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
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
_
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
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
January 2017 04:27
To: sr-users@lists.sip-router.org
Subject: Re: [SR-Users] Help Asterisk with Kamailio unable to register with
remote VOIP providers
Hi,
just remove:
#!define WITH_ASTERISK
From your kamailio.cfg and restart it.
--
Daniel Grotti
On 01/02/2017 06:36 PM, Manoj Gupta wrote
sage.
-Original Message-
From: sr-users [mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of
Daniel Grotti
Sent: 02 January 2017 10:52
To: sr-users@lists.sip-router.org
Subject: Re: [SR-Users] Help Asterisk with Kamailio unable to register with
remote VOIP providers
Kamailio-asterisk
2
To: sr-users@lists.sip-router.org
Subject: Re: [SR-Users] Help Asterisk with Kamailio unable to register with
remote VOIP providers
Kamailio-asterisk doc:
http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb
There are tones of documentation about kamailio out there.
Cons
:sr-users-boun...@lists.sip-router.org] On Behalf Of
Daniel Grotti
Sent: 02 January 2017 08:54
To: sr-users@lists.sip-router.org
Subject: Re: [SR-Users] Help Asterisk with Kamailio unable to register with
remote VOIP providers
Hi,
please configure this in your kamailio.cfg:
debug=3 # debu
January 2017 10:34
To: sr-users@lists.sip-router.org
Subject: Re: [SR-Users] Help Asterisk with Kamailio unable to register with
remote VOIP providers
You should add "ims.airtel.in" as kamailio local domain, in your
kamailio.domain table.
--
Daniel Grotti
On 01/02/2017 05:31 PM,
if
memdbg=5
memlog=5
#log_facility=LOG_LOCAL0
log_facility=LOG_LOCAL6
Manoj K. Gupta
-Original Message-
From: sr-users [mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of
Daniel Grotti
Sent: 02 January 2017 08:10
To: sr-users@lists.sip-router.org
Subject: Re: [SR-Users] Help
sr-users [mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of
Daniel Grotti
Sent: 02 January 2017 08:54
To: sr-users@lists.sip-router.org
Subject: Re: [SR-Users] Help Asterisk with Kamailio unable to register with
remote VOIP providers
Hi,
please configure this in your kamailio.cfg:
de
el Grotti
Sent: 02 January 2017 08:54
To: sr-users@lists.sip-router.org
Subject: Re: [SR-Users] Help Asterisk with Kamailio unable to register with
remote VOIP providers
Hi,
please configure this in your kamailio.cfg:
debug=3 # debug level, 1 is low and 4 is high (lots of output)
log_facility=L
al Message-
From: sr-users [mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of
Daniel Grotti
Sent: 02 January 2017 08:10
To: sr-users@lists.sip-router.org
Subject: Re: [SR-Users] Help Asterisk with Kamailio unable to register with
remote VOIP providers
Hi,
have you configured kamail
=5
memlog=5
#log_facility=LOG_LOCAL0
log_facility=LOG_LOCAL6
Manoj K. Gupta
-Original Message-
From: sr-users [mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of
Daniel Grotti
Sent: 02 January 2017 08:10
To: sr-users@lists.sip-router.org
Subject: Re: [SR-Users] Help Asterisk
Hi,
have you configured kamailio in order to log to /var/log/kamailio
instead of syslog ?
https://www.kamailio.org/dokuwiki/doku.php/utils:basic-syslog-configuration
--
Daniel Grotti
On 01/02/2017 03:36 PM, Manoj Gupta wrote:
Request to all – Please help we are BADLY stuck in this asterisk
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
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
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 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
Hello Everyone,
This message as continue conversation from
http://lists.sip-router.org/pipermail/sr-users/2015-March/087557.html. That my
previous post about it. I never was be be able forward NOTIFY from asterisk to
client through kamailio.
Right now in use asterisk 14 pjsip.
