and keeps respawning it all the time. Of course
new processes won't stay around for long because they'll try to bind
on some ports which are already taken. But if you're respawning often
enough, the total count could go into thousands.
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to wait until the database operation is completed.
That's not the case for the delayed mode.
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to
Asterisk, bypassing OpenSIPS.
Most phones take the address from the From header - you might want
to change that instead. To do this, look at the uac module.
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should use the uac module and
uac_replace_from(somebody, sip:number@host);
Otherwise, you're in a world of pain to get all the rewriting correct
in both ways (answers, subsequent requests, detecting direction, etc.)
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On 2 March 2011 13:47, Razvan Crainea razvancrai...@opensips.org wrote:
Note that there will be no functional changes, but only structural ones, all
current functions exported by nathelper module will still be available (some
name changes might be possible though), just that they will be
2011/2/15 Stanisław Pitucha virap...@gmail.com:
On 15/02/11 15:05, Bogdan-Andrei Iancu wrote:
When
-
The meetings will be monthly, in the second Wednesday of each month.
...
As a first meeting we want to start with 22th of February, 17:00 CET
I like the idea, just wanted to point out
On 20/12/10 13:51, Denis Putyato wrote:
Thank you Bogdan for your answer. Now I understood that apply changes is a
bad idea.
But during process a call I have to make some changes to INVITE message.
For example,
I need to add Remote-Party-ID (RPI) and/or P-Asserted-ID (PAI) and make
On 03/12/10 16:03, Vic Jolin wrote:
I have been having problems with my opensip installation, I cant even do
more than 100 cps,
You'll have to see what is it that is holding up the traffic really.
Use the benchmark module to get the general idea of the packet
processing time. Benchmark your
On 02/11/10 12:25, Deon Vermeulen wrote:
I'm trying to setup NAT to NOT use MediaProxy when it detects that 2
devices are behind the same NAT Device, but rather have coms go
directly between them.
Just a warning in case you didn't consider this: do you want to support
random users you have
On 29/10/10 06:06, Bradley Falzon wrote:
This issue had been discussed before on this list, I don't have the
exact conversation, however, the advise seemed to be make the proxy
stateful.
You can either look for the stale=true solution (I don't know how to
implement that one, I'd like to know
On 20/10/10 15:08, Victor Gamov wrote:
Ca somebody explain me which result expected when loose_route() called
by X.X.X.X for request like following:
-
ACK sip:5700...@x.x.x.x SIP/2.0.
Record-Route: sip:X.X.X.X;lr=on
To: sip:5700...@domain.com;tag=4ded008d6ca9692485d1918f60c7da12
-
On 30/09/10 10:48, Deon Vermeulen wrote:
Is there perhaps someone that could give me a 2 Context config
example for OpenSIPS?
It depends what you mean by a context. If you want to sort by the source
host, just use
if ($si == 1.2.3.4) {
route(something);
} else if ($si == 3.4.5.6) {
Hi,
You cannot just migrate the concept of contexts and try to apply it to
opensips. These projects simply have a very different architecture.
Don't try to force something because you already know it...
Contexts basically match on registered users or on the source of calls
really. You can do
On 30/09/10 11:59, Deon Vermeulen wrote:
Let me try and explain a scenario as brief as I possibly can.
I have Company A (Domain A) and Company B (Domain B).
Domain A and B should be completely Transparent to each other.
See like Domain A and B in their own respective Bubbles completely
On 30.09.2010 23:10, Deon Vermeulen wrote:
I'm able to setup a call between the 2 users, but the call drops after
+-30 seconds.
VOICE data is sent and received between the devices while in the +-30
seconds call duration.
30 seconds sounds like a typical ACK timeout. Try to dump your SIP
about Asterisks crashing (well... apart from calls dropping
of course) and all your registrations go to one place, instead of each
phone being configured differently.
In general - yeah - Opensips should work just fine in this case.
Regards,
Stanisław Pitucha
Hi all,
I've got 2 situations where opensips doesn't store the `acc` info
properly (imo), for some reason.
The first case is a bit complicated (best viewed with your favourite
monospace font):
---8---
A Proxy B
INV-
-100-INV
INV-
-100-INV
On 27/08/10 16:36, Joan wrote:
At the moment I still have some doubts on where to put the logging
part, to minimize the false positives (setting like in the example it
marks the first packet as wrong)
Not tested at all - but something like that should work.
if (is_method(REGISTER)) {
On 02/08/10 10:58, Jayesh Nambiar wrote:
This shows to assign a value to $du and then call t_relay. But is it
possible that I can assign a value to $dd. When i try to use it, opensips
fails to start with error Invalid Left Operand in Assignment.
