Hello All,
I have my kamailio server running and working properly as much as it's
listening to an IP address..>As soon as i add an alias as a domain mae, I'm
not be able to make call anymore and i'm getting Proxy Authentication
required on my X-lite registered on the server...Please find below SIP
Hi All.
Does anyone know where to find training guide for Kamailio HA? I want to have 2
Kamailio in HA with four (4) Freeswitch servers behind them. I will also like
to customer cgrate to manage the accounting side in a multi-tenant environment.
I am just researching for this on my own and I w
ject: [SR-Users] Help
Hi,
I work for a VOIP service provider, and have been tasked
with optimizing our infrastructure. We have been providing VOIP services
to our clients via Asterisk VM’s (PIAF) in an ESXi environment, hosted in a
datacenter. We are looking for some kind of S
Fri, 26 Feb 2016 15:35:50 -0500
Subject: [SR-Users] Help
Hi,
I work for a VOIP service provider, and have been tasked with optimizing
our infrastructure. We have been providing VOIP services to our clients
via Asterisk VM’s (PIAF) in an ESXi environment, hosted in a datacenter.
We are looking f
[mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of
SamyGo
Sent: Sunday 28 February 2016 15:51
To: SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) - Users
Mailing List
Subject: Re: [SR-Users] Help
Hi,
I think the best guide closest to your description is here :
http
Hello,
run kamailio in higher debug mode to see the log messages with hints why
is not authenticated.
I see that you have double authentication, one for realm asterisk and
one for realm aliasdomain.com. Is it what you want to have?
One more hint, in 3.1 you can load debugger module and prin
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 SI
As a word of advice.
You will have to pay someone for the mobile or PSTN calls. It is very
unsafe to open your system to the public without very good security,
especially very strong user passwords.
There are constant attacks by robots that attempt to break into systems
and get free (for them) ca
t for possible
partners.
Cheers.
Amar Tuladhar,
Smart Telecom
From: sr-users-boun...@lists.sip-router.org
[mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of ameneh
pourmoghadas
Sent: Friday, December 07, 2012 9:38 AM
To: serus...@iptel.org
Subject: [SR-Users] Help
Dear Admin,
My
gt;
> Amar Tuladhar,
>
> Smart Telecom**
>
> ** **
>
> *From:* sr-users-boun...@lists.sip-router.org [mailto:
> sr-users-boun...@lists.sip-router.org] *On Behalf Of *ameneh pourmoghadas
> *Sent:* Friday, December 07, 2012 9:38 AM
> *To:* serus...@iptel.org
> *Subject:* [SR-Users]
d payments. You can easily search the internet for
>> possible partners.
>>
>> ** **
>>
>> Cheers.
>>
>> Amar Tuladhar,
>>
>> Smart Telecom**
>>
>> ** **
>>
>> *From:* sr-users-boun...@lists.sip-rout
Hi Roy,
Please keep emails on list.
Regards,
Peter
On Fri, 2012-12-14 at 09:31 -0800, Raj Roy Ghandhi wrote:
> Dear Peter,
>
> Thanks for all the support that you gave me during the Kamalio
> configuration with RTPProxy.
> Currently it works fine with IP Phones but not soft clients. (Jitsi)
>
I got my sip account ipadv...@iptel.org what is next step VoIP to complete plz
reply thank you my email yahya_sat...@yahoo.com
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-
HI
I am newbie, I have a centos pc with httpd, php and mysql installed in it
i followed the procedure at http://siremis.asipto.com/install/
when i m trying to browse my siremis web it is unable to submit my
request...
may be i am missing something configuring Apache
please guide me how can i get
Hello,
before any major release of kamailio (openser), we install and run it
live on voipuser.org sip service. It runs there for more than two weeks
now, it even survived a 18 hours scan attack (approx 12.5 mio requests
"with love from Latvia" - I will write an article about it), so it looks
Hello there,
I am trying to setup a Kamailio (3.3) + Linphone + Jitsi based private calling
system (users can call each other but not call out of the network).
Agents can be behind NAT (think Verizon cellphone or home-users behind their
router). I started with the default setup and have tried
Hello Everyone,
I am trying forward NOTIFY to client.
