Dear All,
Greetings,
I am working Kamailio (4.0.1) server with GUI package Siremis (4.0) of it.
I was successful in get through Calls and SMS features in our VoIP phones
using this kamailio sever. And now i am trying to get GUI for my kamailio
server with SIREMIS (4.0.0) package. Installation was good as i am getting
subscribers list in GUI which are in MySQL database. As it comes accounting
section. i am Failed to get the CDR list when the call session between two
clients have been done. I gone through Acc module's documentation and
followed the following links to set up the accounting in GUI :
http://kb.asipto.com/siremis:install40x:accounting.
And i read this is description in accounting module documentation :
'Note that CDR generation does not involve any kind of database storage
(yet). In order to persist the CDRs into a database you will have to set up
an exterior process (i.e., a script living outside of Kamailio) and
implement the storage task yourself'.
So what does it mean ? should i have to install external process like CDR
tool and Freeradius?
Or is there any other way to get the CDR related information in GUI using
database backend.
Please find below my kamailio configuration file.
By the way CDR data is logged to syslog fine, but i dont know where is
wrong for database backend.
Any help will greatly appreciate.
Regards,
Nandini
Hey everyone, currently i have this configuration set on kamailio:
modparam("dispatcher", "list_file", "/etc/kamailio/dispatcher.list")
modparam("dispatcher", "force_dst", 0) #forzado de la reescritura direccion
de destino
modparam("dispatcher", "flags", 2) #banderas de funcionamiento, 2 significa
"soporte para failover"
modparam("dispatcher", "dst_avp", "$avp(dsdst)")
modparam("dispatcher", "grp_avp", "$avp(dsgrp)")
modparam("dispatcher", "cnt_avp", "$avp(dscnt)")
modparam("dispatcher", "ds_ping_method", "OPTIONS")
modparam("dispatcher", "ds_ping_interval", 5) #tiempo que transcurre antes
de verificar nuevamente una salida inactiva
modparam("dispatcher", "ds_probing_threshhold", 5) #Numero de intentos
antes de marcar una salida como inactiva
modparam("dispatcher", "ds_ping_reply_codes",
"class=2;code=403;code=488;class=3")
modparam("dispatcher", "ds_probing_mode", 1)
modparam("dispatcher", "ds_hash_expire", 3600)
modparam("dispatcher", "ds_hash_initexpire", 60)
#loadmodule "dispatcher.so"
####### Routing Logic ########
# Main SIP request routing logic
# - processing of any incoming SIP request starts with this route
# - note: this is the same as route { ... }
request_route {
if (is_method("SUBSCRIBE")){
if (src_ip == 192.168.2.1 | src_ip == 192.168.2.2){
t_relay();
}
route(REGISTRAR);
}
if (src_ip == 192.168.2.1 | src_ip == 192.168.2.2){
t_relay();
}
else{
route(ASTERISK);
}
}
route[ASTERISK]{
ds_select_dst("1", "8");
t_relay();
exit();
}
Everything else is left with the same basic configuration it had when i
installed the software.
I want to use it kinda of a sip router, so what Kamailio does is just
forward the packets betwen my Asterisk boxes and the Sip Phones. Currently
it seems to work (partially) but i have doubts about if this is correctly
done (it's the first time using Kamailio and i need this working withing a
week at most).
As you can see i use the module dispatcher for failover/failback (this is
the purpose of using Kamailio, a failover/failback setup). One major
problem i've found with this setup is, if the phones are currently
connected and working with one of my 2 asterisk boxes and if that box
fails, Kamailio starts sending the traffic to the second box (as intended),
but the phones don't try to subscribe to the new asterisk box, rather they
just keep sending traffic (and obviously kamailio forwarding it).
Sometimes one of the phones subscribe to the new box, but that's not always
the case, the packets reach the new Asterisk box, but since the phones
aren't registered to it, they can't make calls.
Other times the behavior of the setup is rather weird, i can call some
extensions and some others i can not (even though they are registered
within the asterisk box) The traffic gets to the Asterisk box (as shown in
the asterisk logs) but the call is shown as "service unavailable". I've
checked a lot of times the Asterisk setup and it seems to be fine so i
think it has something to do with Kamailio.
I know this might be a really bad configuration file, but it's been at most
3 days since i started using Kamailio which i sometimes find kinda hard to
understand and i really need this working within a week.
To summarize all i want is kamailio forwarding packets between the
currently active server and the phones so if it fails, then the packets go
to the second one, the phones must re-subscribe to the new active server.
