Hello,
just asking to see your experience deploying sip platforms on virtual
systems. So far I was running Kamailio in virtual machines and no
problems, but I insisted that media servers to be on physical machines.
Lately is more pressure from the market to go everything virtual.
So the question is more about having everything on virtual systems,
proxy and media server, where the media server can deal with
transcoding, conference rooms and IVRs.
Any strong comments pro or against?
What is your preferred virtualization system for such deployments?
Cheers,
Daniel
--
Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat
Hi all,
I announce the first stable release of OverSIP SIP Proxy:
http://www.oversip.net
OverSIP features:
-----------------------------------------------------------------------------
- SIP over UDP, TCP, TLS and WebSocket transports.
- Full support for IPv4, IPv6 and DNS resolution (NAPTR, SRV, A, AAAA).
- Fully programmable using Ruby programming language.
- The perfect Outbound (RFC 5626) Edge Proxy.
- Fully asynchronous event-based design.
OverSIP is not a replacement for well-known proxies as Kamailio
(sip-router). OverSIP does not implement basic proxy features such as
Digest authentication, registration or parallel forking.
OverSIP can behave as an Outbound Edge Proxy between clients and SIP
registrars, or can be placed anywhere for building custom SIP /
WebSocket applications.
Cheers.
--
Iñaki Baz Castillo
<ibc(a)aliax.net>
>> just asking to see your experience deploying sip platforms on virtual
>> systems. So far I was running Kamailio in virtual machines and no problems,
>> but I insisted that media servers to be on physical machines. Lately is more
>> pressure from the market to go everything virtual.
>>
>> So the question is more about having everything on virtual systems, proxy
>> and media server, where the media server can deal with transcoding,
>> conference rooms and IVRs.
>>
>> Any strong comments pro or against?
>>
>> What is your preferred virtualization system for such deployments?
>>
I host ~120 asterisk virtual serves on OpneVZ for more than 3 years,
runs great. These are business customers so they can be very
demanding, allot of custom apps and such. I host up to 150 phones in
virtual environment, if more is needed, I switch over to a physical
server. This is just a personal preference, not limited virtually,
but customers with that many or more phones usually want a dedicated
server and can certainly afford the up charge.
Ensure your resources for each VM are enough for the voice apps you
want to run, be cautious about disk I/O like high volume recording,
database dips per call and such. I just converted my physical
sip-routers to virtual about a week now on OpenVZ, so far so good.
Transcoding required higher proc and mem loads so partition resources
accordingly. I do run Asterisk 10.0 conf bridge server virtual, SIP
only, no need for timing interface like DAHDI. This is running well,
no problem that I'm aware of, just works.
I use VirtualBox and VMware for demo systems and lab work but OpenVZ
for production due to higher density you can achive per physical
server.
Whatever virtualization you want to use or are comfortable with,
ensure you know it backwards and forwards and what to do if it breaks
and how to increase resource when needed and by all means do your lab
work ahead of time. You will not be able to simulate every situation
in the lab, but you can certainly stress test your environment with
high call volume, disk I/O recordings, databases operations, live
migration, upgrades, downgrades, backups, resource monitoring, alarm
triggers, test, test, test and then test some more.
Virtualizing voice systems has been around a long time, most virt
platforms can be implemented to perform well for voice, just got to do
your homework.
Good luck!.
JR
--
JR Richardson
Engineering for the Masses
Hi all.
I'm working with kamailio for about 1 year and still don't understand
difference between integer and string IDs of AVPs.
Is it just a question of usability ?
Dear List
I am trying to run the below command from a REMOTE server where i use Cacti
to graph all my data. I am trying to graph on a per trunk basis.
kamctl fifo profile_get_size trunk | awk -F '=' '{print $4}'
This command yields an output on the local server.
Could someone perhaps point me in the right direction?
Thanks!
I haven't been able to find the answer in the documentation.
It seems DROUTING does not take regex in the dr_rules table to match
routable numbers, are there any kind of wilcards that could be used.
How does one specify different rules for 8 digit numbers and 10 digit
numbers.
TIA
--
O: 4000-1020
C: 6040-3624
Hello,
please do not send private messages from mailing lists discussions, keep
the mailing list cc-ed so other can participate or learn from archives.
On 8/27/12 5:22 PM, Simon Hintermann wrote:
> Hello,
>
> I will try with a smaller "Subscription Expiry (s):" in my SNOM.
>
> Just to be clear with the method of test:
>
> - firewall redirects SNOM to my first ISP
> - telephone works perfectly, registered in Kamailio...