My setup is
PBX Server local lan ---kamailio internet client
The problem with whole setup that kamailio is not forwarding NOTIFY to client.
Here are log.
U 2016/10/19 14:23:29.581609 10.18.130.50:5160 -> 10.18.130.46:
Hello,
we are using a kamailio 3.3.1 version here and suddenly it crashed with
a core.
Any point about how to debug this will be appreciated!
# /usr/local/kamailio/sbin/kamailio -V
version: kamailio 3.3.1 (x86_64/linux) 91e8cb
flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW
Hi All,
I wonder if some one could help me to diagnose a recurring issue?
It happens at random times/intervals and under varying load. But always,
just before the time of crash, I see the same critical error in log:
CRITICAL: dialog [dlg_hash.c:841]: log_next_state_dlg(): bogus event 6 in
state
Hello,
i'm on my first try with kamailio. I need to build a SIP balancer that
should keep SIP
registration from VoIP provider and route the calls to the asterisk boxes
where an IVR
will take care to answer.
Here's my network topology:
+---> [asterisk1]
[publi
engineerzuhairr...@gmail.com
To: sr-users@lists.sip-router.org
Subject: [SR-Users] Help regarding SIREMIS
HI
I am newbie, I have a centos pc with httpd, php and mysql installed in it
i followed the procedure at http://siremis.asipto.com/install/
when i m trying to browse my siremis web it is una
thanks for all support from the forum before hand.
i am having an issue with my setup, i am trying to make the dispatcher
dynamic by asigning a value from a variable, that variable is define on my
dispatcher.lst but i am getting an error, if anyone can help thanks.
my config on the relay route
fi
Hello,
it is clear you have an ALG in the middle that breaks the signalling. Can
you try using TLS? You may try first using a different port than 5060 and
see if the ALG is still capturing the traffic.
Cheers,
Daniel
On Tue, Oct 22, 2013 at 6:42 AM, P. S. wrote:
> Hello there,
>
> I am trying
you.
volga629
From: "volga629"
To: "sr-users"
Sent: Wednesday, 19 October, 2016 18:36:52
Subject: [SR-Users] help with NOTIFY
Hello Everyone,
I am trying forward NOTIFY to client.
My setup is
PBX Server local lan ---kamailio internet client
Th
Hello All.
I have some troubles with upstream sip switch.
It ignores SIP packets which contains:
User-Agent: FPBX-2.11.0(11.17.1)
or
Server: User-Agent: FPBX-2.11.0(11.17.1)
If space is present before first "(" then sip switch works as expected
So my question is: how corektly make analyze and m
Hello,
it seems that the core file of the process that crashed was overwritten
by a core dumped by the main process while cleaning up.
Did you get many core files? If not, can you look in the log messages to
see if there were some error messages printed at the time of crash?
You should enab
Hello,
can you try this small patch?
diff --git a/modules/pua_dialoginfo/pua_dialoginfo.c
b/modules/pua_dialoginfo/pua_dialoginfo.c
index 1e88a04..0f02b2b 100644
--- a/modules/pua_dialoginfo/pua_dialoginfo.c
+++ b/modules/pua_dialoginfo/pua_dialoginfo.c
@@ -347,7 +347,7 @@ struct str_list* get
Thanks, Daniel.
It can be hours, days or weeks between occurrences, but I will report back
after a day or two initially then continue to monitor.
Cheers,
Charles
On 22 Aug 2014 12:18, "Daniel-Constantin Mierla" wrote:
> Hello,
>
> can you try this small patch?
>
> diff --git a/modules/pua_d
Hi Daniel,
The patch has tested OK so far.
Regards,
Charles
On 22 August 2014 12:37, Charles Chance
wrote:
> Thanks, Daniel.
>
> It can be hours, days or weeks between occurrences, but I will report
> back after a day or two initially then continue to monitor.
>
> Cheers,
>
> Charles
> On
Hey Daniel,
I am puzzled by how this could make any difference? Could you explain? Is
this dependent on the compiler used and whether or not void* arithmetic is
allowed?