Any help thank
Hello
I tested 4.3 and 4.4 and kamailio -E -DDD gives the sameI compiled from source
- the same results:[root@kazootest2 kamailio]# kamailio -E -DDDloading modules
under config path: /usr/local/lib64/kamailio/modules/ 0(1) INFO:
[sctp_core.c:75]: sctp_core_check_support(): SCTP API not enab
Hello
I tested a package
http://download.opensuse.org/repositories/home:/kamailio:/v4.3.x-rpms/CentOS_6/x86_64/kamailio-4.3.6-1.1.x86_64.rpm
( I downlowaded 4.3.6 version rpm and installed it. Kamailio restarts well.
The behaviour is the same. the phone registeres without nonce.
I install k
RPMBUILD produces several kamailio rpms Now I install the following rpms:
[root@kazootest3 ~]# rpm -qa | grep
kamakamailio-presence-4.3.4-0.x86_64kamailio-4.3.4-0.x86_64kamailio-utils-4.3.4-0.x86_64kamailio-outbound-4.3.4-0.x86_64kamailio-tls-4.3.4-0.x86_64kamailio-kazoo-4.3.4-0.x86_64
still no
I made loadmodule and modparam("debugger", "cfgtrace", 1)
but anyway - no logs when I register.
As I understand - it's like no config file.
On Thursday, September 15, 2016 6:01 PM, Daniel-Constantin Mierla
wrote:
I am not familiar with kazoo configs, maybe asking on their mailing list
I am not familiar with kazoo configs, maybe asking on their mailing list
can help you more.
>From Kamailio point of view, you can load debugger module and set its
cfgtrace parameter to 1, then see what actions from config are executed
and why is not getting to the authentication part.
Cheers,
Dan
here are my "define_with flags" from SPEC file (opensuse one)
# list of flags to enable extra packages%define _with_bdb 0%define
_with_carrierroute 0%define _with_cnxcc 0%define _with_dnssec 0%define
_with_erlang 0%define _with_ev 0%define _with_geoip 0%define _with_java
0%define _with_json 0%d
/etc/kazoo/kamailio/default.cfg - which containes all
routes.2600hz/kazoo-configs
|
|
|
| ||
|
|
|
||
2600hz/kazoo-configs
kazoo-configs - Kazoo Configuration Files for Software We Use | |
|
|
I test on a working server (testing one) and a working con
Are you using default kamailio.cfg or another one?
Cheers,
Daniel
On 15/09/16 12:39, Dmitry wrote:
> Hello
>
> I took this spec from suse.
>
> It generates no errors.
>
> When I installed from the RPM I had made - the phone register, but
>
> The phone sends a REGISTER and the KAmailio sends 200o
Hello
I took this spec from suse.
It generates no errors.
When I installed from the RPM I had made - the phone register, but
The phone sends a REGISTER and the KAmailio sends 200ok back to the phone (so
no NONCE authorization) and no logs during it.
In default.cfg I set L_DBG but no logs are gener
Then you just need add those files in various packages inside the spec
file, so they are not detected to be orphaned.
Maybe you can inspire from:
-
https://build.opensuse.org/package/view_file/home:kamailio:v4.3.x-rpms/kamailio43/kamailio.spec?expand=1
Cheers,
Daniel
On 14/09/16 16:35, Dmitry
4.3.4 version is for KazooIt is on production server currently.
I need to rebuild the current RPM so as to apply patches.
But first I want to get a working Kamailio and only after it I will apply the
patches.
I think I may take a list of modules from the production Kazoo-kamailio and
rearchive th
Hello,
any reason not to use series 4.4.x? Iirc, the latest spec that got
update on 4.4 are those for oracle enterprise linux, perhaps is
something that you can reuse a lot for upgrading to the centos flavour.
On the other hand, you can use opensuse build service if you want to
build yourself, th
Hellowhich SPEC file is used by the Kamailio group to build rpm?