Hello,
anyone out there using kamailio and oracle database in production system?
How's your experience on that?
We may be forced to migrate from mysql to oracle dbms.
Mit freundlichen Grüßen / Best regards
Marco Barthel
Robert Bosch GmbH
(CI/AFU1)
Postfach 30 02 20
70442 Stuttgart
GERMANY
www.bosch.com<http://www.bosch.com>
Marco.Barthel(a)de.bosch.com<mailto:Marco.Barthel@de.bosch.com>
Sitz: Stuttgart, Registergericht: Amtsgericht Stuttgart, HRB 14000;
Aufsichtsratsvorsitzender: Franz Fehrenbach; Geschäftsführung: Dr. Volkmar Denner,
Dr. Stefan Asenkerschbaumer, Dr. Rolf Bulander, Dr. Stefan Hartung, Dr. Dirk Hoheisel, Christoph Kübel,
Uwe Raschke, Wolf-Henning Scheider, Dr. Werner Struth, Peter Tyroller
Hi there,
I am trying to implement a sip/rtp proxy.
I have one public ip address with a domain name. example (whatever.com)
On the lan side i have multiple asterisk instances all listening on 5060
for sip requests. These instances will be assigned to different customers
I am hoping to run kamailio in front of these instances and route sip and
rtp traffic to the correct asterisk instance.
So in this use case asterisk instances will be used as application servers
and kamailio will be used as a sip/rtpproxy with nathelper
So my question is what would be the best way and using which modules can i
route the requests for customerone.whatever.com to the first asterisk
instance and customertwo.whatever.com to the second.
is this achieved with the dispatcher module? and if so, where can it be
defined that requests for cusomertone.whatever.com is routed to the first
asterisk instance.
Kind regards,
Cormac O'Loughlin
Hello.
I'm trying to configure kamailio as a gateway between Websocket and
TCP/TLS transports.
When I call record_route() for an initial INVITE that comes via WS and
will be forwarded via TCP to a registered UA kamailio inserts only a one
record-route header with its IP and transport=ws instead of two
record-route headers with both incoming/outgoing transports.
This behaviour breaks in-dialog requests routing.
rr module parameters are:
modparam("rr", "enable_full_lr", 1)
modparam("rr", "append_fromtag", 0)
modparam("rr", "enable_double_rr", 1)
I use:
- kamailio-4.0.2
- sipml5 as sip client
Can anybody point me in the right direction to understand why it happens ?
Sorry, this test was done on a public server that's why it was showing
public IPs. I was trying out opensips and kamailio and same response for
both. Do you think this is a network firewall issue?
On Tue, Jul 30, 2013 at 2:31 PM, Gertjan Wolzak <g.wolzak(a)foize.com> wrote:
> Hello Jason,****
>
> ** **
>
> Well, as you can see now the error is:****
>
> ** **
>
> U 2013/07/29 18:24:03.328456 198.23.160.81:5060 -> 49.144.184.97:41998
> SIP/2.0 477 Send failed (477/TM).
> Via: SIP/2.0/UDP 192.168.0.107:41998
> ;received=49.144.184.97;branch=z9hG4bK-d8754z-1cfc096228010333-1---d8754z-;rport=41998.
> To: <sip:639195015475@198.23.160.81
> >;tag=eb0b3ccde0b882baee99bc071578cb61-d8d9.
> From: "639178864952"<sip:639178864952@198.23.160.81>;tag=1b3b463c.
> Call-ID: NjhmYzk0ODZhNWVmOGE5MTU4ZWExM2Q5NjI1NDVhODg.
> CSeq: 2 INVITE.
> Server: OpenSIPS (1.8.3-notls (x86_64/linux)).
> Content-Length: 0.****
>
> ** **
>
> But as I am not a guru at all and this is a Kamailio users mailing list
> and not an opensips list….****
>
> ** **
>
> I just googled the error, found that for Opensips and Kamailio it has
> something to do with not being able to make a tcp connection.****
>
> ** **
>
> So, check the port translations, make a tcpdump of wireshark dump and see
> if what is going wrong.****
>
> ** **
>
> You said that the clients and Kamailio were on the same lan, but from the
> logging you can see that you are on the internet…….****
>
> ** **
>
> Hope this helps a bit.****
>
> ** **
>
> Good luck.****
>
> ** **
>
> ** **
>
> *From:* sr-users-bounces(a)lists.sip-router.org [mailto:
> sr-users-bounces(a)lists.sip-router.org] *On Behalf Of *Jason Sia
> *Sent:* maandag 29 juli 2013 18:16
> *To:* Kamailio (SER) - Users Mailing List
> *Subject:* Re: [SR-Users] kamailio call problem****
>
> ** **
>
> Hi,****
>
> Thanks for the advice. Here is the ngrep logs:
>
> U 2013/07/29 18:23:52.994331 49.144.184.97:41998 -> 198.23.160.81:5060
> INVITE sip:639195015475@198.23.160.81 SIP/2.0.