Cheers
Jason
On Fri, Aug 22, 2014 at 1:17 PM, Daniel-Constantin Mierla wrote:
> Hello,
>
> can you try this small patch?
>
Hello,
On 28/08/14 20:32, Jason Penton wrote:
Hey Daniel,
I am puzzled by how this could make any difference? Could you explain?
Is this dependent on the compiler used and whether or not void*
arithmetic is allowed?
void is incomplete type, of no defined data size, you cannot have:
void x;
Yeah, would be cool to see what compiler Charles is using.
Thanks Daniel
On 28 Aug 2014 22:25, "Daniel-Constantin Mierla" wrote:
> Hello,
>
> On 28/08/14 20:32, Jason Penton wrote:
>
> Hey Daniel,
>
> I am puzzled by how this could make any difference? Could you explain?
> Is this dependent on
Hi Both,
In this case the system was using binaries from opensuse build service, so
gcc I believe?
Cheers,
Charles
On 28 Aug 2014 21:25, "Daniel-Constantin Mierla" wrote:
> Hello,
>
> On 28/08/14 20:32, Jason Penton wrote:
>
> Hey Daniel,
>
> I am puzzled by how this could make any differen
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 $dlg_var(cs
Hello,
I am seeking help with a new test server.
I have SIP messages coming from a working FreeSwitch and getting posted into
the sip_capture table.
Homer web service is responding allowing me to manage log on accounts and other
database settings however I do not see any packets in the web gui.
First of all I'd suggest to use
http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb
guide in combination with
http://saevolgo.blogspot.com/2011/11/how-to-increasing-voip-services.html
But, assuming your platform is behind NAT, you need:
1st. Use rtpengine instead of rtpproxy
Also, take a look at kamailio-advanced.cfg, there is PSTN GW route already
included. Also you can use LCR for routing calls to different providers, a
simple guide can be found here
http://dopensource.com/least-cost-routing-with-kamailio-v4-1/
2015-08-12 0:41 GMT+03:00 Alexandru Covalschi <568...@g
Thankyou Alexandru for your suggestions.
I'll give it a try tomorrow and will report my progress here.
It seems that i'm not so far from the result!
Bruno
Il giorno mar 11 ago 2015 alle 23:44 Alexandru Covalschi <568...@gmail.com>
ha scritto:
> Also, take a look at kamailio-advanced.cfg, there is
Hi!
I am a newbie and I need help understanding how to use the java module with
Kamailio. I am trying to create a basic java application that will print the
SIP message type being processed by Kamailio.
I read the app_java module but can't still figure it out. Any help or reference
will be gre
Hi!
I am trying to use the app_java module with kamailio-4.4.0 SIP server.
Every time I load the app_java module, my users can't register and the
sever replies with destintion unreachable (port unreachable).
How can I fix this issue?
Also, how can I use the Kamailio.java file since it doesn't hav
Hi!
I am trying to use the app_java module with kamailio-4.4.0 SIP server.
Every time I load the app_java module, my users can't register and the
sever replies with destintion unreachable (port unreachable).
How can I fix this issue?
Also, how can I use the Kamailio.java file since it doesn't hav
Everything you want to do is possible with the default configuration.
Just enable all configuration parts with "fix_nated_contact" and
fix_nated_register".
This will enable NAT traversal for SIP. Depeing if you want to relay
media or not just enable/disable the "force_rtpproxy" calls.
regards
kl
com
> CC: sr-users@lists.sip-router.org
> Subject: Re: [SR-Users] Help need on NAT_Traversal
>
> Everything you want to do is possible with the default configuration.
>
> Just enable all configuration parts with "fix_nated_contact" and
> fix_nated_register".
> This
hi all,
I hae configure RTP proxy it work ok.
please help me to configure that :
if sip client is in the same NAT(same LAN network) , they can make call
directly
(RTP go direct without through RTP proxy).
thanks so much.
ThanhTruong
HTK-INC Co.,Ltd VN
www.htk-inc.com
http://pbxvn.com/forum/
Dear sir,
How are you? I have installed SER and it runs well. Now I am adding a SIP
client to make phone call. I downloaded SERCTL and up, but do not know to
use it. Could you tell me how to install and use the software?
