Inc.
>
> *From:* users-boun...@lists.opensips.org [mailto:
> users-boun...@lists.opensips.org] *On Behalf Of *Brett Nemeroff
> *Sent:* Sunday, October 04, 2009 6:27 PM
>
> *To:* users@lists.opensips.org
> *Subject:* [OpenSIPS-Users] Global Variables
>
>
>
> Hey all,
>
Hey all,I'm wondering if there is a way to use global variables? I see that
$var variables are persistent across the process, but I need something that
will persist across all processes. Right now, I'm using memcache, but I'm
not sure if I'll have race conditions relying on that as I expect many
pr
What you describe is exactly what you can do with OpenSIPs. The term "SBC"
is overloaded and for certain vendors implies certain features.
I wouldn't say that you are "re-inventing the wheel" as someone else
mentioned, but more like making the wheel more useful to you. :)
On Thu, Oct 1, 2009 at 5
What's the problem? Maybe I didn't read closely enough, but I see your
configuration stated, but no specific mention of what isn't working
properly.
On Wed, Sep 30, 2009 at 3:57 AM, Jan D. wrote:
>
> I'm having problems with force_rtp_proxy(). My final goal is to use
> rtp_proxy
> for user to u
All,I'm doing a rl_check("$avp(s:accountid)")
where the avp isn't a valid pipe. Looks like this is throwing a segfault.
Here's the BT:
Core was generated by `/usr/local/sbin/opensips -P /var/run/opensips.pid -m
3072'.
Program terminated with signal 11, Segmentation fault.
[New process 5388]
#0 0x
pp that (in the same time with your tests) do
> run some caching commands??
>
> BTW, can you simply run via fifo the the cache_store command via
> memcached backend? Just to see if that is blocking or not.
>
> Regards,
> Bogdan
>
> Brett Nemeroff wrote:
> > Bogda
Noel,I get the same thing. I think this is normal behavior. The issue is the
i/o bottleneck writing the data to the hard drive. You can watch this
activity with vmstat 1
To really get the most performance possible, I normally turn my non critical
logging off entirely.
5cps however is pretty low
Andrew,Opensips alone isn't much of an SBC. It's possible that opensips +
the b2bua module (top hiding) + media proxy you can get what you are looking
for with acceptable performance.
I haven't heard anything *good* about opensbc. Most of what I've heard is
that you won't get much performance out
You may want to consider something like UCARP with a cold standby. Of course
you can use the REAL IP of the backup for a warm standby.
if both instances of opensips are using the SAME IP, it'll cause problems..
There are better ways.
-Brett
On Mon, Sep 21, 2009 at 3:00 PM, Noel R. Morais wrote:
e load balance is made in a
> network level). I'm having problems when a INVITE goes to one of the
> opensips and, for instance, the ACK goes to the other opensips. In
> this case, the TM module complains that couldn't find the transaction.
>
> got it?
>
> Regards,
>
What do you mean by share transactions? This doesn't sound like something
you'd normally want to do. Can you give us an example?
-Brett
On Fri, Sep 18, 2009 at 5:30 PM, Noel R. Morais wrote:
> Hi Guys,
>
>
> I would like to know if there is a way to share transactions (TM
> module) between ope
009 at 11:10 AM, Bogdan-Andrei Iancu <
bog...@voice-system.ro> wrote:
> Hi Brett,
>
> if you get a hang, do try to attach with gdb to the fifo proc (you can
> do a "fifo ps" in the beginning to see the PID of the fifo process).
>
> A backtrace of the fifo proc wil
JM,I'm not sure if I'm reading too little into what you are saying, but just
adding entries into the LCR table isn't going to do anything. OpenSIPs is
largely a toolkit. Like asterisk, it doesn't do anything until you set the
configuration to perform a certain way. So out of the box, the example
co
ou can
> do a "fifo ps" in the beginning to see the PID of the fifo process).
>
> A backtrace of the fifo proc will help in understanding the issue.
>
> BTW - what about cpu load ?
>
> Regards,
> Bogdan
>
> Brett Nemeroff wrote:
> > Hello list,
> &
Hello list,I periodically call opensipsctl fifo profile_get_values . Manually, it always works great.. but called from cron every minute,
it frequently hangs indefinitely until I kill it. Often I run ps ax and see
about 5 of them in there.
