Hello,
I'm trying to migrate an existing configuration from OpenSIPS 1.4.x to Kamailio 3.2.x. The old configuration replicates REGISTER requests to multiple servers using this code in the REGISTER routing block:
add_sock_hdr("Local-Sock");
add_rcv_param();
append_branch("sip:1.1.1.1:5060");
t_replicate("sip:2.2.2.2:5060");
exit;
From what I read in the documentation, the behaviour of append_branch() is different in Kamailio, so I modified my code like this:
add_sock_hdr("Local-Sock");
add_rcv_param();
append_branch();
seturi(sip:1.1.1.1:5060");
t_replicate("sip:2.2.2.2:5060");
exit;
Unfortunately, I do not see any REGISTER requests arriving on 1.1.1.1 so I assume that I made a mistake when forking the request. Can anyone point me in the right direction?
Thanks,
Henning Holtschneider
--
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tel +49 231 91596-25, fax +49 231 91596-55
sip 25(a)voip.loca.net
Registergericht Amtsgericht Dortmund HRA 14208
Geschäftsführer Sven Haufe, Henning Holtschneider
Hi Konstantin,
Thanks for your kind reply. You can check the iptables output at : http://pastebin.com/i3zUfVeb.
I hope that this will give you enough clue in right direction.
With Regards,
Vijay Thakur
======================================================================================================================
Message: 3
Date: Fri, 3 Aug 2012 17:38:23 +0300
From: "Konstantin M."<evilzluk(a)gmail.com>
Subject: Re: [SR-Users] kernel: nf_ct_sip: dropping packetIN= OUT=eth0
Error
To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) -
Users Mailing List"<sr-users(a)lists.sip-router.org>
Message-ID:
<CAEYtTzdeY88X-GC+wDwLou+smsB60h+M36h7xwCu448V0YtD2g(a)mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
Looks like your firewall is passing a state RELATED,ESTABLISHED and don't
have a permit rule for a state NEW.
Check your iptables or simply do a command: iptables-save >
/tmp/iptables.txt and paste this file (/tmp/iptables.txt) to pastebin
2012/8/3 Vijay Thakur<vijay.thakur(a)loopmethods.com>
> Hello all,
>
> I have configure Kamailio 3.1.5 Server. All things are working fine. When
> i make a call from Soft phone (X-Lite) to iphone, all is working fine. But
> in other case call from iphone to Softphone is not working, even not
> ringing. During checking the logs i am getting the error:
>
> Aug 3 04:36:09 localhost kernel: nf_ct_sip: dropping packetIN=eth0 OUT=
> MAC=f2:3c:91:ae:92:36:c8:4c:**75:f5:c4:ff:08:00 SRC=122.xxx.xxx.77
> DST=xx.116.xx.23 LEN=1482 TOS=0x00 PREC=0x00 TTL=51 ID=50183 DF PROTO=TCP
> SPT=15587 DPT=5060 SEQ=3285635734 ACK=3113844065 WINDOW=5763 RES=0x00 ACK
> URGP=0 OPT (0101080A000E20610932B25A)
>
> This is server is hosted on Linnode.
>
> Kindly guide me to solve the problem
>
> --
> Best Regards,
>
> Vijay Thakur
> (Assistant Manager - Networks)
> Mobile : +91 8744018065
> Mail :vijay.thakur@loopmethods.com
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Hello! I'm new to this list and Kamailio.
First, thanks to all the programmers that developed this software.
I'd like to ask several questions on how to get started with Kamailio
and the basic understanding of it. I'm trying to figure out Kamailio
only since yesterday. Right now I have a single Asterisk server that
handles signaling and media. For accounting and user administration I'm
using a2billing with MySQL on it. For redundancy and scalability I would
like to create the following setup:
France:
BGP anycast 195.5.5.0/24
2 nodes with Kamailio (IP 195.5.5.5) active + standby
2 nodes with Asterisk + a2billing + DRBD for active-active MySQL cluster.