Or another way is to add user to the tables in MySQL? could you tell me wha
Hi
I'm trying to integrate a (K) front end cluster with an Asterisk back end
cluster and Asterisk RT (legacy system)
I've followed the recipe at
http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb
(with one minor exception …)
if (from_uri!=myself && uri!=myself
Hello,
put here the text that appears when you press show error button. Also,
say the version of kamailio and siremis.
Another place to check is the error logs from apache/http server.
Cheers,
Daniel
On 1/24/13 6:11 AM, 문종민 wrote:
I'm developers who trying to install siremis.
will go in t
Hello
you may try to cast $var(y) to int using this transformation:
http://www.kamailio.org/wiki/cookbooks/4.0.x/transformations#sint
Regards
Javi
On 04/09/2013 12:17 AM, julian arsanches wrote:
thanks for all support from the forum before hand.
i am having an issue with my setup, i am tryi
Hi all, I am trying to install the latest version using git (new to git) and
after running "git checkout -b 4.0 origin/4.0" I got these messages:
[root@test kamailio-4.0.2]# git checkout -b 4.0 origin/4.0
fatal: Cannot update paths and switch to branch '4.0' at the same time.
Did you intend to
>
> Hi all, I am trying to install the latest version using git (new to git) and
> after running "git checkout -b 4.0 origin/4.0" I got these messages:
>
>
> [root@test kamailio-4.0.2]# git checkout -b 4.0 origin/4.0
>
> fatal: Cannot update paths and switch to branch '4.0' at the same time.
> Did
Hello ,
This is the first time to put a question on sr-users mailing list and i
hope to help me to solve my issue...i'm following this post
http://saevolgo.blogspot.com/2013/08/rtpproxy-revisited-kamailio-40.html to
setup RTP Proxy for kamailio 4 but it's NOT working as predicted and i'm
receivin
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 to go further. How can I authenticate Asterisk users
through Kamailio. I am trying to authenticate using a sipphone. But no
luc
On Fri, Nov 25, 2016 at 02:08:07PM +0200, Sergey Basov wrote:
> Hello All.
>
> I have some troubles with upstream sip switch.
> It ignores SIP packets which contains:
>
> User-Agent: FPBX-2.11.0(11.17.1)
> or
> Server: User-Agent: FPBX-2.11.0(11.17.1)
>
> If space is present before first "(" the
Thank you Daniel.
Is it safe to use remove_hf("User-Agent") without check if this header
exist?
or better use if(is_present_hf("User-Agent")) { remove_hf("User-Agent"); } ?
Thank you.
25 нояб. 2016 г. 2:56 PM пользователь "Daniel Tryba"
написал:
> On Fri, Nov 25, 2016 at 02:08:07PM +0200, Serg
On Fri, Nov 25, 2016 at 06:55:34PM +0200, Sergey Basov wrote:
> Is it safe to use remove_hf("User-Agent") without check if this header
> exist?
> or better use if(is_present_hf("User-Agent")) { remove_hf("User-Agent"); } ?
Just remove_hf is enough. is_present_hf/remove_hf might be more readable
th
Hi, All.
One more question related to remove_hf...
I have added route:
# Fix user-agent and server
route[RemoveHeader] {
remove_hf("server");
remove_hf("user-agent");
return;
}
I use it form
request_route {
route(RemoveHeader);
.
}
failure_route[--- all what i
Hi All.
Today I have problem with connection from 1 of the clients.
Their PBX sends KEEP-ALIVE after some time after REGISTER.
I have next error in kamailio log
Feb 28 14:26:19 sbc2 /usr/sbin/kamailio[3657]: ERROR:
[tcp_read.c:1354]: tcp_read_req(): bad request, state=7, error=4
buf:#012KEEP-AL
Hi to everyone,
I’m trying to implement a SBC for my network based on Kamailio
The base idea is putting Kamailio on a dual home machine (one public interface
for clients and a private interface for media servers and database), using
rtpproxy to handle the relaying from public to private network.
Hello,
$dlg_var(cseq_diff) is incremented after sending the invite out from
failure route, being done when forwarding callback in dialog detects
that the cseq value has to be incremented.