Any suggestions on how to do this.
Only $du is
On 25.07.2010 07:53, Adrian Georgescu wrote:
These are all valid points. And DNS is not single thing that causes this
behavior, any operation can block like radius, mysql query, and the result
is the same.
There's a big difference here. Mysql, radius and other
typically-blocking services
Hi all,
I wanted to collect some ideas on how do you solve DNS connectivity
problems. I've run into those issues a couple of times already and don't
see a perfect solution so far. Maybe I can trigger some discussion:
Some background:
- opensips blocks the child process while resolving a domain /
On 11 May 2010 13:32, Paris Stamatopoulos mob...@realize.gr wrote:
You get the last digit of the caller's number (0-9):
$var(group) = $(fU{s.substr,-1,1});
So you're basically doing the same as a simple dispatcher, which has
the option builtin:
On 11.05.2010 15:21, Paris Stamatopoulos wrote:
No I was talking about attended transfers as well.
Then I don't think your solution actually works.
In a simple scenario you have A calling B and B calling C. If you only
consider from addresses, you have no guarantee that you get both calls
on
On 05.05.2010 11:50, Paris Stamatopoulos wrote:
Has anyone experience this before? Do you have any working solutions?
Yes and yes. Short version: It's impossible without hacking asterisk.
Long version:
First we tried routing all related calls to the same destination using
the dialog module. It
Hi,
You may have different environment at your site, but this is my experience:
- NDB is hard to setup / maintain - it might seem easy at the start
(trivial even), but when something fails and a node doesn't want to
reconnect to the cluster again, you're left on your own with the source.
Not many
On 29.03.2010 07:18, rajib deka wrote:
put, I am not able to uderstand the 'state' and timer parameters. Can
State is the dialog state - it goes more or less like this:
After INVITE - 1
After 18X - 2
After 200 - 3
After ACK - 4
After BYE - 5
Not sure what you mean by timer. If it's timestart /
Hi,
I'm trying to reduce the number of connections to the database, because
the current setup is completely killing table cache unfortunately. 20
children (6 listening ips) * 5 modules not nice. I was wondering if
there is any way to cut the number of connections via the virtual db
module. If
Ok - I already see that was a bad question, since it's one connection
per process, not per module, so virtual db won't save me here... Please
ignore :)
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On 20.03.2010 17:02, Kyle Romulas wrote:
Have you considered optimizing your DB? We are handling thousands of calls
per minute with a cluster of proxies balanced via SRV and using a centralized
DB server. We had to optimize when we discovered that under peak call
periods we were exceeding
On 17/03/10 17:15, Matthew S. Crocker wrote:
UA - PROXY 200 Ok (SDP/G711)
PROXY - PSTN 200 Ok (SDP/G711)
** RTP Established between UA PSTN (mediaproxy/rtpproxy ??) **
** Gateway detects fax tone and attempts to REINVITE to T.38 **
PSTN - PROXY INVITE (SDP/T38)
PROXY - PSTN 180 Trying
On 16.03.2010 12:23, rajib deka wrote:
I am not able to understand the use of dlg_flag in dialog module. what is
the diffrernce between following statements,
modparam(dialog, dlg_flag, 4)
modparam(dialog, dlg_flag, 13)
http://www.opensips.org/html/docs/modules/1.6.x/dialog.html#dlg-flag-id
On 08.03.2010 05:05, osiris123d wrote:
Have two servers using Linux Heartbeat and have OpenSIPS Load Balancer
running on it. The load balancer will balance between two OpenSIPS proxy
servers. This way I have 100% redundancy.
Complicated.
Why not simply 2 proxies running with Heartbeat / Carp
.
Did that point... let's say, occur on any 2.0 development schedule?
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On 04.03.2010 20:05, Brad Bendy wrote:
That makes since now, we had a float actually. At this time no way to
work with floats at all right?
There's always a workaround :)
$avp(i:55) = $avp(i:70) * .006;
$avp(i:55) = $avp(i:70) * 6;
Just treat i:55 as if it was multiplied by 1000. It seems
On 02.03.2010 18:18, Bogdan-Andrei Iancu wrote:
SF offers the following options
- SVN
- git
- mercurial
- bazaar
Should we keep SVN ? pros ? minuses ? something much better ?
