much traffic)?
Use debug=3 in kamailio.cfg and send to me all the logs from kamailio
start. I will look to see what happens.
Cheers,
Daniel
On 05/02/15 22:21, SamyGo wrote:
Hi community,
I'm dealing with a problem here related to presence module handling BLF.
My BLF phones are Yealink
Here are the contents of the presentity table for the test call.
On Thu, Feb 5, 2015 at 5:54 PM, SamyGo govoi...@gmail.com wrote:
Hi Daniel,
Thanks alot for your time, please see the log file attached.
If needed, I can provide sip captures received at the phone .
Thanks,
On Thu, Feb 5
Hey there,
You have to make sure that the same fifo filename exists in the kamctlrc
file too. This file is located in the same directory as of kamailio.cfg
On Fri, Oct 31, 2014 at 10:37 PM, Mahmoud Ramadan Ali
cisco.and.more.b...@gmail.com wrote:
Hi Everyone
I have encountered that error
Hey Errol,
What I get from Daniel's email is that you only need to add this route in
your script and it will trigger itself automatically when uac_req_send()
function is executed.
event_route [tm:local-request] { # Handle locally generated requests
xlog(L_INFO, Routing locally generated
error in config file
/etc/kamailio/kamailio.cfg, line 1034, column 31: bad command
0(44073) : core [cfg.y:3570]: yyerror_at(): parse error in config file
/etc/kamailio/kamailio.cfg, line 1034, column 33: bad command
regards,
Errol
On Tue, Oct 7, 2014 at 9:28 PM, SamyGo govoi...@gmail.com
Thanks Vitaliy,
That seems working, I am testing with it..it's great.
Can I ask if there is any psuedo variable for this thing !
Best Regards,
Sammy
On Sat, Aug 24, 2013 at 3:10 PM, Vitaliy Aleksandrov vitalik.v...@gmail.com
wrote:
How about next config snippet:
*route[GET_NEXT_HOP]**
Yes indeed it is everything that I wanted to get. Thanks alot for the quick
route.
On Sun, Aug 25, 2013 at 10:13 PM, Vitaliy Aleksandrov
vitalik.v...@gmail.com wrote:
AFAIK there is no any single pv that provides a destination for both sip
message types (request/reply).
As you can see from
Dear list,
Its just a little thing to ask but I somehow can't seem to figure out which
psuedo variable to use to find out where a particular 200 OK is destined to
go.
the scenario is I've multiple media servers in dispatcher and calls are
distributed. I need to capture a 200OK that is destined
through Asterisk ?
Thanks in advance.
Regards,
Nishar Hamsa
On Sat, Aug 17, 2013 at 1:06 PM, SamyGo govoi...@gmail.com wrote:
Check regular expression as the logs state. Modify with correct one.
sent from a little smart phone
On Aug 17, 2013 2:03 PM, Nishar M.H nisharm...@gmail.com
created in kamailio
server* ?
*
*
*
*Regards,
*
*Nishar Hamsa.
*
*
*
*
*
On Sun, Aug 18, 2013 at 1:39 PM, SamyGo govoi...@gmail.com wrote:
Since you've removed the second regex then only way to make call reach
to Asterisk via PSTN route is when you dial anything that matches
Dear Nishar,
From the mentioned URL the configurations need the following changes.
On the top start on configs you need to use: WITH_PSTN and might not need
WITH_ASTERISK
#!KAMAILIO
#!define WITH_MYSQL#!define WITH_AUTH#!define WITH_USRLOCDB#!define WITH_PSTN
The insert your asterisk IP in
:29 AM, SamyGo govoi...@gmail.com wrote:
Dear Nishar,
From the mentioned URL the configurations need the following changes.
On the top start on configs you need to use: WITH_PSTN and might not need
WITH_ASTERISK
#!KAMAILIO
#!define WITH_MYSQL#!define WITH_AUTH#!define WITH_USRLOCDB
Dear Khoa,
As Daniel stated you need to see if your SIP phones are able to sense the
change in its network parameters and trigger a Re-INVITE to Kamailio with
new SDP to handle the audio.
