kamailio dispatcher module in my environment uses only UDP
I have not been able to find how I can configure kamailio dispatcher module to use TCP
Would appreciate any help
Bijan
Hi all,
I have an issue with a Kamailio and rtpProxy, when Asymmetric RTP is
used.
I have the system running, lines registered in Kamailio are able to call
to several destinations, but I have a problem with one provider that use
Asymmetric RTP (others providers use Symmetric RTP and I dont have any
issues).
A -----> Kamailio & RTPProxy ----------> Trunk -------> Provider
(Asymmetric RTP) -------> B
A is calling B.
The problem is that line in kamailio A can listen to B, but B can not
listen what A says.
I have captured the sip dialog and rtp traffic and I can see that:
Kamailio tells to provider that is using the port 52388 (media port in
SDP).
Providers tells Kamailio that is using the port 5394 (media port in
SDP).
RTP traffic from Providers to kamailio goes from port 5392 to 52388.
RTP traffic from Kamailio to Provider goes from port 52388 to 5392.
I have not found an RFC regarding Asymmetric RTP. The question are,
why RTP is sent from Kamailio (rtpproxy) to Provider to destination
port 5392?
It should use 5394 as SDP indicate?
or, port 5392 is learned when RTP traffic was received from 5392 to
52388?
# kamailio -V
version: kamailio 3.3.0 (i386/linux)
flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS,
DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC,
F_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE,
USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 4MB
poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
id: unknown
compiled on 22:17:07 Jun 18 2012 with gcc 4.4.5
# rtpproxy -v
Basic version: 20040107
Extension 20050322: Support for multiple RTP streams and MOH
Extension 20060704: Support for extra parameter in the V command
Extension 20071116: Support for RTP re-packetization
Extension 20071218: Support for forking (copying) RTP stream
Extension 20080403: Support for RTP statistics querying
Extension 20081102: Support for setting codecs in the update/lookup
command
Extension 20081224: Support for session timeout notifications
I hope to be clear with this explanation.
Regards,
Lucas Girard
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
receiving this error message on asterisk...
[2013-10-16 05:20:04] ERROR[3216][C-0000002c]: netsock2.c:269
ast_sockaddr_resolve: getaddrinfo("192.168.1.80192.168.1.80", "(null)",
...): Name or service not known
[2013-10-16 05:20:04] WARNING[3216][C-0000002c]: chan_sip.c:10873
process_sdp_c: Unable to lookup RTP Audio host in c= line, 'IN IP4
192.168.1.80192.168.1.80'
[2013-10-16 05:20:04] WARNING[3216][C-0000002c]: chan_sip.c:10464
process_sdp: Insufficient information in SDP (c=)...
The problem is NOT at the asterisk side but at the proxy side...when
kamailio rewrites the SDP message it adds the public IP twice in the "C"
filed of the SDP message instead of only one time like this 'IN IP4
192.168.1.80192.168.1.80' this is because the scripte is calling
rtpproxy_manage() in the routes TOASTERISK /FROMASTERISK and in
MANAGE_BRANCH / NATMANAGE. It should be either or,not both....but i don't
know how to do this since i don't have any scripting skills so if anyone
can modify the config file in the attachment i will be thankful...
Best Regards'
Mahmoud Ramadan Ali
Hello,
I'm looking for the best practises for setting Kamailio and the associated
hardware requirements.
Kamailio must deliver registrar and routing functions for 500k aliases, 5k
registered accounts and ~10k routes set in carrierroute module or similar.
My worries mainly concern DB integration. Is it better to setup Kamailio DB
on the same server (no network issue, etc.) or on another server?
Your opinion/feedback experience would be appreciated. Thanks!
Regards,
Igor.
---
Ce courrier électronique ne contient aucun virus ou logiciel malveillant parce que la protection avast! Antivirus est active.
http://www.avast.com
Hello,
this is a call for developers and community users to update the page
collecting upgrade guidelines from current stable (4.0.x) to upcoming
major release (4.1.0):
- https://www.kamailio.org/wiki/install/upgrade/4.0.x-to-4.1.0
If you are aware of a change in the behaviour for some parameters,
functions or statements, list them there.
I added the sql statements for MySQL that will allow to get to the new
database structure. Would be good to have the same for other popular sql
servers (maybe postgres, sqlite ...).
Cheers,
Daniel
--
Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Trainings - Berlin, Nov 25-28
- more details about Kamailio trainings at http://www.asipto.com -
Hi,
I can see that we are missing a lot of Sip messages in or db, so, checking
Homer FAQ there is a way to compile MySql with Insert Delayed for
partitioning table.
The problem is that I don't understand how to recompile MySql and make the
necessary changes.
The URL is
https://code.google.com/p/homer/wiki/FAQ#Q:_I_have_a_lot_of_SIP_traffic_an
d_some_SIP_messages_are_missing
The question is, can you send me a step by step how to do this? I really
appreciate your help.