On Tuesday, September 13, 2016 7:56 PM, Dmitry
wrote:
I use Centos 6.7
On Tuesday, September 13, 2016 7:51 PM, Dmitry
wrote:
Hello
I used:
kamailio-4.3.4_src.tar.gz
/kamailio-4.3.4/pkg/kamailio/centos/6/
I
I use Centos 6.7
On Tuesday, September 13, 2016 7:51 PM, Dmitry
wrote:
Hello
I used:
kamailio-4.3.4_src.tar.gz
/kamailio-4.3.4/pkg/kamailio/centos/6/
I found several spec files:[root@kazootest2 kamailio-4.3.4]# find . -name
*.spec./pkg/ser/suse/ser.spec./pkg/ser/opensuse/ser.spec.
Hello
I used:
kamailio-4.3.4_src.tar.gz
/kamailio-4.3.4/pkg/kamailio/centos/6/
I found several spec files:[root@kazootest2 kamailio-4.3.4]# find . -name
*.spec./pkg/ser/suse/ser.spec./pkg/ser/opensuse/ser.spec./pkg/kamailio/centos/6/kamailio.spec./pkg/kamailio/fedora/17/kamailio.spec./pkg/kamai
Hello,
which rpm spec did you use? There are several of them in the source
tree, some not really maintained.
Cheers,
Daniel
On 13/09/16 14:33, Dmitry wrote:
> Hello, All
>
> When I take a SPEC file from kamailiotar.gz - during rpmbuild I
> encounter:
>
> Checking for unpackaged file(s): /us
Hello, All
When I take a SPEC file from kamailiotar.gz - during rpmbuild I encounter:
Checking for unpackaged file(s): /usr/lib/rpm/check-files
/root/rpmbuild/BUILDROOT/kamailio-4.3.4-0.0.el6.x86_64error: Installed (but
unpackaged) file(s) found: /usr/lib64/kamailio/modules/auth_xkeys.so
I see :
ERROR: [tcp_main.c:2790]: tcp_init(): bind(9, 0x7fd50bd8ee34, 16) on
127.0.0.1:5060 : Address already in use
But I commented out all TCP (listen TCP) so why is this error happen?
On Friday, September 9, 2016 10:52 AM, ycaner
wrote:
Hello;
it is clear that kamailio crashs. Co
Hello;
it is clear that kamailio crashs. Could you start with "kamailio -E -ddd"
and then see logs. it gives hit. Probably libraries has some conflicts.
--
View this message in context:
http://sip-router.1086192.n5.nabble.com/help-with-kamailio-rpm-made-from-source-tp151601p151626.html
Sent fr
Thank you very much, I did it this way, and it worked perfect!!!
if (is_subscriber("$ru", "subscriber", "2")) {
...
logic
...
}
2016-06-13 3:05 GMT-06:00 Daniel-Constantin Mierla :
> Hello,
>
> you have to show the request_route block with the part where the
> route(DISPATCH) is e
Hello,
you have to show the request_route block with the part where the
route(DISPATCH) is executed. You can use is_subscriber() to see if the
target number is a local subscriber and route via location.
Cheers,
Daniel
On 10/06/16 03:39, pablo rosales wrote:
> Hi everyone! I am a newbie with Kam
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
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
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
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
[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
Hi,
I think the best guide closest to your description is here :
http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb
Here is what you need to do. (*Besides mentioning what you tried and what
problems were faced*).
1 - Configure kamailio to use the DB schema where your use
Hello Kevin,If I understood properly you want to build a system which
authenticates users and routes the Asterisk servers for communication.
First, Kamailio supports the routing, balancing and authentication. For example
we use Kamailio and Freeswitch. Here the how its work:We have 1 Kamailio ser
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
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
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
Specifically, after auth_check line add:
xlog("The return code is $rc\n");
Can add additional lines to view the values of other pseudovariables
http://www.kamailio.org/wiki/cookbooks/4.0.x/pseudovariables
On 3/9/2015 7:56 AM, Daniel-Constantin Mierla wrote:
On 09/03/15 15:41, Agiftel wrote:
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
On 09/03/15 15:41, Agiftel wrote:
> Thanks Olle for reply but password is correct.
Set the debug=3 in kamailio.cfg and check the log messages, you should
get more hints for what part mismatches there.