That's very important to do because once RTPproxy allocates the ports it
can't just start sending RTPs to
.
*/
On Thu, Aug 15, 2013 at 1:22 PM, SamyGo govoi...@gmail.com wrote:
Dear Khoa,
As Daniel stated you need to see if your SIP phones are able to sense the
change in its network parameters and trigger a Re-INVITE to Kamailio with
new SDP to handle the audio.
That's very important to do because once
Hi Nishar,
Once you've separate Asterisk and Kamailio installation you'll need to
modify the variables and DB parameters in kamailio.cfg. Simply copying the
configurations file may give you errors. Please see the log files syslog
or messages according to your OS and see why starting of kamailio
the Asterisk communicate with
Kamailio SIP users.
Do i have to create SIP trunk in Asterisk ?
Regards,
Nishar Hamsa.
On Thu, Aug 15, 2013 at 3:49 PM, SamyGo govoi...@gmail.com wrote:
Hi Nishar,
Once you've separate Asterisk and Kamailio installation you'll need to
modify the variables
Dear Alexandr,
You can connect Kamailio to RTPproxy via socket as well, use modparam like
this:
modparam(rtpproxy, rtpproxy_sock, udp:127.0.0.1:12221)
Then if your rtprpoxy is started in bridged mode you should use the i and
e flags while you call the rtpproxy-manage() function in the
016960,
len 000160)
Got RTP packet from2.2.2.2:42346 (type 00, seq 000987, ts 017120,
len 000160)
Sent RTP packet to 1.1.1.1:63232 (type 00, seq 003839, ts 017120,
len 000160)
But no voicing)
2013/8/6 SamyGo govoi...@gmail.com
Dear Alexandr,
You can connect Kamailio
Hi again,
Still Missing 200OK for this call. It'll be helpful to send a complete
trace for the call coming in to the Asterisk at first place and then
Dialing out to the B-leg whose trace which you've just shared.
On Tue, Aug 6, 2013 at 3:22 AM, Alexandr Usov blessen...@gmail.com wrote:
Please check the rtpproxy function and paste the way it is written in your
configuration file. Share the output of ps -ef | grep rtpproxy and
netstat -pln|grep rtpproxy
--
Sammy
On Tue, Aug 6, 2013 at 3:55 AM, Alexandr Usov blessen...@gmail.com wrote:
Note:
Asrterisk LAN IP real
()) {
if(isbflagset(FLB_NATB)) {
fix_nated_contact();
}
}
#!endif
return;
}
2013/8/6 SamyGo govoi...@gmail.com
Please check the rtpproxy function and paste the way it is written in
your configuration file. Share the output of ps -ef | grep rtpproxy and
netstat
... pike would do, if you had it that way.
What does your kamailio.cfg look like? (offlist is fine if you don't want
to post it).
Skyler
On 6/10/2013 10:33 PM, SamyGo wrote:
Hi again,
So I've conducted test with Yealink IP Phone and the results are the
same, please see the attached
Hi List,
I've been trying to make the Presence thing work with kamailio but the very
basic presence doesn't seem to work. I've tried multiple modules and
different how-tos for running successful presence aware configuration but
seems something is missing. The ultimate goal is to give user
and not doing any PUBLISHes. As such, server presence isn't going to
work.
Regards,
Peter
--
Peter Dunkley
Technical Director
Crocodile RCS Ltd
On 10 Jun 2013, at 11:05, SamyGo govoi...@gmail.com wrote:
Hi List,
I've been trying to make the Presence thing work with kamailio
,
Daniel
On 6/10/13 12:55 PM, SamyGo wrote:
Thank you for the prompt responses, sure I'll try some other client as
I've a yealink IP phone accessible but this x-lite worked perfectly when
connected with Asterisk and gave me Presence info with reginfo+xml and
dialog+xml body types. This is new x
Hi again,
So I've conducted test with Yealink IP Phone and the results are the same,
please see the attached trace and suggest what should I do to have a
working presence model.
