Apr 7 14:49:47 wheezy1
On 04/09/2013 11:59 AM, Juha Heinanen wrote:
because path-value starts with name-addr and my interpretation is that
since there are s around this path header body:
Path:
sip:192.98.102.10;transport=tcp;lr;received='sip:192.98.102.10:58156%3Btransport%3Dtcp'
solely consists of name-addr
Dear 2.8 users,
this update includes the fixes for kamailio-lb config (multiple
sockets-related):
- added record-route param to store socket; review logic for relaying
in-dialog requests (fallback to default send socket if it's not defined
explicitly); fixed socket selection for ACK after 4xx and
Hi,
On 10/11/2013 08:14 PM, Fred Posner wrote:
I use rwie and rwei flags but in ngcp-mediaproxy-ng e and i
seems to be used for IPv4 / IPv6 ..
...
I don't believe that mediaproxy-ng can be used to bridge two ipv4
networks; only bridging for ipv6 - ipv4.
That's true, there's no bridge
Hi,
You should create a branch_route and perform this manipulation there..
BR,
Andrew
On 01/14/2014 10:53 AM, Igor Potjevlesch wrote:
Hello,
No one has an idea? I was thinking that each request goes to RELAY but
even if I try to modify the R-URI in this route, it fails. I still had
Hi,
check if this answers your question: http://kb.asipto.com/asterisk:index
Andrew
On 01/13/2014 04:19 PM, Kasinath wrote:
Hi All,
I just installed Kamailio in one server and Asterisk in another.
Asterisk loads it sipusers info from database which is in Kamailio server.
I don't know how
On 01/23/2014 05:12 PM, Klaus Darilion wrote:
It is necessary to use the cwie / cwei flags in the rtpproxy_manage call?
If rtpproxy uses only a single listen-IP, then these flags are not
needed. Only if you operate rtpproxy in bridge mode, then you need these
flags. Bridge mode is necessary
Hi,
could you please post also your Chrome js developer log?
Does the problem exist if you start the jssip clients without video support?
Andrew
On 02/03/2014 12:00 PM, Mihai Marin wrote:
Hello,
Another weekend struggling to make a call from jssip to another jssip
behind firewall and I
Hi,
See presence_reginfo - Extension to Presence server for registration
info replication (RFC3680)
http://kamailio.org/docs/modules/stable/modules/presence_reginfo.html
On 02/05/2014 07:56 AM, Premchandiran wrote:
Hi All,
May I know whether kamailio supports event header with reg
On 02/05/2014 11:20 AM, Daniel Tryba wrote:
Off to figure out why uac_test incorrectly flags the onhold process as
needing
the nat helper...
Well, nat_uac_test is pretty straightforward, it does what the flags
tell it to do:
Sorry for reviving this old thread. I think I've come across the same
issue as Jayesh and it is triggered by the Route
inserted by sipml5 into in-dialog requests like ACK - Route:
sip:172.18.101.48:5060;lr;sipml5-outbound;transport=udp
Jayesh had two Route headers that belong to this kamailio
On 03/07/14 16:42, Yuriy Gorlichenko wrote:
thanks for fas reply. If I may user rtpengine as rtpproxy maybe you
already use it or just know - does rtpengine provide ridge mode as
rtpproxy between internal and external interfaces? At my instalne if I add
rtpengine
Hi,
let's say I'm running two proxies/registrars that need to access Shared
location DB in db_mode=1 (all changes to usrloc are immediately
reflected in database too). I have observed that if the UAC re-registers
before the previous registration's expiry and the new REGISTER reaches
the other
Hi,
OK, thanks for the clarification. Maybe we will check the
load_db_contacts(userid) way with Victor.
Regards,
Andrew
On 09/15/2014 12:04 PM, Daniel-Constantin Mierla wrote:
Hello,
On 12/09/14 15:42, Andrew Pogrebennyk wrote:
Hi,
let's say I'm running two proxies/registrars that need
Hi,
we are filtering some method names from the Allow header with kamailio
4.1: depending in the configuration:
if (hf_value_exists(Allow, INFO))
{
xlog(L_INFO, Remove INFO from Allow\n);
exclude_hf_value(Allow, INFO);
}
if
See http://www.kamailio.org/wiki/cookbooks/4.2.x/pseudovariables#tv_name
On 10/22/2014 10:40 AM, Grant Bagdasarian wrote:
Hello,
I’m currently using the $TS psuedovariable to get the current unix
timestamp, but this only returns the timestamp up to a second precision.
Is it possible
Daniel,
I'm reviving this old thread..
On 05/27/2015 01:40 PM, Andrew Pogrebennyk wrote:
thanks for the answer, that's what I was thinking - maybe the flags do
not persist or are destroyed after the per-branch failure route.
However, the t_flush_flags description says this function can
Hi Daniel,
thanks for the answer, that's what I was thinking - maybe the flags do
not persist or are destroyed after the per-branch failure route.
However, the t_flush_flags description says this function can be used in
any route, so in should be fixed in the long term.. Let me check if I
can
Hi Daniel and others,
I'm having a problem with acc module if I'm using the event_route/
branch-failure:
say, the call comes from the app server and goes to the registered user.
We arm the the failure route and per-branch failure route for the 302
redirect from the UA. We explicitly reset the
Hi,
I was trying to use registered("location", "$ru", 0, 1)
Last parameter is the flag according to
http://kamailio.org/docs/modules/stable/modules/registrar.html#registrar.f.registered
flag values is as follows:
1 - set xavp_rcd with value from matched contact
But I'm getting NULL instead
Hi,
I have a similar issue with kamailio 4.3.4. I want to append a header to
external 486 or 603 reply. If I got it right I should call append_hf,
not append_to_reply which is for locally generated replies.
I've added append_hf() and in the end of failure route call exit but for
some reason that
Hello Efelin,
I stumbled upon the issue you described here. Have you been able to find
a solution? I've tried to play with t_reply(), but no luck so far.
Regards,
Andrew
On 11/18/2013 10:00 AM, Efelin Novak wrote:
> Hi,
>
> I would like to append a header to a 'winning' negative reply in
>
Thanks Dmitri, I have to look closer at it, but for me it breaks
parallel forking, e.g. I need to reply only when it's the last branch.
Andrew
On 12/22/2015 05:46 PM, Dmitri Savolainen wrote:
> Andrew, I use smth like this for adding header to any response
>
> request_route{
>
> if
Alex Balashov wrote:
> Strictly speaking, CANCEL is a different request, and accordingly, a
> different transaction.
>
> However, you should be able to access INVITE transaction data from the
> failure_route triggered in connection with transaction cancellation.
Thanks, Alex!
Andrew
Per my understanding the uac module stores the "vsf" parameter in
Record-Route and should be able to update the From/To URIs automatically
in all in-dialog requests that carry this parameter.
http://kamailio.org/docs/modules/stable/modules/uac.html#uac.p.restore_mode
“auto” - all sequential
Hi all,
it seems that the AVPs not available when when processing CANCEL
message, even though they have been set for this transaction initially.
Is this the expected behavior?
P.S. kamailio 4.3.4
Andrew
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