Running 1.5.3. Any ideas?
-Brett
_
Look in config.h for:#define PKG_MEM_POOL_SIZE 1024*1024
and try changing that to:
#define PKG_MEM_POOL_SIZE 1024*1024*2
On Wed, Sep 16, 2009 at 5:34 PM, Ron McCarthy wrote:
> Hi List,
>
> I just upgraded to the latest SVN relase and after a few thousand calls I
> start getting:
>
> Sep 16 15:
Check the dialog modules "bye on timeout":
http://www.opensips.org/html/docs/modules/1.5.x/dialog.html#bye-on-timeout-flag-id
On Thu, Sep 10, 2009 at 5:48 PM, Thiago Rondon wrote:
>
>
> I want to control the maximum duration of the entire call.
>
> Thanks!
> -Thiago Rondon
>
> Bogdan-Andrei Ian
Usually you set the group based on something else. I use memcache for this..
so I toss the IP against memcache, which returns a group id, then I use that
in the do_routing() cmd.
Of course, this requires pre-populating the cache src_ip => group_id
The newly released startup_route can do that for y
It can do some things like a SBC, but it isn't an SBC at heart.
What functionality are you looking for specifically?
-Brett
On Tue, Sep 8, 2009 at 1:05 PM, Kemp, Larry wrote:
> Can OpenSIPS be used as a Session Border controller sitting at my edge
> passing and receiving SIP traffic to others I
I hate to sound like a pain, but the information really isn't splintered at
all. The docs are all at www.opensips.org. Much of the older openser docs
are still very relevant, but you shouldn't even need those anymore. Check
the opensips web site. Go over the tutorials. Check the example
configurati
I *think* you mean the LCR module.
You want to use the drouting module. It's effectively the replacement for
LCR. It's faster and supports more routes. My dr_rules table has over 3
million rules in it.
-Brett
On Fri, Sep 4, 2009 at 9:34 AM, Szasz Szabolcs wrote:
> Hi,
>
> What are main benefits
Why not just use whatever value you are using as a parameter for
uac_replace_from ?
On Fri, Sep 4, 2009 at 8:07 AM, ASHWINI NAIDU wrote:
> Hi Bogdan,
>
> The whole scenario is i want the new from(after uac_replace_from)
> header value to be used by the call controller. is there any way i can
> forced ratelimiting for every new INVITE (which will trigger a new
> call).
> It is very powerful and flexible.
>
> Regards,
> Ovidiu Sas
>
> On Thu, Sep 3, 2009 at 4:01 PM, Brett Nemeroff wrote:
> > Hello list. I need to roughly limit cps. I was thinking that
Hello list. I need to roughly limit cps. I was thinking that the ratelimit
module would be good for this. I see it limits queues on packets per second.
How does this correspond to calls per second? Or does it? Does anyone have
any recommendations on how to effectively limit CPS? So basically I just
try opensips -creally, it's all you need. That and the debug log
On Thu, Sep 3, 2009 at 2:49 AM, Ghaith ALKAYYEM
wrote:
> What debugger for development do you recommend as well?
>
> Regards.
>
> On Thu, 2009-09-03 at 09:21 +0200, Saúl Ibarra wrote:
> > What about this? http://www.vim.org/scrip
0 PM, Olle E. Johansson wrote:
>
> 2 sep 2009 kl. 20.46 skrev Brett Nemeroff:
>
> > UCARP is pretty simple as well:
> > http://www.ucarp.org/project/ucarp
> >
> > Similar to the heartbeat (linuxHA) stuff, but a lot more lightweight
> > from my experience.
>
>
UCARP is pretty simple as well:http://www.ucarp.org/project/ucarp
Similar to the heartbeat (linuxHA) stuff, but a lot more lightweight from my
experience.
On Wed, Sep 2, 2009 at 1:13 PM, Noel R. Morais wrote:
> Hi Sergio,
>
> Thats is a good solution. but what happen if my loadbalancer goes do
I was thinking the same thing.. :)
On Wed, Sep 2, 2009 at 9:15 AM, Jeff Pyle wrote:
> vim?