The same phenomenon with PUBLISH originated from Kamailio destined to
Kamailio is observed again.
Thanks for the
stuff now.
Vitaliy, please explain further how can I add the PUA component.
Thanks,
Sammy
On Thu, May 30, 2013 at 6:17 PM, Olle E. Johansson o...@edvina.net wrote:
30 maj 2013 kl. 13:07 skrev SamyGo govoi...@gmail.com:
Hi again,
I doubt SCA module will work for me but let me get
dialog_publish for URI
sip:4...@wistle.myvoipdomain.com:5060
Unfortunately I still don't see any lights blinking on my phones.
Any help here will be much appreciated.
BR,
Sammy
On Fri, May 31, 2013 at 11:10 AM, SamyGo govoi...@gmail.com wrote:
Hi again,
Yes Vitaliy, that's exactly where I
... increasing your
chance to get more people to an answer.
Cheers,
Daniel
On 5/28/13 11:41 PM, SamyGo wrote:
Yeah, I'm getting your point.
So now, here are my loaded modules.
#!ifdef WITH_PRESENCE
loadmodule presence.so
loadmodule presence_xml.so
loadmodule presence_dialoginfo.so
loadmodule
:17 PM, Olle E. Johansson o...@edvina.net wrote:
30 maj 2013 kl. 13:07 skrev SamyGo govoi...@gmail.com:
Hi again,
I doubt SCA module will work for me but let me get this a shot tonight and
get back with a sipgrep trace. Meanwhile I was just thinking if the 'skype
like service' blog
Hi list,
I've been trying to make my Yealink phone to give BLF indications but I
haven't been able to achieve this successfully yet so I need some expert
advise here.
My Yealink phone, as soon as it registers to Asterisk, gives me BLF lights.
The same phone registering to Kamailio sends SUBSCRIBE
Attached are the relevant sip traces.
On Wed, May 29, 2013 at 2:32 AM, SamyGo govoi...@gmail.com wrote:
Yeah, I'm getting your point.
So now, here are my loaded modules.
#!ifdef WITH_PRESENCE
loadmodule presence.so
loadmodule presence_xml.so
loadmodule presence_dialoginfo.so
loadmodule pua.so
Hi,
Its simple, just before t_relay() the signalling to destination B use
rtpproxy module's function manage_rtpproxy()
I hope you'll have it working.
Regards,
Sammy
On Feb 27, 2013 2:55 PM, Khoa Pham onmyway...@gmail.com wrote:
Hi,
Supposed A call B in a non-NAT environment. How to let
Hi Prakash,
Please paste All ERROR appearing and Warnings when you start your kamailio.
What is the free memory situation on your server ?
Regards,
Sammy
On Tue, Feb 12, 2013 at 5:12 PM, Prakash N prakas...@tevatel.com wrote:
Hi Muhammad,
Yes, I have define Mysql on top in
you have any clue about that ?
Regards,
Rumen
On 29 January 2013 08:20, SamyGo govoi...@gmail.com wrote:
Hi Rumen,
Can you tell how are you creating users in your sip table? Are you sure
that the passwords are calculated using the real/domain part of the SIP
User
definition
the person managing the list at
sr-users-ow...@lists.sip-router.org
When replying, please edit your Subject line so it is more specific
than Re: Contents of sr-users digest...
Today's Topics:
1. Re: Asterisk and dispatcher (SamyGo)
2. Re: Asterisk and dispatcher (SamyGo)
3
+1 Klaus - I used your tips and Kamailio is working great behind NAT. I'll
try to test different types of NATs and see if SIP and RTP works perfectly
for all scenarios.