Also, after doing auth_check(), you can print the $rc to see the return
code value.
Cheers,
Da
Thanks Olle for reply but password is correct.
--
View this message in context:
http://sip-router.1086192.n5.nabble.com/Help-with-407-Proxy-Auth-Required-tp136043p136045.html
Sent from the Users mailing list archive at Nabble.com.
___
SIP Express Ro
On 09 Mar 2015, at 15:07, Agiftel wrote:
> 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?
This is typical - it happens when the password is wrong e
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
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
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
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
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,
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
On 02/09/14 19:05, Alex Villacís Lasso wrote:
El 02/09/14 05:17, Daniel-Constantin Mierla escribió:
If you get signling routed ok but no audio, then you have problems
bridging rtp stream.
Most probably you need to use rtpproxy (eventually with advertise
address (there is a patch or use seco
El 02/09/14 05:17, Daniel-Constantin Mierla escribió:
If you get signling routed ok but no audio, then you have problems bridging rtp
stream.
Most probably you need to use rtpproxy (eventually with advertise address
(there is a patch or use second parameter for rtpproxy_manage())) to bridge.
If you get signling routed ok but no audio, then you have problems
bridging rtp stream.
Most probably you need to use rtpproxy (eventually with advertise
address (there is a patch or use second parameter for
rtpproxy_manage())) to bridge.
I never used sip-natting in kernel, so I am not aware
El 01/09/14 10:50, Alex Villacís Lasso escribió:
El 01/09/14 05:15, Daniel-Constantin Mierla escribió:
On 29/08/14 23:58, Andres wrote:
On 8/29/14, 1:42 PM, Alex Villacís Lasso wrote:
Please consider the following SIP packet exchange, as seen by a tcpdump running
on 201.234.196.170. Here 1
El 01/09/14 05:15, Daniel-Constantin Mierla escribió:
On 29/08/14 23:58, Andres wrote:
On 8/29/14, 1:42 PM, Alex Villacís Lasso wrote:
Please consider the following SIP packet exchange, as seen by a tcpdump running
on 201.234.196.170. Here 198.58.101.75 initiates a call to 201.234.196.170:
On 29/08/14 23:58, Andres wrote:
On 8/29/14, 1:42 PM, Alex Villacís Lasso wrote:
Please consider the following SIP packet exchange, as seen by a
tcpdump running on 201.234.196.170. Here 198.58.101.75 initiates a
call to 201.234.196.170:
IP 198.58.101.75.5060 > 201.234.196.170.5060
INVITE si
On 8/29/14, 1:42 PM, Alex Villacís Lasso wrote:
Please consider the following SIP packet exchange, as seen by a
tcpdump running on 201.234.196.170. Here 198.58.101.75 initiates a
call to 201.234.196.170:
IP 198.58.101.75.5060 > 201.234.196.170.5060
INVITE sip:*43@201.234.196.170:5060 SIP/2.0
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
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
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;
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?
>
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
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
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
You need to add user to Kamailio with username and password by using
kamctl. Please try UDP on JITSI, if you still have problems please collect
wireshark logs and send to us.
Thanks,
Veera
On Tue, Jul 8, 2014 at 2:39 AM, Salman Zafar wrote:
> Hi,
>Is your soft-phone packets reaching your k
Hi,
Is your soft-phone packets reaching your kamailio server?, if so what
happens to REGISTER packet?.
On Tue, Jul 8, 2014 at 1:28 PM, Jayaraman, Kamalakannan <
kamalakannan.jayara...@pearson.com> wrote:
> Hi,
>I had installed kamailio on my AWS EC2, by following the steps in
> http://kam
Hello,
what you can do is to route the messages to the other proxy if there is
no contacts in the local location table, something like:
if(!lookup("location") {
if(src_ip==_THE_OTHER_KAMAILIO_IP_) {
send_reply("404", "Not found");
exit;
}
$du = "sip:_THE_OTHER_KAMAI
Indeed, of course that way works, but I am pretty sure that Kamailio can
intercept and give the right response.