>
>
> On 9/2/09 10:17 AM, "Ghaith ALKAYYEM" >
> wrote:
>
> > Hello,
> >
> > Could anybody suggest an IDE that will ease the development & debugging
> > of OpenSIPS?
> >
> > Thank you
> >
> >
> > _
, you are right, here the correct way to do it is via an intermediary
> INT AVP.
>
> Regards,
> Bogdan
>
> Brett Nemeroff wrote:
> > Bogdan,
> > I'm a little confused as to which modules can take avp transformations
> > INSIDE of the parameters and which c
Bogdan,I'm a little confused as to which modules can take avp
transformations INSIDE of the parameters and which can't..
For example, I didn't think this worked (as you suggested):
do_routing("$(avp(dr_id){s.int})
I thought you had to do:
$avp(n_dr_id) = $(avp(dr_id){s.int});
do_ro
All,I'm running OpenSIPs 1.5.1. I use dialog profiling to "count" calls up.
I notice that comparing my numbers to my providers using SBCs that my
numbers are always MUCH higher than my provider for "simultaneous calls
connected". For example, I may show 300 calls up, but they only show 75. The
numb
Did you make it executable? You may also need to adjust the script to point
to the proper binary location. Open up the script, it's not too tricky.
I'm pretty sure that this line:
test -f $DAEMON || exit 0
Says, "If the binary isn't there, just quietly die, without giving the user
a useful error m
RI
outside with translated TOURI ->b2bua -> inside see's restored TO URI
(original TO URI)
Forgive me for not entirely understanding the B2BUA scenarios and rules
quite yet. ;)
-Brett
On Tue, Aug 25, 2009 at 10:28 AM, Anca Vamanu wrote:
> Hi Brett,
>
>
> Brett Nemeroff
All,Question about the direction of the B2BUA module. I know one of the key
feature is topology hiding. Does this also occur in the SDP? I would expect
that it would need to still be paired with something like mediaproxy or
rtpproxy to achieve topology hiding with SDP as well, is this correct? Do
y
HAHAHA!
Well said!
On Thu, Aug 20, 2009 at 12:18 PM, Iñaki Baz Castillo wrote:
> El Jueves, 20 de Agosto de 2009, happyalways escribió:
> > Hiii..I installed mysql5.o...and Kamailio 1.5
> succesfully...Authentication
> > is working properly. Next i'm going through blind call forwarding. I need
Urmi,You log shows the call having failed. I'm not sure why you think it
runs for the proper duration. But as far as OpenSIPs is concerned, the call
failed. It's likely a problem in your sipp scenario. It's very possible that
sipp thinks the call is up, but the proxy does not.
In any case, OpenSIP
Urmi,I think Bogdan was referring to the trunk code. Dialog is stable in the
other releases. I use it pretty heavily on many servers.
With your issue, the dialog is created and since the relay fails, the dialog
is destroyed. This happens *so fast* that you simply won't catch it unless
you are wat
t; In log it shows the dialog is created successfully, and then it is
> destroyed as well.
> Dont know y its behaving like this !!
>
> Once again Thank you for your support.
>
> -Urmi
>
>
>
> On Wed, Aug 19, 2009 at 7:38 PM, Brett Nemeroff wrote:
>
>> Is it po
Roger,This is a common complaint of the opensips documentation. The reality
is that the module docs on their own arn't very good if you don't know how
opensips works. You need to go thru the examples and read the tutorials to
get a better idea of how opensips works and then the module docs will
pro
I suggest that you read up on the website, especially the tutorials under
the documentation links. Read up on some basic configs, look over the
examples provided with the source and then ask some well thought out
questions. :)
-Brett
On Wed, Aug 19, 2009 at 9:19 AM, Amon Werner wrote:
> Hi,
>
>
Is it possible that your dialog is being destroyed so quickly you don't see
that it was added to the db and then subsequently removed?
Aug 19 17:46:27 [6060] DBG:dialog:dlg_onreply: dialog 0x2d55af90 failed
(negative reply)
On Wed, Aug 19, 2009 at 8:40 AM, urmi lakkad wrote:
> Hello,
>
> I
Gabriel,You can't really set a call limit if you perform the redirects
statelessly. Basically the server wouldn't know the call count per server if
you did that.