I have CentOS 6.4:
[root@Homerdb02 ~]# kamailio -v
version: kamailio 4.1.0-dev9 (x86_64/linux) f8f3d3
[root@Homerdb02 ~]# cat /etc/redhat-release
Red Hat Enterprise Linux Server release 6.4 (Santiago)
[root@Homerdb02 ~]# uname -a
Linux Homerdb02 2.6.32-358.18.1.el6.x86_64 #1 SMP Fri Aug 2 17:04:38 EDT
2013 x86_64 x86_64 x86_64 GNU/Linux
[root@Homerdb02 ~]# mysql -V
mysql Ver 14.14 Distrib 5.5.34, for Linux (x86_64) using readline 5.1
Thanks,
Gus
Hello,
being discussed during last Devel IRC Meeting, we are planing to build a
Kamailio Project Technical Administration Group:
https://www.kamailio.org/wiki/devel/irc-meetings/2013blog#technical_adminis…
Its goal is to get a bunch of people that volunteer to do administration
tasks for the project, such as:
- helping with releases (e.g., patch backports, packaging, uploading
files for download, etc)
- doing sysadmin tasks for our servers (e.g., performing upgrades to
wiki, web site, etc)
- preparing technical decisions and doing them (e.g., what applications
to use to make operations easier, cloning git repository to github, ...)
From the devel meeting, so far we have Victor Seva, Fred Posner, Peter
Dunkley and Olle Johansson. Existing people doing admin tasks will
probably stay in (if they don't opt out): me, Elena-Ramona Modroiu,
Henning Westerholt (owner of devel.kamailio.org hardware), Jan Janak
(owner of sip-router.org hardware), Jesus Rodriguez and Oriol Capsada
(owners of kamailio.org hardware).
Requirements for candidates and other details:
- volunteer to do the work, it is not a paid job
- an existing record of activity within the project is a plus (e.g.,
developer, active mailing list member)
- reply to the lists detailing where and how you can help
- possibility to spend 1-4 hours a week for project administration (more
is welcome, sometime is not necessary at all)
Rewards:
- you will be listed as part of project administration on the website
- get to interact more with the project and the nice guys around it ;-)
- more spam - admin list address will be public and the list open so
everyone can send in case of critical situations (content/archive will
be kept private)
Note that we will try to build a group of an adequate size, thus not
everyone willing to participate may get in (at least on the first
phase). One criteria is to have skills that complement existing team
knowledge.
Cheers,
Daniel
--
Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Trainings - Berlin, Nov 25-28
- more details about Kamailio trainings at http://www.asipto.com -
Hi,
After restarting kamailio, pua_dialoginfo doesn't get updated when active dialogs change status.
I see the dialogs are read from the database, but when they receive eg. BYE the callback are not called.
I use db_mode 0 in pua, but tried mode 2 with no luck.
Should pua_dialoginfo register DLGCB_LOADED?
Regards,
Kristian Høgh.
Date: Fri, 25 Oct 2013 17:48:13 -0400
From: Fred Posner <fred(a)palner.com>
To: "Kamailio (SER) - Users Mailing List"
<sr-users(a)lists.sip-router.org>
Cc: SR-Users <sr-users(a)lists.sip-router.org>
Subject: Re: [SR-Users] Kamailio network edge for registration and rtp
pass-through
Message-ID: <6D231DE3-5BED-4601-A097-5A79CEF452F0(a)palner.com>
Content-Type: text/plain; charset=us-ascii
Do you need the registration be local to the asterisk?
I would have all the asterisks send calls to the Kamailio.
You can have a lookup on endpoint outbound to decide which asterisk should
handle the outbound call for that did.
Also a lookup for incoming DIDs, etc.
---Fred
> On Oct 25, 2013, at 5:43 PM, Jr Richardson <jmr.richardson(a)gmail.com>
wrote:
>
> Hi All,
>
> Starting a new project, roll your own SBC, not a full SBC, just need some
minor functionality. I'm interested in deploying Kamailio as a edge device
on a VSP for single entry point for hosted PBX's, Asterisk based. I had
some wonderful and informative conversations at Astricon 2013, several folks
assuring me Kamailio w/rtpproxy was the tool for the job, so this is a
follow up to delve more into details.
>
> I've been researching configs, topology, modules needed, ect... Most of
the examples I'm reading about for this scenario are spreading the
registrations across many PBX's without distinction. One concept I'm
struggling with is having a specific phone register to a specific PBX.
>
> phone-customer-A-x101><internet><kamailio><PBX-customer-A-x101
> phone-customer-B-x101><internet><kamailio><PBX-customer-B-x101
> phone-customer-C-x101><internet><kamailio><PBX-customer-B-x101
>
> I could add a unique identifier to some part of the registration of each
phone like 'custA-101@kamailio_server', custB-101@kamailio_server, ect.
What I'm not clear on is when the request comes to kamailio, where would I
identify what PBX the phone should register to and how to re-write the
'custA-101@kamailio_server' to '101@custA-pbx' and forward to the correct
PBX and ensure rtp flows through kamailio.
>
> Could this function be derived using dbaliases or possibly using
dispatcher with group number for each customer PBX?
>
> So assuming I can get the registrations to work properly, would standard
invite for calling just work or would I also have to have specific config in
place to ensure an invite from customer A phone also reaches the correct
customer A PBX?
>
> A point in the right direction?
>
> Thanks.
>
> JR
I would like kamailio to be transparent in the process, i.e. phones register
locally to their respective PBX.
Thanks.
JR