I am going to test and see if there is an issue -- uac_auth() should set
some internal flag to tell dialog to
Great! I will waiting for answer. If it needed I may make some tests. We
building new system and want to use this technology insread of classic
gateway. We will happy to cooperate with you for findinf issues and solve
it as faster as we may. Thanks!
2014-11-03 20:03 GMT+04:00 Daniel-Constantin Mie
I just pushed a patch to master, can you try with it and if all is ok,
then I will backport.
Cheers,
Daniel
On 03/11/14 17:31, Yuriy Gorlichenko wrote:
> Great! I will waiting for answer. If it needed I may make some tests.
> We building new system and want to use this technology insread of
> cla
Hi all, i cannot understand where is the problem with this transaction:
Kamailio ask for Proxy authorization and in the second INVITE credentials
are present.
Can you help me understand?
Regards
U 2015/03/09 15:03:42.191831 10.160.21.51:5060 -> 10.160.20.18:5060
INVITE sip:1...@sip.longwave.labo
Hello,
check the network traffic to see if the packages are sent to the
sipcapture node.
If kamailio is involved somehow, try to run it with debug=3 in kamailio.cfg
Cheers,
Daniel
On 06/03/15 21:33, David Dunlap wrote:
> Hello,
>
> I am seeking help with a new test server.
> I have SIP messages
Hello,
I see the module has some txt docs in the folder:
https://github.com/kamailio/kamailio/tree/master/modules/app_java
Is it what you have tried?
Cheers,
Daniel
On 09/04/16 08:10, Bonjour Madame wrote:
>
> Hi!
>
>
>
> I am a newbie and I need help understanding how to use the java module
Thanks for responding. I followed the steps in the readme documents and was
able to compile my application. However, when I try to run it, I get the
error unsatisfiedlinkerror undefined symbol: log_stderr.
I don't know what is the reason and how I can solve it.
On Apr 11, 2016 1:05 AM, "Daniel-Con
I am not using the app_java module, so not familiar with how was built.
log_stderr is a variable inside kamailio.
Can you share the java code you try to run?
Cheers,
Daniel
On 11/04/16 10:53, Bonjour Madame wrote:
>
> Thanks for responding. I followed the steps in the readme documents
> and was
Hello,
can you check if kamailio is actually running?
ps auxw | grep kamailioo
Also, look inside the syslog file to see what messages are written there
by kamailio.
Cheers,
Daniel
On 14/04/16 20:03, Clarence Sandjon wrote:
> Hi!
>
> I am trying to use the app_java module with kamailio-4.4.0 SI
Dear All,
I am very new to OpenSer. I want to use latest version of OpenSer with
Radius. I need the documentation/tutorial on how to do this. Googling, Ionly
found for the old version. Please help me.
Regards,
Pratik
___
SIP Express Router (SER) and Kam
2010/8/27 truong ngoc THANH :
> please help me to configure that :
> if sip client is in the same NAT(same LAN network) , they can make call
> directly (RTP go direct without through RTP proxy).
> thanks so much
Truong, nobody will do your work for you by free. This is a users
maillist not a free
I too can confirm it.
--
Alex Balashov - Principal
Evariste Systems LLC
1170 Peachtree Street
12th Floor, Suite 1200
Atlanta, GA 30309
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/
On Aug 27, 2010, at 8:40 AM, Iñaki Baz Castillo wrote:
2010/8/27 truong ngoc THANH
/
From: Alex Balashov
To: Iñaki Baz Castillo
Cc: truong ngoc THANH ; kamailio
Sent: Fri, August 27, 2010 7:43:21 PM
Subject: Re: [SR-Users] help to configre RTP proxy.
I too can confirm it.
--
Alex Balashov - Principal
Evariste Systems LLC
1170 Peachtree Street
12th Floor, Suite 1200
2010/8/27 truong ngoc THANH :
> hi all,
> thanks for your answer.
> but i think this is a problem when configure nathelper. because when we use
> rtpproxy, we must divide some case as different NAT and same NAT.
Which problem do you mean? It's as easy as matching two IP's.