What I see most people do here is to add some sort of header like
X-Original-IP with
the original IP of the request. Then on the destina
Actually,I just checked with gmail on a list with "The List" set and when
you hit "reply all" it ONLY places the list address in the To:
-Brett
2009/8/5 Raúl Alexis Betancor Santana
> On Wednesday 05 August 2009 14:58:32 Brett Nemeroff wrote:
> > I agre
I agree with you here. Even if it's "different" than the way most of your
other lists work, "Poster" simply makes more sense. And I don't think
another way should be picked because the poster might do the wrong thing. I
think it's up to all of us to reply to the list as appropriate.
For me, If I'm
Just don't include the auth bits. In fact, I think in the examples the auth
bits are commented out. Have you tried it?
On Fri, Jul 31, 2009 at 11:02 AM, Ghaith ALKAYYEM
wrote:
> Hello,
>
> Does anybody know how we can cancel the authentication functionality in
> OpenSIPS and make it run as a pr
For what it's worth...I've ran across tons of providers that don't like the
To URI not matching the R-URI. And yes, it's ridiculous and it means they
don't have a clue what they are doing.
You can use an actual B2BUA (sippy for example) to get around these
restrictions if you can't otherwise make
Hey All,
Weird problem here. This does not work:
$rU = $avp(s:lnp_prefix) + $rU;
When I do that, $rU appears "unchanged"
But if I do:
$rU = '' + $avp(s:lnp_prefix) + $rU;
It works fine!
Am I doing something wrong here?
Thanks,
Brett
___
Users mailin
I know no one is asking my opinion here... :) but as long as these features
are well isolated in their own modules I say the more features the merrier.
Just don't go hacking up the core to support features that were never really
intended to support. We've all seen what THAT does to these projects.
All,I was reading the thread regarding the uac_replace_from issues Jeff
brought up and was thinking my issue may be similar.
I have a carrier who sends me BYE messages with a RURI that does NOT match
the Contact header in the 200 OK. Of course, OpenSIPs replies with a 404 Not
Here.
The last messa
<
>> bog...@voice-system.ro> wrote:
>> Hi Brett,
>>
>> You mean an PV returning the list with all the available codecs ?
>>
>> Regards,
>> Bogdan
>>
>> Brett Nemeroff wrote:
>> Is there anyway to write to an AVP the negotiated codec?
turning the list with all the available codecs ?
>
> Regards,
> Bogdan
>
> Brett Nemeroff wrote:
>
>> Is there anyway to write to an AVP the negotiated codec? That'd be good
>> for CDR purposes. Would I need a bunch of codec_exists in the on_reply route
&g
Is there anyway to write to an AVP the negotiated codec? That'd be good for
CDR purposes. Would I need a bunch of codec_exists in the on_reply route
checking for 200 OK?
On Thu, Jul 23, 2009 at 4:46 AM, andrei dragus wrote:
>
> Hello,
>
> Methods have been added for SDP codec manipulation in the
Read the docs on the dialplan module:
http://www.opensips.org/html/docs/modules/1.5.x/dialplan.html#id271074
On Tue, Jul 21, 2009 at 10:44 AM, Julien Chavanton wrote:
> Do you have a suggestion on how to do this more dynamicaly ?
>
>
> --
> *From:*
Once again.. another good use for the dialplan module. :)
You really don't want to do this with a static subst I don't think..
-Brett
On Tue, Jul 21, 2009 at 8:30 AM, Julien Chavanton wrote:
> Thank you, we will move to 1.6.0 later.
>
> We have partner not sending tech prefix, we need to add it
I too am very curious how this would perform.. Please let us know! :)-Brett
On Tue, Jul 21, 2009 at 5:54 AM, Alex Balashov wrote:
> DangVinh Nguyen wrote:
>
> > From my old experience, 100 calls recording per server is a good
> result.
> >
> > This stuff is hdd-bounded, not cpu-bounded. With o
It may be worthwhile speaking with the devs for sipp to see if they can make
an adjustment to the default scenarios to account for this. It's a real
pain. I seem to remember running into this alot. I usually mock up my own
scenario with the RURIs changed..
-Brett
On Thu, Jul 16, 2009 at 11:29 PM
Sure,I think this is a real need with plenty of demand. But it's not a
solution that a proxy alone can fix, which is what OpenSIPs is.