Thanks and cheers
Sammy
On Fri, Jan 11, 2013 at 1:42 PM, Klaus Darilion
klaus.mailingli...@pernau.at wrote:
Am 11.01.2013
Hi,
You've metioned this already in your code:
if(ds_is_from_list(1))
Is this not working for you?
Regards,
Sammy
On Wed, Jan 23, 2013 at 10:44 AM, Ian French fre...@gmail.com wrote:
Hi,
I've been working my way through this tutorial (
Except that I find missing curly brackets for the if() condition which will
always return 1/positive.
On Wed, Jan 23, 2013 at 10:26 PM, SamyGo govoi...@gmail.com wrote:
Hi,
You've metioned this already in your code:
if(ds_is_from_list(1))
Is this not working for you?
Regards,
Sammy
Hi Rumen,
Im sorry to have missed the chance to give you extended details earlier. As
per my experience you've to do the following:
1. Load the domain.so module. Point the domain names (DNS URLs) to your
kamailio IP adress and add those domain names into the domain table.
2. Set the Registrar,
Hi,
I think you want asterisk help ! or is it related to iptel sip trunk ?
Anyway here are few simple answers.
1- For asterisk you can install a Digium or Sangoma Card to connect PSTN
lines to your server and make calls out to India using those PSTN lines.
OR
1- As you said you can buy a local
2012 07:00, SamyGo govoi...@gmail.com wrote:
Hi,
Just few quick questions , can you ping the second asterisk from
secondary
Kamailio ? what is the routing set for that. Does your second asterisk
box
gets inactive in dispatcher show command after 30 seconds ?
Mine worked fine, the key
:5060 flags=IP priority=1 attrs=
SET:: 1
URI:: sip:10.2.45.98:5060 flags=IP priority=1 attrs=
on both machines. Moreover now without any changes dispatcher is not
working at all. Only sending calls to the asterisk-1 (10.2.45.98)
On 28 November 2012 18:20, SamyGo govoi...@gmail.com
works like a charm, when I start kamailio 1 again the same
issue appears calls are only routed to the 10.2.45.103 asterisk
What might be the problem that they are in different set ?
On 28 November 2012 18:37, SamyGo govoi...@gmail.com wrote:
Well these are two different dispatcher
Hi,
Just few quick questions , can you ping the second asterisk from secondary
Kamailio ? what is the routing set for that. Does your second asterisk box
gets inactive in dispatcher show command after 30 seconds ?
Mine worked fine, the key is the linux routing table.
BR
Sammy
On Tue, Nov 27,
Hi,
Besides what Daniel has asked for , can you tell what does this mean ?
: ERROR: db_mysql [km_my_con.c:109]: driver error: *Can't connect to local
MySQL* server through socket '/var/run/mysqld/mysqld.sock' (2)
In my experience Kamailio starts and runs perfectly even with these
RTPproxy
Did you forget to answer
what is the command you use to start rtpproxy
?
On Fri, Oct 12, 2012 at 3:26 PM, శ్రీధర్ asridh...@gmail.com wrote:
Installed rtp proxy using apt-get install rtpproxy and here is the line
from kamailio.cfg for rtpproxy
modparam(rtpproxy, rtpproxy_sock,
Hello,
I've a scenario in which I've to deploy a couple Sangoma PRI cards with
kamailio. What I wish is that I've some drivers for this purpose and so I
don't ned to install FreeSWITCH or Asterisk in between the PRIs and
Kamailio.
Kindly give any feedback on what are the possibilities and
,
Sammy
On Thu, Oct 11, 2012 at 3:31 PM, David J da...@styleflare.com wrote:
Asterisk yate or free switch.
You need something as a gateway between PRI and sip. Kamailio does not
handle this conversion
On Oct 11, 2012 6:24 AM, SamyGo govoi...@gmail.com wrote:
Hello,
I've a scenario in which
neill.wilkin...@btinternet.com
wrote:
You might consider:
http://sangoma.com/products/voip_gateways/netborder_software/netborder_express.html
Then put Kamailio in front of that... Simple Gateway PRI - SIP.