Regards.
--
Iñaki Ba
: Alex Balashov ; kamailio
Sent: Fri, August 27, 2010 10:14:49 PM
Subject: Re: [SR-Users] help to configre RTP proxy.
2010/8/27 truong ngoc THANH :
> hi all,
> thanks for your answer.
> but i think this is a problem when configure nathelper. because when we use
> rtpproxy, we must
Subject: Re: [SR-Users] help to configre RTP proxy.
2010/8/27 truong ngoc THANH :
> hi,
> this problem is, the clients in the same NAT make call , and i see that
> media always be handled by rtpproxy.
> what is wrong ?
And why do you expect that force_rtp_proxy() would do nothing
2010/8/27 truong ngoc THANH :
> hi,
> this problem is, the clients in the same NAT make call , and i see that
> media always be handled by rtpproxy.
> what is wrong ?
And why do you expect that force_rtp_proxy() would do nothing in case
both endpoints are in the same LAN? where in the nathelper mo
hi all,
i am a new user kamailio.
i have configure kamailio with RTP proxy, but i have a problem in using :
force_rtp_proxy("c","192.168.1.10")because i want to change to force to this ip.
my configure is :
when server receive "200 OK" it change value in "c= IN IP4 " to
"c = IN IP4 192.168.1
...@gmail.com
To: serus...@iptel.org
Subject: [SR-Users] help for SIP client configuration
Dear sir,
How are you? I have installed SER and it runs well. Now I am adding a SIP
client to make phone call. I downloaded SERCTL and up, but do not know to use
it. Could you tell me how to install and use the
i want to instal tunnel server in kamailio .
i want to ask if there is any opensource voip tunnel server .
and if there is an tutorial to configure kamailio with tunnel server
thanks for help
hala
___
SIP Express Router (SER) and Kamailio (Ope
Hello,
On 12/12/12 4:19 AM, Jon Morby wrote:
Hi
I'm trying to integrate a (K) front end cluster with an Asterisk back end
cluster and Asterisk RT (legacy system)
I've followed the recipe at
http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb
(with one minor exception
On Thursday 13 December 2012 12:09:46 Daniel-Constantin Mierla wrote:
> > The problem I'm seeing currently is that when a call is passed down a SIP
> > trunk to an end user on the (K) platform we're losing the DNID
> >
> > Asterisk delivers the call to SIP/account/DNID
> >
> > (K) however just tr
On 13 Dec 2012, at 11:09, Daniel-Constantin Mierla wrote:
> Hello,
>
> On 12/12/12 4:19 AM, Jon Morby wrote:
>> Hi
>>
>> I'm trying to integrate a (K) front end cluster with an Asterisk back end
>> cluster and Asterisk RT (legacy system)
>>
>> I've followed the recipe at
>> http://kb.asipto
On Wednesday 12 December 2012 04:19:48 Jon Morby wrote:
> For legacy reasons Asterisk needs to be in the critical path on this
> particular build … what I'm looking for is a simple recipe and some
> helpful pointers on how to implement it that will allow enable me to swing
> (K) into the path betwe
Hello,
I thought some people may have spare cycles and are willing to help with
the documentation. Here is some updates I would like to get for modules:
1) adding ids for parameters, functions, and other sections of the
docbook xml files. Now we generate the alphabetic indexes pointing to
th
Hello,
recent versions of git changed behaviour for --depth 1, fetching only
one branch. You have to give --no-single-branch as parameter, see:
http://www.kamailio.org/wiki/install/4.0.x/git#getting_sources_from_git
Cheers,
Daniel
On 6/18/13 9:29 PM, John Doe wrote:
Hi all, I am trying to i
Thank you Daniel, it worked.
> From: jdoe_...@hotmail.com
> To: sr-users@lists.sip-router.org
> Subject: Help with install using GIT
> Date: Wed, 19 Jun 2013 10:06:36 -0400
>
>>
>> Hi all, I am trying to install the latest version using git (new to git) and
Hello,
you can use vim, nano or other editor to remove rtpproxy_manage() from
the config file -- I would recommend to keep the one from branch_routes
and be sure those branch routes are executed even for communication with
asterisk. Asking someone from the community to do file editing is rathe
Hello,
I like to thank u for the reply and also i want to tell u that i'm good at
Linux but scripting is NOT my domain so i will send this issue to the
mailing list hoping that someone will modify the script and make the
required changes...by any way thank u so much and i will message u if i
have a
Hello all,
Have a nice day.