I'd go with - please no SVN ;) (or at least keep an official mirror in
another repo type)
Why?
Sorry, my message seems to be cut by mailman (according to the
sourceforge archive) - resending full version:
I'd go with - please no SVN ;) (or at least keep an official mirror
in another repo type)
Why?
- We're keeping a number of patches that are company-specific, so will
never go into
)
They are branches - there's only one branch incoming, but proxy
forwards the message with XXX.0 and XXX.1 branches.
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On 25.02.2010 14:47, Bogdan-Andrei Iancu wrote:
and printing the $(branch(uri)[*]) just before the t_relay shows
anything (a branch extra to RURI) ?
I have this in my config now:
---8---
} else if(is_method(SUBSCRIBE)) {
xlog(L_ERR, subscribe branches1: $(branch(uri)[*])\n);
Hi,
I've noticed today some strange behaviour with branches in routing
SUBSCRIBE packets. I'm running a setup like this:
phone - proxy (opensips 1.5.3) - presence server (opensips 1.6.0)
The first packet goes through without any problems, presence server
returns 200/ok, proxy does
On 23.02.2010 09:31, Jan Rozhon wrote:
chieftec:~# opensipsctl fifo get_statistics 'shmem:'
shmem:total_size = 33554432
That's how much you've got allocated for shmem ~36M
shmem:used_size = 24018088
Used right now
shmem:real_used_size = 26151984
Used right now + the overhead (from allocator
On 22.02.2010 10:16, Jan Rozhon wrote:
So there is a problem with database as well. Does it mean, that even if
system utilization tools (SAR) shows, that only about 1200 MB is used,
opensips has run out of memory?
Opensips preallocates the memory, so system tools won't show you the
real
On 18.02.2010 13:57, Jan Rozhon wrote:
AMD Athlon processor. As a generator of SIP traffic I use SIPp v3.1
running on 4 virtual computers as UAC and two computers as UAS all
You're not running them all on the same physical box, are you?
If yes, then you might find sipp itself to be a CPU hog
On 02.02.2010 14:17, Julien Chavanton wrote:
I still have a problem with binding all IP some headers do not have the IP
set :
Record-Route: sip:0.0.0.0;lr;.
Via: SIP/2.0/UDP 0.0.0.0;branch=z9hG4bKfc58.1427b764.0.
Is there a way to tell Opensip to use the routing source IP, or other
might try to crash your host, since the start of
the packet is fine (which allows to start the parsing), but then some
random characters follow.
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On 20.12.2009 03:22, Josip Djuricic wrote:
I am wondering about shared memory and childs...when you dedicate for
example 128mb shared memory to opensips on startup does that mean this 128mb
is shared between all of the childs or does it mean that every child gets
128? Since it is not private I
On 19.12.2009 18:26, Josip Djuricic wrote:
I have a user 12...@domain.com registering to my opensips server. Now
from some equipment I get a sip invite with 12...@ip-address. Ofcourse
with lookup(location) usrloc I get Not found. Is there a possibility
alias can help me with this and what
On 18/12/09 14:01, Yoo Chan Jeon wrote:
But I still do not know if the Dispatcher module dose the load balancing on
the SUBSCRIBE, and REGISTER message too.
I know that it does on the INVITE message.
dispatcher will handle whatever request you're processing right now. It
doesn't matter what
?
Can you show the algorithm? If you can get repeat it in opensips, then
you can repeat the Digest Hash Algorithm too and check the password
that way.
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://en.wikipedia.org/wiki/Digest_access_authentication
I'm curious though - why would you want to use the password in case
it's not correct?
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packet and call t_newtran() on it:
- will modification of the message body take effect on t_relay()?
(done both before and after newtran)
- will modification of $ru and $du have any effect after newtran()
- what other scenarios should I think about that might be affected?
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on
the channel is that any change to the text of the message that is done
*after* newtran(), will not be present in the failure route, or new
branches.
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to be accessing and updating the globals for every call.
If you're doing something especially tricky, you might consider
writing a module to do that... What is it exactly that you want to
achieve?
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outside.
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to subscribe the client to all
other phones' events?
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everyone else, but
it doesn't mean that your client will display the status of a client
it didn't ask for. So you'll have to test this one...
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% between two servers.
Maybe that will be enough for you?