> Via: SIP/2.0/UDP 192.168.0.107:41998;branch=****
>
> z9hG4bK-d8754z-2999840441d3d05a-1---d8754z-;rport.
> Max-Forwards: 70.
> Contact: <sip:639178864952@49.144.184.97:41998>.
> To: <sip:639195015475@198.23.160.81>.
> From: "639178864952"<sip:639178864952@198.23.160.81>;tag=1b3b463c.
> Call-ID: NjhmYzk0ODZhNWVmOGE5MTU4ZWExM2Q5NjI1NDVhODg.
> CSeq: 1 INVITE.
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
> SUBSCRIBE, INFO.
> Content-Type: application/sdp.
> Supported: replaces.
> User-Agent: X-Lite release 4.5.3 stamp 70576.
> Content-Length: 357.
> .
> v=0.
> o=- 1375107832761441 1 IN IP4 192.168.0.107.
> s=X-Lite 4 release 4.5.3 stamp 70576.
> c=IN IP4 192.168.0.107.
> t=0 0.
> m=audio 53084 RTP/AVP 123 9 0 8 97 100 98 101.
> a=rtpmap:123 opus/48000/2.
> a=fmtp:123 useinbandfec=1.
> a=rtpmap:97 speex/8000.
> a=rtpmap:100 speex/16000.
> a=rtpmap:98 ILBC/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-15.
> a=sendrecv.
>
>
> U 2013/07/29 18:23:52.994422 198.23.160.81:5060 -> 49.144.184.97:41998
> SIP/2.0 407 Proxy Authentication Required.
> Via: SIP/2.0/UDP 192.168.0.107:41998
> ;received=49.144.184.97;branch=z9hG4bK-d8754z-2999840441d3d05a-1---d8754z-;rport=41998.
> To: <sip:639195015475@198.23.160.81
> >;tag=80403214130c49515b8f7d7842a4c119.ef65.
> From: "639178864952"<sip:639178864952@198.23.160.81>;tag=1b3b463c.
> Call-ID: NjhmYzk0ODZhNWVmOGE5MTU4ZWExM2Q5NjI1NDVhODg.
> CSeq: 1 INVITE.
> Proxy-Authenticate: Digest realm="198.23.160.81",
> nonce="51f67b16000000fae8c70592af760c78fa39cd43f84c1a6b".
> Server: OpenSIPS (1.8.3-notls (x86_64/linux)).
> Content-Length: 0.
> .
>
>
> U 2013/07/29 18:23:53.299025 49.144.184.97:41998 -> 198.23.160.81:5060
> ACK sip:639195015475@198.23.160.81 SIP/2.0.
> Via: SIP/2.0/UDP 192.168.0.107:41998
> ;branch=z9hG4bK-d8754z-2999840441d3d05a-1---d8754z-;rport.
> Max-Forwards: 70.
> To: <sip:639195015475@198.23.160.81
> >;tag=80403214130c49515b8f7d7842a4c119.ef65.
> From: "639178864952"<sip:639178864952@198.23.160.81>;tag=1b3b463c.
> Call-ID: NjhmYzk0ODZhNWVmOGE5MTU4ZWExM2Q5NjI1NDVhODg.
> CSeq: 1 ACK.
> Content-Length: 0.
> .
>
>
> U 2013/07/29 18:23:53.317842 49.144.184.97:41998 -> 198.23.160.81:5060
> INVITE sip:639195015475@198.23.160.81 SIP/2.0.
> Via: SIP/2.0/UDP 192.168.0.107:41998
> ;branch=z9hG4bK-d8754z-1cfc096228010333-1---d8754z-;rport.
> Max-Forwards: 70.
> Contact: <sip:639178864952@49.144.184.97:41998>.
> To: <sip:639195015475@198.23.160.81>.
> From: "639178864952"<sip:639178864952@198.23.160.81>;tag=1b3b463c.