Neill;o)
Aeonvista Ltd
Opening Up New Ideas
On 11 October 2012 11:43, SamyGo govoi
.
Thanks,
Sammy
On Thu, Oct 11, 2012 at 6:04 PM, Olle E. Johansson o...@edvina.net wrote:
11 okt 2012 kl. 13:57 skrev SamyGo govoi...@gmail.com:
:) Soon... But Not Today.
Not everyone can afford the Gateways. Thanks for the replies. I was hoping
maybe someone else be thinking of freeing
Is it typo or by chance you are missing a $ in line 507 before var(100)
On Sep 30, 2012 2:50 PM, Aft nix aft...@gmail.com wrote:
Hi,
After implementing a Call limit application with the following config:
#!ifdef WITH_CALL_LIMIT
modparam(dialog,enable_stats,1)
modparam(dialog,dlg_flag,
further into the right direction?
Sincerely,
Brandon Armstead
On Fri, Sep 14, 2012 at 9:13 PM, SamyGo govoi...@gmail.com wrote:
Yes, you should find the function engage_rtpproxy on module docs and use
it. It will work exactly like your old force-rtp-proxy but with more
enhanced way.
On Sep 15
side.
UAC 1 - KAM - UAC 2
I see RTP packets flow from UAC 1 - KAM - UAC 2
I also see RTP packets from from UAC 2 - KAM
The caller does not hear any audio, as KAM/RTPProxy is not sending audio
back to the caller.
Sincerely,
Brandon Armstead
On Fri, Sep 14, 2012 at 11:12 PM, SamyGo govoi
Yes, you should find the function engage_rtpproxy on module docs and use
it. It will work exactly like your old force-rtp-proxy but with more
enhanced way.
On Sep 15, 2012 7:34 AM, Brandon Armstead bran...@cryy.com wrote:
Klaus,
It seems that force_rtp_proxy is removed in Kamailio 3.3 ---
Hello,
I recently posted email relating to using MI_XMLRPC for kamailio
monitoring. As Daniel suggested and as mentioned in module docs Mi-xmlrpc
is getting obsolete and don't compile on newer OSs. So I need to use XMLRPC
module for pulling statistics from kamailio like get_statistics all:
Please
3:24 PM, SamyGo wrote:
Hello,
I've just finished monitoring all the core modules of kamailio via
XML-RPC. I use a script placed on kamailio server which sends command to
Kamailio over MI_XMLRPC and fetch details of each requested module. The
results fetched are then sent back
Hello,
I've just finished monitoring all the core modules of kamailio via XML-RPC.
I use a script placed on kamailio server which sends command to Kamailio
over MI_XMLRPC and fetch details of each requested module. The results
fetched are then sent back to the monitoring tool for graphing using
Hi,
You kind of sound a little different here. Are you saying that the
REGISTRATIONs will be handled by Freeswitch but store the registration Data
in Kamailio location table !?
Just go through the Kamailio blog by-Miconda or kb.asipto.com specially the
one on integrating the Asterisk Realtime
Hi,
Though not an expert, I got few things in mind for your case. In the
upgrade process the modules folder remain the same ? right ! if that is so
is there any chances that the new installation skipped installation of
newer rtpproxy module and hence the old version rtpproxy module could've
been
but please someone test what needs
to be changed for newer version.
Thanks
BR
Sammy.
On Aug 10, 2012 3:41 PM, SamyGo govoi...@gmail.com wrote:
Please see the SIP capture. As I was changing the online status from my
jitis and eyebeam phone I could see the publish requests handled by
kamailio
the problem.
Regards,
Peter
On Thu, 2012-08-09 at 15:30 +0500, SamyGo wrote:
Hi,
I've followed the tutorial on kab.asipto.com for presence using built-in
xcap server. http://kb.asipto.com/kamailio:presence:k31-made-simple
I'm using Kamailio version 3.3.1 and did minor changes
at all - so I don't think
that's going to work at all. I can't see XCAP/XDMS listed as an Ekiga
feature either.
