Hi,
Kamailio is definitely the exact tool for this purpose, I have exactly the
same setup running as yours and for scalability we started using Kamailio
in front of our asterisk servers. Long story short, read these articles.
http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-as
Hello,
We have been using Asterisk for sometime and over the last year have
started hosting instances for our clients on a vmware platform. These
virtual pbx are located on public ip addresses and each customer has
their own SIP trunk arrangements with various providers. We have
decide
For the sake of argument, let's say it was a smart-ass answer-- if your post
was to "be pointed in the right direction where I could get (further) help"
then that next help should either be:
a) the wiki
b) a training class
c) a consultant
On forums I notice that those who get a better response
All right, fair enough. I realise that this list has left you with
little useful takeaways thus far, and, upon further reflection, that
does seem a little unfair, given the amount of effort and thought that
went into your post.
In due recognition of that, let me _try_ to tackle some of the mo
You will need to break down your e-mail into several simpler questions
and you will get some useful replies.
Most of the things that you want to do are possible. To craft a proper
e-mail to all your questions in your original e-mail would take quite
some time ... and time is expensive for all. I t
It's not a smart-arse reply; it's sincere, earnest advice.
Really?! Perhaps you could explain to me how exactly is the "you should
get a consultant" comment on a routine set of questions I posted on a
mailing list created for that very purpose - for Kamailio users like
myself - anything other
Come on, let's be reasonable here. People are very helpful here and
do volunteer a great deal of time to help others solve their problems,
but ultimately, people have jobs and there are only so many hours in
the day.
I agree that the original post is well-thought out, extensive and
structure
Jeff,
On 01/30/2012 06:59 PM, Jeff Brower wrote:
start experimenting with basic call routing. Just take it one step
at a time, and as you encounter specific questions, post here
and I'm sure you'll get great help.
I think that's all anyone was really saying:
Either learn the technology yours
Mojo-
I'm surprised at the replies you received. Normally the people on this group
are extremely helpful.
Clearly you've spent significant time thinking about this, and your current
system and problem description below is
detailed and well-presented, with a clear rationale for using Kamailio.
It's not a smart-arse reply; it's sincere, earnest advice. There's only so
much of a massive conceptual nexus that people can reasonably traverse on a
mailing list. For the most part, mailing lists exist to answer specific
questions, not provide broad, fundamental guidance or extensive pedagog
Sounds like you would really benefit from hiring a consultant.
Smart-arse replies are not appreciated - if you can't contribute just
move along - nothing to see here!
___
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sr-us
Sounds like you would really benefit from hiring a consultant.
On 1/30/12 12:00 PM, Me wrote:
Apologies if any of the questions below are a bit dumb - I don't
pretend to be an expert in SIP/VOIP - I am just an ordinary user
looking for answers.
Our current setup involves processing a small n
Hello,
full xpath syntax should be exported, iirc, there is an xpath function
named position() that can be used to build expression to select the
desired note. perhaps googling this xpath function will reveal some
examples/hints.
Cheers,
Daniel
On 1/30/12 3:28 PM, Alex Balashov wrote:
Hi,
On Mon, Jan 30, 2012 at 7:39 PM, Uri Shacked wrote:
> Hi,
> thanks for the options. i will test it soon.
> looks like a perfect solution.
>
> BR,
> Uri
>
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sr-users@lists.sip-rou
Since this thread will probably end up in Google I''ll share my
experience. I ended up with this"
if(t_check_status("301|302"))
{
#NOTE: must assign to $du to keep R-URI intact
$var(contact) = $T_rpl($ct);
$var(contact) = $(var(contac
Apologies if any of the questions below are a bit dumb - I don't pretend
to be an expert in SIP/VOIP - I am just an ordinary user looking for
answers.
Our current setup involves processing a small number of internal sip
accounts (up to 10, no more than that) and one "public" one (with a
separ
Hello,
also have in mind that web auth is enabled, all requests from local
users (like those generated by pua_usrloc) are challenged for
authentication. Either add 127.0.0.1 and server ip in address table and
enable WITH_IPAUTH or add at the top of route[AUTH] a condition like:
if(src_ip==my
Hi,
If by auth you mean the WITH_AUTH #!define in the configuration file
then it sounds like a configuration error. You should check that the
pua_usrloc stuff you've added isn't excluded when WITH_AUTH is not set.
It is also worth remembering that, if you are using a client that does
not support
Hello,
great. I got presence... but only when I do not enable auth.
Is there something I have to look out for when enabling auth
I've take the stock config for 3.2.0 as packed in the ubuntu package from the
website, implementet your changes and reinit the database.
Apart from your changes
Hi
I have been testing with versions throughout the last week and I can
confirm that 3.2.0 does not have this issue.
Regards
On Wed, Jan 25, 2012 at 2:42 PM, Daniel-Constantin Mierla wrote:
> Hello,
>
> there were many fixes in branch 3.1 since 3.1.0 was released, the latest
> patch release in
Thanks for your response.
What I found is:
1. If call is from phone registered to IP (external or internal) - then I
do not need any of my modifications - ACK goes through loose_route,
or t_check_trans() is OK and ACK is also OK.