I am given a task in office to setup Kamailio-3.1.1 on Fedora 14 machine. Being
new to this, I went online for guidance.
I got a proper method on the link given below and followed the instructions
given.
http://www.kamailio.org/dokuwiki/doku.php/install:kamailio-3.1.x
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 kno
Hi,
I'm currently running Kamailio 4.1 and was wondering if there is a way to
control ds_ping_interval on a dispatcher setid.
Thanks.
--
Andy Chen
Sr. Telephony Lead Engineer
415 516 5535 (M)
ac...@thinkingphones.com
___
SIP Express Router (SER) and K
Hi,
I’m very appreciated if someone can figure out this problem for me.
My aim is to setup HA Kamailio using Dispatcher module in Amazon EC2, system
like this:
soft phone 1(Linphone) —> | Load balancer—> |
Sip Server
huancomputer
Hello,
On 28/02/2017 17:05, Sergey Basov wrote:
> Hi All.
>
> Today I have problem with connection from 1 of the clients.
> Their PBX sends KEEP-ALIVE after some time after REGISTER.
>
> I have next error in kamailio log
>
> Feb 28 14:26:19 sbc2 /usr/sbin/kamailio[3657]: ERROR:
> [tcp_read.c:135
Hi, Daniel
Yes it is without any headers...
I have attached screenshot from wireshark, I can not save it because
this is sip tls...
Thank you
--
Best regards,
Sergey Basov e-mail: sergey.v.ba...@gmail.com
2017-03-01 15:57 GMT+02:00 Daniel-Constantin Mierla :
> Hello,
>
>
> O
2017-03-01 15:48 GMT+01:00 Sergey Basov :
> 2017-03-01 15:57 GMT+02:00 Daniel-Constantin Mierla :
>> If yes, this is not a valid SIP message, because it lacks mandatory
>> headers such as call-id, cseq, from/to.
>>
> Yes it is without any headers...
So is not a valid SIP message
_
Thank you for clarification.
--
Best regards,
Sergey Basov e-mail: sergey.v.ba...@gmail.com
2017-03-01 20:05 GMT+02:00 Victor Seva :
> 2017-03-01 15:48 GMT+01:00 Sergey Basov :
>> 2017-03-01 15:57 GMT+02:00 Daniel-Constantin Mierla :
>>> If yes, this is not a valid SIP message
I have this setup for kamailio + asterisk, in which kamailio is supposed to listen on all ethernet interfaces on UDP port 5060, and will forward traffic from/to asterisk running on the same machine, and listening on localhost, udp port 5080. The scenario
for the problematic call is somewhat like t
Hi,
I had installed kamailio on my AWS EC2, by following the steps in
http://kamailio.org/docs/install/INSTALL.kamailio-4.1.x . Since I installed
in EC2, i changed the listen= advertise= in
kamailio.cfg file too. I open the 5060 port in AWS console too. Kamailio
starts without any error, but I c
Hello,
understanding the config file is going to take some time, so it is
unlikely many will have the spare time for it.
You have to provide more specific details, like the sip trace (ngrep on
kamailio server sip ports) for the issue, presenting what are the
parties involved in sending and r
Hello,
the situation is:
client -> kamailio + rtproxy -> asterisk -> rtpproxy + kamailio -> other client
the idea is that rtpproxy has to proxy the whole rtp traffic between asterisk
or other media proxy and the client.
the strange problem is that my client which is receiving the call (csipsimp
Hello,
On 12/09/14 14:49, [PRE s.r.l.] - Alex wrote:
Hello,
the situation is:
client -> kamailio + rtproxy -> asterisk -> rtpproxy + kamailio ->
other client
the idea is that rtpproxy has to proxy the whole rtp traffic between
asterisk or other media proxy and the client.
the strange prob
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