Right now what would be needed is to make a complete SIP Notify with the
according digest, using the password picked from the database, and send it
back. The answer would be a 200 'OK' .
On Thu, Mar 13, 2014 at 8:26 AM, Pedro Niño wrote:
> The other (ugly) option, is to remove the auth from the phone, for the Sip
> Provisioning, but that would leave and open door to a reboot attack without
> auth needed from any IP. And I dont like that option.
>
This might not be as bad of an o
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
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,
On 10/24/13 3:49 PM, anfecora wrote:
Hi all, can anyone help me to find out what is wrong with my setup, i
have an asterisk behind a kamailio, kamailio is proxying all packages
to the outside.
when the call is bridge it gets disconnected after a few seconds, it
seems that our voip ca
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
I am not sure about your situation, but in my case - Asterisk respond to
any message, and wrong paths were corrected the way I showed.
By the way - I am using also rtpproxy. Although it should not interfere
here at all.
I used wireshark on Asterisk and on Kamailio servers to find what exactly
happe
Thank you Stoyan, i tried but i ended up creating a loop with the carrier,
i believe this is more a asterisk receiving the package and ignoring the
record-route and because i am just proxying the signalling it does ack to
the contact, i have to find a way to tell asterisk that answer everything
to
I had same problem - with BYE also.
My "go around" was (replaced name of domain and IP of kamailio):
route[ACKBYE] {
#!ifdef WITH_MYFORWARD
if(($sht(forw=>$ft))=~$td){
$du=$sht(forw=>$ft);
}else if((($td=="name.of.company.com")||($td=="ip"))&&($si=="ip")){
$du=$sht(forw=>$ft);
return;
}
#!endif
re
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
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,
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
Have you done:
make prepare
and set the right owner for the files inside siremis? Check:
http://kb.asipto.com/siremis:install40x:main#local_configuration
Regards,
Ramona
On 5/15/13 3:01 AM, Future Lian wrote:
I can't install siremis 3.0 for kaimailio after install kamailio, It
take a long l
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
Brad Johns writes:
> Do you have "method_filtering" in registrar params set to 0 or 1?
i have
modparam("registrar", "method_filtering", 1)
and it worked ok with tryit client.
-- juha
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mai
Peter Dunkley writes:
> There is a much simpler WebSocket Kamailio configuration file in the
> examples directory in the source tree:
> http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=blob;f=examples/welcome.cfg;h=4bde0ae71be6f8da6a5bd3089d5af57569cc0178;hb=HEAD
the link to websocket
Sorry,
Wrong link. The correct one is:
http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=blob;f=examples/websocket.cfg;h=4176af0a86985dc88d768b31f4ebe4021abb093f;hb=HEAD
Peter
> Hi,
>
> There is a much simpler WebSocket Kamailio configuration file in the
> examples directory in the so
Hi,
There is a much simpler WebSocket Kamailio configuration file in the
examples directory in the source tree:
http://git.sip-router.org/cgi-bin/gitweb.cgi?p=sip-router;a=blob;f=examples/welcome.cfg;h=4bde0ae71be6f8da6a5bd3089d5af57569cc0178;hb=HEAD
It doesn't have accounting or any of the other
Very interesting. I am still not able to have an audio call complete. Can
I see your kamailio.cfg, under separate cover? Or sent to the list?
Do you have "method_filtering" in registrar params set to 0 or 1? I had it
set to 1 and by default the JsSIP tryit must not have been sending an Allow
o
Peter,
Thank you. By changing the "method_filtering" modparam to 0 (it was
actually 1), I am now able to make it past this, and the INVITE is
processed over WS transport. However, the audio call is still not
completing.
I am seeing a "180 Ringing" message for a while, followed by a "408 Request
Hello,
In SIP you can put an Allow: header in REGISTER requests to say which
methods the registering end-point is capable of receiving.
If you get a -2 returned from lookup() it means that the method for the
request (in this case INVITE) was not in the "Allow:" header in the
REGISTER.
You can ch
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