Spain:
BGP anycast 195.5.5.0/24
2 nodes with Kamailio (IP 195.5.5.5) active + standby
2 nodes with Asterisk + a2billing + DRBD for active-active MySQL cluster.
(a2billing MySQL DB will be always synchronous over all nodes in Spain
and France)
This will give me:
- Network redundancy (data center burns down in France, I stay online in
Spain). This will be handled by BGP anycast.
- Signaling redundancy (Kamailio active + standby). This will be handled
by Linux-HA or something like that.
- Media redundancy (2x Asterisk per country). This will be handled by
Kamailio dispatcher.
- Application/database redundancy (2x a2billing per country +
synchronous MySQL DB everyhwere). This will be handled by DRBD and MySQL.
= High redundancy, I can sleep and it can scale.
What I would like to achieve besides the above:
- Give only a single IP address to all customers and termination
providers (the same IP address), 195.5.5.5.
I'm a SIP noob, so I have to ask:
- How do I do the Kamailio part? ;-) ... I have seen this how-to for
Kamailio/Asterisk realtime:
http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb
But it feels like "overkill". If I would not have any users that
actually REGISTER (e.g. in a pure wholesale termination environment), I
would not need Kamailio/Asterisk realtime integration. Correct?
- Is there no way around changing the Asterisk side (activating
realtime, new MySQL DB) when I have users that do need to REGISTER? If I
would not be using a2billing I could probably handle all the
registrations in Kamailio only?
- Since Kamailio and Asterisk will not be on the same box, what is the
recommended way for Kamailio securely communicating with the MySQL
database on the Asterisk server? Does Kamailio support SSL with MySQL?
- If I use RTPproxy on the Kamailio server, every customer and
termination provider would connect to 1 single IP address, because both
media and signaling comes from that IP. Correct?
- If I don't use RTPproxy, and have canreinvite=yes on my Asterisk
servers, customers would get the media, when placing PSTN calls,
directly from my termination providers (I would like to avoid that).
Correct?
- If I don't use RTPproxy, and have canreinvite=no on my Asterisk
servers, customers will get the media directly from my Asterisk servers,
but termination providers that filter based on IP addresses they would
have to allow all Asterisk IP addresses in their filters (same for
customers, actually). Correct?
Right now I didn't have to worry about such things because media and
signaling were handled by a single Asterisk box with canreinvite=no
everywhere.
- Last question for now: why does it seem like important developers of
SIP software such as Kamailio and yate are originating from Romania and
are female? Just a coincidence? :-)
Regards
Markus
Hi All,
Is there any possibility of sending RTP over TCP, because we are building a
softphone based on PJSIP, but seems that some ISP are blocking RTP packets
over UDP. Any solutions for this.
Regards
Gloria
Hello all,
I have configure Kamailio 3.1.5 Server. All things are working fine.
When i make a call from Soft phone (X-Lite) to iphone, all is working
fine. But in other case call from iphone to Softphone is not working,
even not ringing. During checking the logs i am getting the error:
Aug 3 04:36:09 localhost kernel: nf_ct_sip: dropping packetIN=eth0 OUT=
MAC=f2:3c:91:ae:92:36:c8:4c:75:f5:c4:ff:08:00 SRC=122.xxx.xxx.77
DST=xx.116.xx.23 LEN=1482 TOS=0x00 PREC=0x00 TTL=51 ID=50183 DF
PROTO=TCP SPT=15587 DPT=5060 SEQ=3285635734 ACK=3113844065 WINDOW=5763
RES=0x00 ACK URGP=0 OPT (0101080A000E20610932B25A)
This is server is hosted on Linnode.
Kindly guide me to solve the problem
--
Best Regards,
Vijay Thakur
(Assistant Manager - Networks)
Mobile : +91 8744018065
Mail : vijay.thakur(a)loopmethods.com
Loop IT Methods Private Limited
1st Floor, B-10, Sector-7, Noida, (U.P) India
Ph: +91 120 305 3481,82 (INDIA), +1 347 468 8631 (USA), +61 390 011 178 (AUS)
Fax: +91 971 728 330
Web: www.loopmethods.com
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Hi,
one number is registered on 2 phones. Phone1 has Always redirect set
to another number. When incoming call is initiated, Phone2 is ringing
and Phone1 sends 302 to the proxy. However the proxy does not send 302
to the caller (for ex. GW), but it waits for timeout of the Phone2.