There are likely ways to do it. I do a distributed queue system using
libcurl with a PERL/MySQL backend to track the call's position. I perform a
call loop on each
sn't need to be
> a reliable data store. Just need to be able to update a value here and be
> able to read it there.
> This module puts ya'll real close to the top of my "pretty cool people"
> list. Thank you.
>
>
> Richard Revels
>
> On Jul 15, 200
I think it's worth re-iterating that memcache is NOT meant to be a reliable
data store and you should essentially build your applications assuming the
data will NOT be available. Doing some reading on memcache is very
worthwhile for proper use of this fantastic capability. The use of OpenSIPs
using
Thanks! This is great!-Brett
On Wed, Jul 15, 2009 at 8:57 AM, andrei dragus wrote:
>
>
> Hello,
>
> A new module that provides a new caching method using memcached servers was
> added.
> It provides a way to access memcached servers using the existing memcache
> API.
>
> Advantages over the exis
20 5.6.7.8 0 102 NULLExternal gateway 2
>
> If a call comes from 1.2.3.4 I would like to prepend "101" before sending
> the call to the "Internal gateway".
> If on the other hand the call comes from 5.6.7.8 I would like to prepend
> "102
That should be done automatically by the do_routing function, unless of
course you are writing the output of that function to an AVP instead of a
RURI. Where it'll end up in the AVP.
-Brett
On Sun, Jul 12, 2009 at 7:40 PM, Lasse Johnsen wrote:
> Hi,
>
> I use the is_from_gw function in drouting
There is no question that OpenSIPs is a complicated project and that the
beginner level docs are mediocre. However, that's mainly because the
documentation is written such that it *assumes* that the reader has a decent
knowledge of RFC3261.
I think if you really know your SIP.. and you download the
It's been a while since I used the perl module, but I think you can't
manually call those scripts from the command line.
However you can try to copy/symlink the OpenSIPs perl modules into your @INC
path to see if that helps. As Ghaith mentions, you shouldn't have that
problem with OpenSIPs itself c
s because I tried to implement something similar with a
> well-known switch once (I think it was a Metaswitch) and the signaling agent
> reacted to my "spiral" (which I didn't know to be such) as though it were a
> "loop."
>
> Brett Nemeroff wrote:
>
>
Just throwing this out.. Not all equipment can handle SIP Spiral properly.
asterisk (although I know there was work done on
Asterisk+SIP Sprial, I don't know where that ended up)
so be careful before you spend a lot of time on that. I'd love to hear how
all of that works for you. I've got plans
Sorry all,I'm not setting db_missed_flag. whoops!
-Brett
On Fri, Jul 10, 2009 at 10:25 AM, Brett Nemeroff wrote:
> All,I had some problems a while ago with ACC and I thought I had them
> resolved, but looks like I'm still having issues..
>
> I've set failed_transa
All,I had some problems a while ago with ACC and I thought I had them
resolved, but looks like I'm still having issues..
I've set failed_transaction flag (14) and db_flag (15).
I arm both 14 and 15 at the top of my routing script. Then in my failure
routes, I re-arm 14.
My ACC looks pretty good,
I know this may sound like a pretty lame answer, but you'll get a lot of
benefit from reading the definition of a SIP PROXY from RFC3261.
You can't do much with OpenSIPS (properly) if you dont' know the underlying
RFC. This is very different from other SIP software packages, like Asterisk
where you
lowed_packet but for some reason I miss how to
> configure libmysqlclient one (the one php is using). Is this something that
> I miss from the link you have sent me?
>
> Ta,
> DanB
>
>
> On Wed, Jul 8, 2009 at 7:37 PM, Brett Nemeroff wrote:
>
>> Quick go
Quick google got me this:
http://www.astahost.com/info.php/Max_allowed_packet-Mysql_t2725.html
??
-Brett
On Wed, Jul 8, 2009 at 11:34 AM, DanB wrote:
> Adrian,
> I spent quite a long time searching for a way to modify this setting, but
> could not find it anywhere.
>
> One extra info I found
I'm sorry if I gave you the impression that I was sending you away or
telling you it's too hard to handle. That certainly wasn't my intention.