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like that from the
diagrams) handle shared transaction is Yxa. But that's a completely
different beast :)
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2009/9/14 Italo Dacosta idaco...@gatech.edu:
With the above configuration the proxy is able to reach around 20,000
calls per second (cps). However, I have noticed that the CPUs in the
I just thought about something different. What kind of link are you
using on each side and what throughput can
2009/9/14 Italo Dacosta idaco...@gatech.edu:
I am using a stateless proxy configuration with 4 children processes,
just required modules (i.e., not billing) and a very simple routing logic:
Ok - that's the first problem. Number of children == number of
processes. You cannot handle more than 4
2009/9/8 ghaith.alkay...@telecom-bretagne.eu:
The mysterious point is that all packets orginating from the interface
(192.168.20.20) towards UAC(192.168.20.1) have Bad checksum in the UDP
level.
If you're capturing at the machine that has the 192.168.20.20
interface, then everything is ok.
2009/8/24 Ratheendran R ratheendra...@gmail.com:
Too much shared memory demanded: 33554432
You just don't have enough memory available (that's ~32MB). You can
change pkg_mem when compiling and shm with a commandline option
(probably... that's a pretty old version, so things might have been
If you posted the capture of that call to some pastebin, it could give
us some nice information too.
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an INVITE, state 5 is deleted - more or
less).
Capture the traffic and see what's going on.
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T: 01225 800 831 | F: 01225 800 801 | E: s...@gradwell.net | www.gradwell.com
Gradwell – Internet for Business People
Phone Services | Business
2009/8/19 星宇 刘 besti...@yahoo.com.cn:
Aug 19 15:38:46 [1116] ERROR:core:set_mod_param_regex: parameter
max_expires_subscribe not found in module presence
Which version of opensips are you using? 1.5.X has separate options:
http://www.opensips.org/html/docs/modules/1.5.x/presence.html#id228247
2009/7/16 Alex Balashov abalas...@evaristesys.com:
What is the benefit of creating a new transaction on top of the retrans
checks? Why would I not just want to wait until I call t_relay(), which
will also create a transaction if it does not already exist. Why it would
be beneficial to have
2009/7/14 Alex Balashov abalas...@evaristesys.com:
http://www.opensips.org/html/docs/modules/1.5.x/tm.html#id272150
A bit related question. Since the docs mention:
If the processing of requests may take long time (e.g. DB lookups)
and the retransmission arrives before t_relay() is called, you
2009/7/14 Alex Balashov abalas...@evaristesys.com:
Are you saying that t_check_trans() will create a new transaction for a
non-ACK/CANCEL retransmission too? Or that it retransmits the last reply
sent statelessly somehow?
I was confused about it too at the beginning. Let me paste what Bogdan
I'd go with:
You can, but don't expect wonders. For example, check your maximal
possible throughput on udp - I've seen that some kernels can't cross
1mbps boundary (just by blocking the queue, so it's the program
slowing down on send()). They were ok with tcp at the same time.
Also delays go up a
2009/6/16 Iñaki Baz Castillo i...@aliax.net:
What do you mean with a tool that recognizes the configs?
There is noly one: OpenSIPS itself :)
BTW, when I edit my config files, I use Bash syntax. It works great
(except when using switch/case stament).
I sometimes just enable autoindent and
Hi,
I've got a performance-related question:
Is there any good way to get a statistic of how busy opensips
processes are on average? While the processes are blocked to process a
message, is there a way to check what the udp-sip processes are doing
- idle / working without going into stuff like
2009/4/15 Bogdan-Andrei Iancu bog...@voice-system.ro:
Right now the function is not allowed in failure route, but on a short
overview on the code, there is no reason not to. So, as a trick you can call
from failure route a generic route block and use the ds_selectxxX() from
there - this will
2009/4/8 Bogdan-Andrei Iancu bog...@voice-system.ro:
if I got it right, you want to have a kind of dispatching to guarantee that
all in or out calls for user A are going through the same PBX. Correct?
In short:
Yes. But internal calls should use only one PBX in the cluster, not two.
Long
Hi,
I'm trying to find a solution for using both a dispatcher and attended
transfers reliably.
Standard problems are:
- calls from user A must go to the same pbx as all ongoing calls To and From A
- dispatcher failover must work properly if there's a record of A's
dialog with PBX-1, but PBX-1 goes
2009/4/7 Adrian Georgescu a...@ag-projects.com:
You cannot do this reliable the way you propose. The only reliable way is to
sit behind a PBX/B2BUA that your control and behaves in a consistent and
reliable way. Otherwise you are at the mercy at the combinations of the SIP
User Agents that are
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