I am using presence and XCAP on Kamailio 3.3 (my own configuration, not
the one from the tutorial) and it works fine.
Regards,
Peter
On Fri, 2012-08-10 at 13:41 +0500, SamyGo
by the SIP routing part of
Kamailio.
If you want to do offline message handling you need the msilo Kamailio
module (which isn't part of the tutorial).
Regards,
Peter
On Fri, 2012-08-10 at 14:16 +0500, SamyGo wrote:
Yes, Ekiga isn't good with it either. Can you just point me to some
Please see the SIP capture. As I was changing the online status from my
jitis and eyebeam phone I could see the publish requests handled by
kamailio but isn't it strange that I don'tfind any Notify generated and
relayed to the watchers ?
BR
Sammy
On Fri, Aug 10, 2012 at 3:04 PM, SamyGo govoi
Hi,
I've followed the tutorial on kab.asipto.com for presence using built-in
xcap server. http://kb.asipto.com/kamailio:presence:k31-made-simple
I'm using Kamailio version 3.3.1 and did minor changes in modparams and
rtpproxy function calls and the kamailio accepted the configurations file
Hi,
I'd suggest you to use multi-domain i.e separate domain for 1XX users and
a separate domain for 2XX users. Then for each domain map their own group
of load-balanced Asterisk servers (not just one server for 1XX clients but
a pool of servers for capacity and fail safe)
Next - if you want
Hello,
What are your requirements ! What functionality are you specifically
looking for in SIP Server.?
Few details here would help us helping you in deciding something.
Regards,
Sammy
On Sun, Jul 15, 2012 at 7:18 AM, Tony Rosa t...@myipsolution.com wrote:
I am interested sip server software
Hi,
Verify your mpath= directive in kamailio.cfg file.
Its just unable to find the directory holding all the modules.
If you're using a 64bit machine then I can only guess that you need to
change mpath to mpath= /usr/local/lib*64*/kamailio/modules/
Regards,
Sammy
On Wed, Jul 11, 2012 at
michal.yachimov...@gmail.com wrote:
Hi
Thanks, but the mpath is correct
If you have another idea it will be appreciated.
Michal
2012/7/11 SamyGo govoi...@gmail.com
Hi,
Verify your mpath= directive in kamailio.cfg file.
Its just unable to find the directory holding all the modules
know centos 5 used to have
quite old version that didn't work.
Be sure you cloned the last version of git repo. Can you give here the
output of 'git log -1' ?
Cheers,
Daniel
On 6/27/12 10:04 AM, SamyGo wrote:
Its: 1.7.10.4
On Wed, Jun 27, 2012 at 12:59 PM, Daniel-Constantin Mierla
Its: 1.7.10.4
On Wed, Jun 27, 2012 at 12:59 PM, Daniel-Constantin Mierla
mico...@gmail.com wrote:
Hello,
what is the version of git tool?
Cheers,
Daniel
On 6/26/12 6:45 AM, SamyGo wrote:
Hello,
I'm setting up new Kamailio 3.3 from git ref-link:
http://www.kamailio.org/wiki/install
This is a great thread, really full of answers and concepts for me atleast.
:)
On Mon, Jun 25, 2012 at 5:57 PM, Richard Brady rnbr...@gmail.com wrote:
Klaus / Daniel
Thanks again for assistance with this.
I've tried the solution based on add_contact_alias() and
handle_ruri_alias() and it
I had this thread already starred here in inbox, a wiki page is definitely
going to be bookmarked.
On Mon, Jun 25, 2012 at 9:44 PM, Daniel-Constantin Mierla mico...@gmail.com
wrote:
On 6/25/12 3:34 PM, SamyGo wrote:
This is a great thread, really full of answers and concepts for me
atleast
Hello,
I'm setting up new Kamailio 3.3 from git ref-link:
http://www.kamailio.org/wiki/install/3.3.x/git
I'm using CentOS 5.8 and when I execute the line
# git checkout -b 3.3 origin/3.3
it gives me error.
fatal: git checkout: updating paths is incompatible with switching branches.