Yeah, we all learned the same way. Even now, I have a decent idea what I'm
doing and I ask stupid questions all the time! :)
The important thing I was tryi
Bogdan,Can you provide some examples on the wiki for this? :)
I've wanted to implement this in opensips for some time now.
-Brett
On Tue, Jul 7, 2009 at 3:19 AM, Bogdan-Andrei Iancu
wrote:
> Indeed, but you should be also able to do onhold with OpenSIPS +
> RTPproxy only. The rtpproxy has some
I think I'd like to jump on the bandwagon here also
As said by some of the other members on the list, this IS a very big
project. The fact that you are looking for an ISO to test it out, suggests
that you don't really understand how it all works. Without going into too
much detail, let me just
I think you are having a fundamental problem with this. I'm not sure how
you'll actually be able to configure opensips if you can't get past this
part.
127.0.0.1 is a special loopback address. It's not really used for
INTERmachine communications. Therefore, if you want to watch packets on the
loopb
URI is from a server or gateway which manage Number Portability data.
> The rn is the routing number used to route the call (LRN) and the npdi is
> the Number Portability DIP indication (means that the NP DIP was done).
>
> Ragards,
> Juan
>
>
> Iñaki Baz Castillo wrote:
>
tp://www.ditechnetworks.com/products/packet-voice-processor.html
>
> Richard Revels
>
>
> On Jul 3, 2009, at 11:33 AM, Brett Nemeroff wrote:
>
> I too am curious about this.. High volume transcoding.. something that ISNT
> asterisk. :)
> -Brett
>
>
> On Fri, Jul
I too am curious about this.. High volume transcoding.. something that ISNT
asterisk. :)
-Brett
On Fri, Jul 3, 2009 at 10:05 AM, Julien Chavanton wrote:
> Thanks, I understand this is out of scope for Opensips,
> I am going to test a transcoding board full of DSP with an on board NIC to
> handl
Hey All,I've got a stateful dispatcher I've build using dialog profiles.
Apparently I'm doing something stupid in my script generating these errors:
Jul 3 00:54:20 opensips-a /usr/local/sbin/opensips[26189]:
ERROR:dialog:set_dlg_profile: dialog was not yet created - script error
Jul 3 00:54:20 op
Bogdan,How will it work?
-Brett
On Wed, Jul 1, 2009 at 4:54 AM, Bogdan-Andrei Iancu
wrote:
> Hi Julien,
>
> soon, a new module for topology hiding will be available in OpenSIPS
> 1.6.probably this is what you are looking for..
>
> Regards,
> Bogdan
>
> Julien Chavanton wrote:
> > Hi, I have
hello all,I'm using the TLS version of opensips 1.5.1
I can't seem to find an actual source reference where has_sdp shows up. What
module/version is this in? Docs show it in 1.5.
-Brett
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ett
On Tue, Jun 30, 2009 at 12:48 PM, Uwe Kastens wrote:
> Hi Brett,
>
> Could be broken libs. I would start opensips with strace and look for
> errors.
>
> Which OS are you using?
>
> BR
>
> Uwe
>
> Brett Nemeroff schrieb:
> > Still can't connect :(
x27;,'N','N','N','N','N','N','N','N','N','N','','','','','0','0','0','0'
On Tue, Jun 30, 2009 at 11:01 AM, Uwe Kastens
ences_priv | Y | | X509_issuer | ? |
>| Index_priv | Y | | X509_subject| ? |
>| Alter_priv | Y | | Max_questions | 0 |
>| Show_db_priv| N | | Max_updates | 0 |
>| Super_priv | N | | Max_connections | 0 |
>|
connect via mysql client.
>
> Your /etc/hosts have the entry for localhost?
>
> BR
>
> Uwe
>
> Brett Nemeroff schrieb:
> > yeah, I tried localhost, 127.0.0.1, and the actual ip (I usually use
> > localhost)
> >
> > here's my connect string:
> >
Hey all,sorry for such a noob question here, but I just can't figure out
what I'm doing wrong.. I'm getting the error:
Jun 30 15:36:33 nguenj297 /usr/local/sbin/opensips[10159]:
ERROR:db_mysql:db_mysql_new_connection: driver error(1045): Access denied
for user 'opensips'@'localhost' (using password
difficult is to re-design the pike module, as so far,
> the way the internal data is kept is highly IP-format dependent.