2. If call is from phone registered to name (sip.mycompany.com) - th
Hi,
I guess this is more of an xpath question than a Kamailio one, so
forgive me. All the same, how does one deal with a document like this
using $xml(...):
...
...
Specifically, how do I access the second, third, Nth element with the
sam
does siremis provide client interface apart from the admin interface?
On Mon, Jan 30, 2012 at 5:58 PM, Daniel-Constantin Mierla wrote:
> Hello,
>
> for completeness, also Siremis Web Management Interface for Kamailio has a
> basic billing engine based on SQL stored procedures (creating CDRs and
thanks a lot!
really appreciate your help.
-abid
On Mon, Jan 30, 2012 at 5:58 PM, Daniel-Constantin Mierla wrote:
> Hello,
>
> for completeness, also Siremis Web Management Interface for Kamailio has a
> basic billing engine based on SQL stored procedures (creating CDRs and
> rating them) -- i
Hello,
On 1/27/12 9:54 AM, Gustavo Garcia Bernardo wrote:
Hi folks,
I'm having problems with exec_avp because is truncating the output of the
script executed. Looking at the code I see exec_avp is imposing a limit of
MAX_URI_SIZE (1024) but I don't understand why. The limit should be the
Hello,
On 1/23/12 10:43 AM, tro...@st.fri.uniza.sk wrote:
Hi,
I am trying to make kamailio PCSCF working with rtpproxy (for NAT
traversal).
I would like to ask, if there is any tutorial for this, or if you can
give me any advice.
Maybe IMS guys can give other hints, what is on public so far
Hi Laura,
thanks for the patch. I wonder how an xcap related function landed in
utils module, perhaps Juha felt more comfortable adding it there.
At first sight, patch looks ok. I cc-ed Juha in case he wants to have a
look and integrate it. If not, I will commit before 3.2.2.
Cheers,
Daniel
Hello,
for completeness, also Siremis Web Management Interface for Kamailio has
a basic billing engine based on SQL stored procedures (creating CDRs and
rating them) -- it could be a good start for implementing a billing
engine with custom needs. Read more at:
http://siremis.asipto.com/
http
Hello,
On 1/29/12 2:53 PM, Daniel Pocock wrote:
Construct the PEM file in this exact order:
cat server.example.com.pem> chain-server.example.com.pem
cat inter2.pem>> chain-server.example.com.pem
cat inter1.pem>> chain-server.example.com.pem
and then, in tls.cfg:
certificate=chain-server.ex
On 30/01/12 09:08, Klaus Darilion wrote:
>
>
> On 29.01.2012 12:21, Daniel Pocock wrote:
>>
>>
>>
>> There are now a number of TURN implementations available:
>>
>> http://www.resiprocate.org/ReTurn_Overview
>> http://turnserver.sourceforge.net/
>
> http://www.creytiv.com/restund.html
Thanks
Hi Daniel,
I'm sending you some memory leak fix we have found and fixed on
modules/utils/xcap_ auth.c.
The fixed memory leak is in pres_watcher_allowed (), and it is about
the "xmlFreeDoc" that it's never called for the "xcap_tree" variable.
We downloaded the source file from master branch and th
Is there any way how to print it this way ?
sercmd -s unixs:/tmp/kamailio_ctl mi_fifo cr_dump_routes
error: 500 - Internal server error processing 's': buffer too small
(overflow) (-2)
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailin
>> I notice that Asterisk needs to be patched to do it the way Kamailio does:
>>
>> https://issues.asterisk.org/jira/browse/ASTERISK-17727
>
> The Asterisk TCP/TLS implementation is marked experimental for a reason. And
> it's been that way for many years.
All the more reason for people to use
Hello,
the repositories with build RPMs done via OpenSUSE Build Services (OBS)
were updated. Fedora 14 is no longer offered by OBS, instead I added
support for Fedora 15 and 16. Also, there are now dedicated builds for
RedHat 6 and OpenSUSE 12.1.
Links to repos are provided at:
* http://w
Hi Uri,
if(is_present_hf("Reason")) {
$var(cause)=$(hdr(Reason){param.value,cause}{s.int});
xlog("L_INFO", "Our cause code: $var(cause)");
}
Also you can add this value to your CDR
modparam("acc", "log_extra",
"src_user=$fU;src_domain=$fd;dst_user=$rU;dst_domain=$
Hi Mihaylov,
If your Asterisk servers add a Record-Route header to the initial
Invite, for in-dialog requests ( ACK, BYE) you should use *loose_route()
*function to do the routing. This will make sure the requests go the
same path as the initial Invite. It is not a good practice to manually
r
Hi Uri,
You can use the param transformation:
$(hdr(Reason){param.value,cause})
http://www.kamailio.org/dokuwiki/doku.php/transformations:3.1.x#parameters_list_transformations
Regards,
Anca
On 01/30/2012 08:20 AM, Uri Shacked wrote:
Hi,
the gateway i use for sending call to pstn has an optio
On 29.01.2012 12:21, Daniel Pocock wrote:
There are now a number of TURN implementations available:
http://www.resiprocate.org/ReTurn_Overview
http://turnserver.sourceforge.net/
http://www.creytiv.com/restund.html
and for many purposes TURN relays and ICE are much better than relying
on
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