Did you
Hello,
I've been pondering over an architecture of distributed VoIP services on
different geographic regions. I'm sure its nothing new and there are lots
of guys have ideas to share.
There are multiple SIP enabled Regions/locations, each has its own media
services and full services capability.
, event route is great for this.
So lets head to latest version
Thanks alot
Sammy G.
On Thu, Jun 14, 2012 at 1:26 PM, Daniel-Constantin Mierla mico...@gmail.com
wrote:
Hello,
On 6/12/12 11:49 AM, SamyGo wrote:
Yes, I saw the event routes in new version and that seems relevant. Yes
deleting
may have customers with SIP
trunks and I'd have to apply concurrent call limits to them using the same
logic.
Regards,
Sammy.
On Wed, Jun 13, 2012 at 1:23 PM, Daniel-Constantin Mierla mico...@gmail.com
wrote:
Hello,
On 6/12/12 12:01 PM, SamyGo wrote:
Thanks Sir,
Doing a static or even
, SamyGo wrote:
Hi,
sure, I'll give that debug messages a try tonight, Also I was thinking of
alternative approach and like you said, you are using htable. If I've to
use that htable entry manually how do I make sure that any call data gets
only deleted/erased after the call hangsup !? i.e I don't
., via mtree
or htable) and then you can do the alg 8.
Cheers,
Daniel
On 6/7/12 1:37 PM, SamyGo wrote:
Hi again,
yes my scenario is quiet simple. I've lots of users and groups of those
users are defined by UIDs, one UID means 70 users of one client whereas
other UID could've 3 users
11, 2012 at 5:20 PM, SamyGo govoi...@gmail.com wrote:
OK thanks,
I'm trying this all for myself. once I try this, test this, and get
something done I will post the tutorial for all.* I may need help in this.
*
Regards,
Sammy
On Mon, Jun 11, 2012 at 5:15 PM, Juha Heinanen j...@tutpro.com
Infact if kamailio goes down heartbeat restarts it again !! how do I stop
heartbeat doing it and let the other node take over the resource!
On Tue, Jun 12, 2012 at 6:05 PM, SamyGo govoi...@gmail.com wrote:
Hi again,
Some progress.. I have configured heartbeat and with the crm option to set
PM, SamyGo wrote:
Hi,
I personally think that there needs to be an official wiki page giving
details about a basic redundant/HA server setup. A lot of people need this
on regular basis. So I request forum members and contributors to share
their guidelines on this.
Thanks,
Sammy Go
alot of other folks.
Thanks again
BR
Sammy
On Mon, Jun 11, 2012 at 4:56 PM, Juha Heinanen j...@tutpro.com wrote:
SamyGo writes:
Thats just my suggestion, I am trying to setup a clustered service and
still no luck. Only heartbeat works which is pretty basic, anything on
top
of heartbeat
database, is any other query sent to delete the
dialog vars? This features was not used by me so far, still relying on
htable for my needs of such cases.
You can send the logs here for troubleshooting.
Cheers,
Daniel
On 6/6/12 6:45 PM, SamyGo wrote:
Hi Sir,
I've used the funcation
on a number.
Maybe if you explain what is your target to implement, we can provide the
right hints to do it.
Cheers,
Daniel
On 6/6/12 6:09 PM, SamyGo wrote:
Sorry for late reply: this wasn't very helpful. I think Hashing algo code
needs to get bit smarter. If there is any possibility can you
Hi,
Can you be more specific on what you are trying to achieve in terms of
servers topology !? See
TOPOLOGY-HIDINGhttp://www.kamailio.org/docs/modules/3.2.x/modules/topoh.htmlmodule
as well.
Regards.
Sammy G.