>
> Regards,
> Bogdan
>
> Brett Nemeroff wrote:
>
>> Yeah, that's a great idea actually, I could just concatenate some PVs to
>> form a key l
8 PM, Iñaki Baz Castillo wrote:
> 2009/6/26 Brett Nemeroff :
> > All,
> > I've got a customer that is sending me calls with an RURI like this:
> > sip:1311207;npdi=yes;rn=1310...@1.2.3.4
> >
> > when I use anything that parses the RURI like $rU, it s
string you build form script, right ? this
> string will be a kind of key (logical one) to identify the loop.
>
> Regards,
> Bogdan
>
> Brett Nemeroff wrote:
>
>> Hey All,
>> I was wanting to submit a feature request for loop detection. Specifically
>> NOT SI
Is this move to char flags going to be global? It's nice to see the config
syntax becoming a little more friendly. Are we losing anything in
performance that way?
Thanks for all the great work!!
-Brett
On Mon, Jun 29, 2009 at 7:39 AM, Bogdan-Andrei Iancu wrote:
> Hi,
>
> There were couple of ch
This is likely a result of a failed regex. What's the regex you are using?
On Fri, Jun 26, 2009 at 1:39 PM, Opensips Philippines wrote:
> I am trying to activate the dial plan module but I am having this error.
>
> Initially, I tried version 1.5.1 then removed it. I tried it with 1.5.0
> but sti
^(1[0-9]{10}).+
| \1 | |
++--++--+-+---+-+--+---+
Calls now fail with a 500 Overlapping request. SIP Trace shows me receving a
200 OK and it not being relayed I don' think..
Any ideas?
-Brett
On Fri, Jun 26, 2009 at 2:06 PM, Brett Nemeroff wrote:
> All,I've got a customer that
All,I've got a customer that is sending me calls with an RURI like this:
sip:1311207;npdi=yes;rn=1310...@1.2.3.4
when I use anything that parses the RURI like $rU, it shows
$rU=sip:13151207;npdi=yes;rn=131
Which is exactly everything from sip: to the first @ sign. The custome
Hey All,I was wanting to submit a feature request for loop
detection. Specifically NOT SIP loop detection, but when another technology
/ B2BUA is involved where max-forwards can't be used. This is for big
loops.
The idea is similar to the pike module. However, you bascically look at the
to_did and
All,What's the actual status of the load balancer module? I see in 1.5 it's
alpha/NEW. Is it usable yet? I don't know how conservative those labels are.
-Brett
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e and enjoy it! Thanks to all who made
> (and documented) OpenSIPS.
>
>
>
> Hopefully this summer I will have some free time when I could write all my
> notes “New to OpenSIPS” into actual readable text, with examples. So that
> other newbies have something to work from.
>
I said this in an earlier message and I think it's worth saying again.. If
your code REALLY IS 10,000 lines long. You are probably doing something
wrong or at least, the hard way.. Maybe we can help you out there.
-Brett
On Wed, Jun 17, 2009 at 2:18 AM, Matti Zemack
wrote:
> Hi,
>
> Thanks. Will
All,
I ran into an interesting crash in my test env today.
I took a very large acc table and reindexed it with a complicated index..
while this was going on I placed calls and got the following errors:
Jun 17 01:47:40 voicefox-dev /usr/local/sbin/opensips[25170]:
ERROR:db_mysql:db_mysql_do_prepare
All,I'm getting the following error:
Jun 16 12:42:28 voicefoxtelephony /usr/local/sbin/opensips[13281]:
ERROR:db_mysql:db_mysql_do_prepared_query: mysql_stmt_bind_param() failed:
Using unsupported buffer type: 7610151 (parameter: 16)
Jun 16 12:42:28 voicefoxtelephony /usr/local/sbin/opensips[13281
Jeff, I hadn't thought of those scenarios.. I suppose with the dialog module
you can do some pretty neat stuff.. this is a great idea..
On Tue, Jun 16, 2009 at 9:03 AM, Jeff Pyle wrote:
> Hi Raúl,
>
> It comes down to the service we're offering. When a customer buys a PRI
> from us, they don't
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