> Call-ID: NjhmYzk0ODZhNWVmOGE5MTU4ZWExM2Q5NjI1NDVhODg.
> CSeq: 2 INVITE.
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
> SUBSCRIBE, INFO.
> Content-Type: application/sdp.
> Proxy-Authorization: Digest
> username="639178864952",realm="198.23.160.81",nonce="51f67b16000000fae8c70592af760c78fa39cd43f84c1a6b",uri="
> sip:639195015475@198.23.160.81
> ",response="1605e4e6b20dbbd6be5917c94dd3961e",algorithm=MD5.
> Supported: replaces.
> User-Agent: X-Lite release 4.5.3 stamp 70576.
> Content-Length: 357.
> .
> v=0.
> o=- 1375107832761441 1 IN IP4 192.168.0.107.
> s=X-Lite 4 release 4.5.3 stamp 70576.
> c=IN IP4 192.168.0.107.
> t=0 0.
> m=audio 53084 RTP/AVP 123 9 0 8 97 100 98 101.
> a=rtpmap:123 opus/48000/2.
> a=fmtp:123 useinbandfec=1.
> a=rtpmap:97 speex/8000.
> a=rtpmap:100 speex/16000.
> a=rtpmap:98 ILBC/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-15.
> a=sendrecv.
>
>
> U 2013/07/29 18:23:53.318191 198.23.160.81:5060 -> 49.144.184.97:41998
> SIP/2.0 100 Giving a try.
> Via: SIP/2.0/UDP 192.168.0.107:41998
> ;received=49.144.184.97;branch=z9hG4bK-d8754z-1cfc096228010333-1---d8754z-;rport=41998.
> To: <sip:639195015475@198.23.160.81>.
> From: "639178864952"<sip:639178864952@198.23.160.81>;tag=1b3b463c.
> Call-ID: NjhmYzk0ODZhNWVmOGE5MTU4ZWExM2Q5NjI1NDVhODg.
> CSeq: 2 INVITE.
> Server: OpenSIPS (1.8.3-notls (x86_64/linux)).
> Content-Length: 0.
> .
>
>
> U 2013/07/29 18:24:03.328456 198.23.160.81:5060 -> 49.144.184.97:41998
> SIP/2.0 477 Send failed (477/TM).
> Via: SIP/2.0/UDP 192.168.0.107:41998
> ;received=49.144.184.97;branch=z9hG4bK-d8754z-1cfc096228010333-1---d8754z-;rport=41998.
> To: <sip:639195015475@198.23.160.81
> >;tag=eb0b3ccde0b882baee99bc071578cb61-d8d9.
> From: "639178864952"<sip:639178864952@198.23.160.81>;tag=1b3b463c.
> Call-ID: NjhmYzk0ODZhNWVmOGE5MTU4ZWExM2Q5NjI1NDVhODg.
> CSeq: 2 INVITE.
> Server: OpenSIPS (1.8.3-notls (x86_64/linux)).
> Content-Length: 0.
> .
>
>
> U 2013/07/29 18:24:03.582094 49.144.184.97:41998 -> 198.23.160.81:5060
> ACK sip:639195015475@198.23.160.81 SIP/2.0.
> Via: SIP/2.0/UDP 192.168.0.107:41998
> ;branch=z9hG4bK-d8754z-1cfc096228010333-1---d8754z-;rport.
> Max-Forwards: 70.
> To: <sip:639195015475@198.23.160.81
> >;tag=eb0b3ccde0b882baee99bc071578cb61-d8d9.
> From: "639178864952"<sip:639178864952@198.23.160.81>;tag=1b3b463c.
> Call-ID: NjhmYzk0ODZhNWVmOGE5MTU4ZWExM2Q5NjI1NDVhODg.
> CSeq: 2 ACK.
> Content-Length: 0.****
>
> ****
>
> ** **
>
>
> Thanks,
> Jason****
>
> ** **
>
> On Mon, Jul 29, 2013 at 3:33 PM, Gertjan Wolzak <g.wolzak(a)foize.com>
> wrote:****
>
> Hello Jason,****
>
> ****
>
> What you should do is on the Kamailio server do an ngrep, that
> information will make it possible for the people on the list to take a look
> and help you.****
>
> ****
>
> By only telling your experiences it is hard to help you.****
>
> ****
>
> Just give the command: ngrep –qt –W byline port 5060 and then do the
> test calls, so from android to xlite and vice versa.****
>
> ****
>
> Save the info and paste it in the case.****
>
> ****
>
> Good luck.****
>
> ****
>
> Gertjan****
>
> ****
>
> *From:* sr-users-bounces(a)lists.sip-router.org [mailto:
> sr-users-bounces(a)lists.sip-router.org] *On Behalf Of *Jason Sia
> *Sent:* zondag 28 juli 2013 10:52
> *To:* sr-users(a)lists.sip-router.org
> *Subject:* [SR-Users] kamailio call problem****
>
> ****
>
> Hi,****
>
> I installed kamailio v 3.1.6 rpm. I used the default configuration.