On Wed, Jun 6, 2012 at 5:45 PM, Mino Haluz mino.ha...@gmail.com wrote:
Hi,
I know
Hi,
Once I tried and pretty much succeeded in getting wireshark to capture
packets on the go from a server. I could see in wireshark capture interface
Remote:// - But AFAIR that needs some libraries to be installed on remote
server and 2-years ago it wasn't very stable/reliable as well.
Regards,
to servers,
which worked fine.
Carsten
2012/6/4 SamyGo govoi...@gmail.com:
Thanks Sir,
Thats what I expected. Do you think this will change if I use only two
values with huge difference in each other, lets say {26000,29000} ?
Regards,
Sammy
On Mon, Jun 4, 2012 at 3:05 PM, Carsten Bock
))
{
unset_dlg_profile(incall,$fU);
dlg_resetflag(4);
}
Still I can't seem to erase the values :(
Please suggest.
Best Regards,
Sammy G.
On Mon, Jun 4, 2012 at 12:41 PM, SamyGo govoi...@gmail.com wrote:
Thanks,
The reason why I didn't
and hard ware units (if you are testing media also)??
Can you please elaborate
@SamyGo, I didn't understand this,
If you've one good server then you can change SIPp command parameters and
only one machine can have like 2 or more SIPp generating calls in parallel.
how is that possible
record, you have to set a PV for it:
http://kamailio.org/docs/modules/stable/modules_k/dialog.html#id2555403
Alternative, you can use hashtable with a key composed from dialog id (or
callid) and some other string to identify your value.
Cheers,
Daniel
On 6/2/12 9:47 AM, SamyGo wrote:
Hello
function is guaranteed to return same code for same input
value, but there can be collisions of codes for different values.
Cheers,
Daniel
On 6/3/12 3:41 PM, SamyGo wrote:
Hello,
I'm having trouble using algorithm 7 in dispatcher module. Here is my
kamailio version. The problem
of values for your PVAR...
(e.g. try 1000 different values), then you should see a distribution.
Carsten
2012/6/4 SamyGo govoi...@gmail.com:
Hi,
No, nothing at all. I haven't went too deep into debug logs but no
internal
error was appearing. Everything was as calm as ever and I only get one
Hi Vineet,
Actually if you want to know how many concurrent calls any SIP server can
handle then SIPp is a great tool. No matter how SIPp makes calls
(*parallel*/sequentially)
the calls have a duration defined in your script. and SIPp keeps on sending
calls so they all are active for kamailio.
So
Hello
I've two problems regarding dialog module- used for profiling users that
are in-a-call.
My topology is quiet simple.
SIP-USER-71141Kamailio-Asterisk[AGI]||
CALLEE-Kamailio||
1- I wish to store my custom SIP header value
Hi Faisal,
You say you've seen packets on kamailio but kamailio isn't replying !!
1- Please verify that yuor kamailio is indeed started *[ps -ef | grep
kamailio]*
2- Verify if kamailio is actuallay listening on the same port as you
are sending pcakets to ! *[netstat -pln|grep kamailio]*
3-
.
Regards,
Faisal
On Tue, May 29, 2012 at 2:37 PM, SamyGo govoi...@gmail.com wrote:
Hi Faisal,
You say you've seen packets on kamailio but kamailio isn't replying !!
1- Please verify that yuor kamailio is indeed started *[ps -ef | grep
kamailio]*
2- Verify if kamailio is actuallay listening
://www.voipembedded.com
On Tue, May 15, 2012 at 12:49 AM, SamyGo govoi...@gmail.com wrote:
Thanks to both of you Elena and Ovidius - Since I'm not a php
programmer so
I can't promise a patch at this stage, however, I am planning on spending
time on it and, like Elena said, can write somewhat
Hello,
Three things you can check in this regard,
1- Verify the permission of the siremis and directories in that level. Per
my experience not just siremis directory but the directories aside it needs
to give ownership and permission in order for sessions to get created etc.
Extremly
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