Then the proxy sends 302 to the caller.
Can I do in kamailio, that it will ring on the Phone1 and also on the
number where it is redirected? I know kamailio is a proxy and cannot
initiate a call, but is there any solution? Thanks.
Mino
** **
*
*
** **
Hi,
** **
I have a question regarding the below ACK response Kamilaio receives from a
carrier. Can someone please help me understand why the “*Route:
sip:callmanager@192.168.160.43:5060> *sent from Kamailio is missing a “<”
as shown in the captures below? Our Internal proxy is treating that as a
malformed header and dropping the packet.
** **
** **
*ACK from CARRIER*
* *
U 2012/08/01 18:32:52.219852 4.55.18.227:5060 -> 192.168.160.47:5060****
*ACK sip:callmanager@192.168.160.43:5060 SIP/2.0.*
Via: SIP/2.0/UDP 4.55.18.227:5060;branch=z9hG4bK04B0eef33040e9b8e70.****
From: sip:+14088442721@4.55.18.227:5060;tag=gK043001e3.****
To: sip:+19728931740@192.168.160.47:5060;tag=b307370c678f3b44.****
Call-ID: 295226_50030734(a)4.55.18.227.****
CSeq: 18079 ACK.****
Max-Forwards: 70.****
Route: <sip:192.168.160.47:5060;lr=on>.****
*Route: <sip:2c6c6d1ab58c623912f6b8a6ee526982@192.168.160.44:5060>.*
Content-Length: 0.****
.****
*ACK FORWARDED TO SIP PROXY*
** **
U 2012/08/01 18:32:52.220612 192.168.160.47:5060 -> 192.168.160.44:5060****
*ACK sip:2c6c6d1ab58c623912f6b8a6ee526982@192.168.160.44:5060 SIP/2.0.*
Via: SIP/2.0/UDP 192.168.160.47;branch=z9hG4bKcydzigwkX.****
Via: SIP/2.0/UDP 4.55.18.227:5060;branch=z9hG4bK04B0eef33040e9b8e70.****
From: sip:+14088442721@4.55.18.227:5060;tag=gK043001e3.****
To: sip:+19728931740@192.168.160.47:5060;tag=b307370c678f3b44.****
Call-ID: 295226_50030734(a)4.55.18.227.****
CSeq: 18079 ACK.****
Max-Forwards: 69.****
Content-Length: 0.****
*Route: sip:callmanager@192.168.160.43:5060>.*
I have attached the config file for you reference. Kamailio version is 3.2.3
****
** **
Thank You****
Varsha****
Somwhere losted since openser 1.0
file kamailio.init
Please fix in function start:
start() {
echo -n $"Starting $prog: "
.....
[ $RETVAL = 0 ] && touch something
to
[ $RETVAL = 0 ] && touch someting && success
I mean call function
success
in the end of the expression.
Thanks in advance.
Hello,
Kamailio SIP Server v3.3.1 stable release is out.
This is a maintenance release of the latest stable branch, 3.3, that
includes fixes since release of v3.3.0. There is no change to database
schema or configuration language structure that you have to do on
installations of v3.3.0. Deployments running previous v3.x.x versions
are strongly recommended to be upgraded to v3.3.1.
For more details about version 3.3.1 (including links and hints to
download the tarball or from GIT repository), visit:
* http://www.kamailio.org/w/2012/08/kamailio-v3-3-1-released/
RPM, Debian/Ubuntu packages will be available soon as well.
Cheers,
Daniel
--
Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Seattle, USA, Sep 23-26, 2012 - http://asipto.com/u/katu
Kamailio Practical Workshop, Netherlands, Sep 10-12, 2012 - http://asipto.com/u/kpw