> Kamailio started. I have two clients one is an android phone using native
> sip client, and the other one x-lite. I can call x-lite to phone but not
> the other way around. When I restarted kamailio, I can call phone to
> x-lite but not the other way around. The first one I did when I restart
> kamailio will be the only leg that will be working, then I have to restart
> again if I want to do the reverse. Has anyone encountered this problem?
> How do I fix it? I'm on a LAN, kamailio server and the two clients are on
> the same LAN.****
>
> ****
>
> Thanks,
> Jason****
>
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users(a)lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users****
>
> ** **
>
> _______________________________________________
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users(a)lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
Hi Lucky Santiago,
May I know how installed php with pdo-mysql. I kindly request to share steps
to install php with pdo-mysql.
Regards,
Prem Chandiran M
-----Original Message-----
From: sr-users-bounces(a)lists.sip-router.org
[mailto:sr-users-bounces@lists.sip-router.org] On Behalf Of
sr-users-request(a)lists.sip-router.org
Sent: Wednesday, July 31, 2013 03:30 PM
To: sr-users(a)lists.sip-router.org
Subject: sr-users Digest, Vol 98, Issue 128
Send sr-users mailing list submissions to
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When replying, please edit your Subject line so it is more specific than
"Re: Contents of sr-users digest..."
Today's Topics:
1. Kamailio4.0 + Siremis4.0 (Lucky Santiago)
2. siremis login issue (Premchandiran)
3. websocket double record_route() (Vitaliy Aleksandrov)
----------------------------------------------------------------------
Message: 1
Date: Wed, 31 Jul 2013 14:57:58 +0800
From: Lucky Santiago <lucky(a)luckysantiago.com>
To: sr-users(a)lists.sip-router.org
Subject: [SR-Users] Kamailio4.0 + Siremis4.0
Message-ID: <51F8B576.8060306(a)luckysantiago.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Sorry for being a newbie, I have newly installed Kamailio 4.0 + Siremis
4.0 and i have setup until properly registering extension/subscriber, what i
want to attain for now is a simple exten -> kamailio/siremis ->
carrier/telco that includes dialplan and trunking, it will really help if
there is some sort of documentation managing it via the web interface or
siremis. Thanks.
------------------------------
Message: 2
Date: Wed, 31 Jul 2013 12:46:39 +0530
From: "Premchandiran" <premchandiran.marimuthu(a)plintron.com>
To: <sr-users(a)lists.sip-router.org>
Subject: [SR-Users] siremis login issue
Message-ID: <001901ce8dbd$e67dc2e0$b37948a0$@marimuthu@plintron.com>
Content-Type: text/plain; charset="us-ascii"
Hi All,
After installing in siremis when I try to login using admin and admin as
username and pwd a pop window opens in that "SQLSTATE[HY000] [2002] No such
file or directory" . and in error log of apache2 I am getting below error "
[client 192.168.137.7:4758] AH01630: client denied by server configuration:
/opt/siremis-4.0.0/siremis/.htaccess"
I kindly request you all to help me in resolving this issue.
Thanks and Regards,
Prem Chandiran M
Hi All,
After installing in siremis when I try to login using admin and admin as
username and pwd a pop window opens in that "SQLSTATE[HY000] [2002] No such
file or directory" . and in error log of apache2 I am getting below error "
[client 192.168.137.7:4758] AH01630: client denied by server configuration:
/opt/siremis-4.0.0/siremis/.htaccess"
I kindly request you all to help me in resolving this issue.
Thanks and Regards,
Prem Chandiran M
Sorry for being a newbie, I have newly installed Kamailio 4.0 + Siremis
4.0 and i have setup until properly registering extension/subscriber,
what i want to attain for now is a simple exten -> kamailio/siremis ->
carrier/telco that includes dialplan and trunking, it will really help
if there is some sort of documentation managing it via the web interface
or siremis. Thanks.