> - firewall redirects SNOM to my second ISP
> - telephone is unregistered in Kamailio, impossible to "re-register"
> the identity1, says "network error"
> - only way to recover is to change the SNOM's IP address
what do you mean by 'firewall redirects Snom to Nth ISP'? Is the IP of
snom static or taken via dhcp from adsl router?
Cheers,
Daniel
>
> Thanks for your support, greetings
>
> Simon
>
> Le 27.08.2012 09:04, Daniel-Constantin Mierla a écrit :
>> Hello,
>>
>> what is the expire time of your registrations? 24 hours seems a lot
>> to re-register.
>>
>> From kamailio point of view, the contact is expired based on
>> un-registration or timeout. You can look with 'kamctl ul show' to see
>> if the phone is still registered or not.
>>
>> Anyhow, in such cases is good to set registration time quite small,
>> say 5 or 10 minutes, but if there is a problem in the client side, it
>> is not much to do in server side.
>>
>> What is the adsl routed, does it do SIP ALG? I have snom on ADSL
>> using tcp, does not seem to make big problems.
>>
>> Cheers,
>> Daniel
>>
>> On 8/16/12 2:10 PM, Simon Hintermann wrote:
>>>
>>> Hello,
>>>
>>> I am experiencing some disconnections with my VOIP phones.
>>>
>>> Let me explain:
>>>
>>> My server is an Asterisk + Kamailio in a datacenter with a fixed IP,
>>> and a bridge firewall in front of it, so no NAT here.
>>>
>>> My client (several offices) have usually no problem when they have a
>>> fixed IP, but encounter fatal disconnection when they have a dynamic
>>> IP. I had the case with Siemens, SNOM and Aastra phones, without any
>>> difference. They are of course natted behind an ADSL router. I also
>>> have clients doing load-balance between two ADSL lines, which is
>>> also problematic.
>>>
>>> The only way for me to recover is to change the private IP of the
>>> VOIP phones. Seems more like a TCP problem or a routing problem, but
>>> I am no network guru.
>>>
>>> I have two ADSL lines to test it, and when I switch from one line to
>>> another, every phone becomes "Not registered". Even falling back tot
>>> he first line does not help recovering. I did not try to wait for
>>> longer than 24 hours to recover.
>>>
>>> So my question is, more generally: is it possible to allow the
>>> clients (VOIP phones) to re-connect with a new IP address without
>>> having closed correctly the first connection (SIP session, I
>>> presume)? It seems not by default, but I played with the timeouts
>>> server-side and phone-side, without luck.
>>>
>>> I saw many posts with PBX having problems when behind a dynamic IP,
>>> but nothing about the clients having dynamic addresses or
>>> load-balancing. Normally, having a dynamic IP does not mean that
>>> your IP will change « on-the-fly », but each time you shut down and
>>> reboot later your ADSL router, but we came to the conclusion that
>>> every client NOT having a fixed IP was experiencing fatal
>>> disconnects. So I managed to test and validate this behavior with
>>> our two ADSL lines.
>>>
>>> Hope someone can help me.
>>>
>>> Greetings
>>>
>>> Simon
>>>
>>>
>>>
>>> _______________________________________________
>>> 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
>>
>> --
>> Daniel-Constantin Mierla -http://www.asipto.com
>> http://twitter.com/#!/miconda -http://www.linkedin.com/in/miconda
>> Kamailio Advanced Training, Berlin, Nov 5-8, 2012 -http://asipto.com/u/kat
>
--
Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat
Hi,
Can any one help me .. how to configure sms module?
what changes required in scscf.cfg or any other cfg also??
Thanks and regards,
Raman Kumar
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Cheers,
Daniel
On 8/27/12 9:43 AM, andre wrote:
> 27.08.2012 10:09, Daniel-Constantin Mierla пишет:
>> Hello,
>>
>> On 8/14/12 11:31 AM, andre wrote:
>>> 14.08.2012 12:21, Daniel-Constantin Mierla пишет:
>>>> Hello,
>>>>
>>>> thanks, it was committed in git master branch, backports to follow
>>>> before minor releases.
>>>>
>>>> Do you build your own rpms or you install from the opensuse build
>>>> system?
>>> From the opensuse build system.
>>
>> ok, I will update that system as well. is 'success' available in all
>> distros using rpms (like opensuse)?
> Not shure about Suse. I even have no someone to ask
>
> Better add to RH distros only.
>>
>> Cheers,
>> Daniel
>>
>>>>
>>>> Cheers,
>>>> Daniel
>>>>
>>>> On 8/2/12 6:24 PM, andre wrote:
>>>>> Somwhere losted since openser 1.0 :)
>>>>>
>>>>>
>>>>> Please fix in:
>>>>>
>>>>> 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
>>>>>
>>>>> _______________________________________________
>>>>> 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
>>>>
>>>
>>
>
--
Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat
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