I've been playing around with the new Topology Hiding functionality in the
dialog module.
Now I am just merging into my script, but I noticed the following. When
topology_hiding() get's called, it seems that the Contact header looks odd.
Assume:
a.b.c.d to be the IP address of the Asterisk Ser
Vlad,
Just updated from trunk... looks good. Thanks again for the quick fix!
Regards,
AF.
On Wed, Aug 3, 2011 at 4:29 AM, Vlad Paiu wrote:
> **
> Hello Alan,
>
> About the first issue, when the timeout is in the past and the timestart is
> 0. Both these values are updated at the 200 Ok recei
Originally I came across this issue in 1.6.4 and documented it in
http://opensips.org/pipermail/users/2011-January/016358.html
I've seen a few entries in the dialog DB in 1.6.4 where the call was an old
zombie because the start_time was ahead of the timeout. Now I have been
experimenting with 1.7
I'm using the LB module to load balance INVITES to an Asterisk farm.
I've had some naughty UA's who for one reason or another do not
properly "BYE" the call and I've had my LB resource list fill up with
zombie calls.
Now I want to implement SIP Session Timers into my configuration. As
the LB modu
Now I can't say I have a complete understanding of OpenSIPS but was
always wondering about the following code I have seen in various
sample configs. Loose routing and to_tag's have always confused me...
so I am wondering about how secure this is:
Now take the following code:
if (has_tota
ogdan
>
> On 04/11/2011 05:22 AM, Alan Frisch wrote:
>>
>> For some odd reason my OpenSIPs load_balancer module seems to fail
>> with certain clients. Works fine with Asterisk and some other
>> servers, but have had the occasional client get a 503 returned to it.
>>
FYI I mashed up the IPs in the SDP for privacy reasons... so that's why
they look weird. :)
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For some odd reason my OpenSIPs load_balancer module seems to fail
with certain clients. Works fine with Asterisk and some other
servers, but have had the occasional client get a 503 returned to it.
In this case, it's a SIP application for an Android phone... I get a
"bug - cannot find request res
gt;
> Could you post the out of the fifo cmd dlg_list ?
>
> Regards,
> Bogdan
>
> Alan Frisch wrote:
>>
>> Bogdan,
>>
>> The dialogs are still showing in memory with the FIFO command and in
>> MySQL.
>>
>> I have the following in my opensips.
:07 AM, Bogdan-Andrei Iancu
wrote:
> Hi Alan,
>
> So those dialogs were removed from memory (you cannot see them listed by
> "opensipsctl fifo dlg_list"), but they are still in DB ? If so, what are the
> values for "timeout" field in DB for that those dialogs ?
>
&
Using the LB module with Dialog and using FreeSwitch/Asterisk to
handle my media.
Under 1.63 I had the occasional phantom call that hung around, but
would expire based on the dialog expiry value I had set.
When I upgraded to 1.64 the only change to my opensips.cfg that was
made was to remove the
You might get a better response if you kept your subject line shorter
and grouped all your messages into one.
AF.
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.
On Mon, Jun 21, 2010 at 5:30 PM, Saúl Ibarra Corretgé
wrote:
> Hi,
>
> On 21/06/10 19:03, Alan Frisch wrote:
>> Been running Mediaproxy in production for the last 12 hours with some
>> heavy loads and have noticed that there are many "zombie" media
>> sess
Been running Mediaproxy in production for the last 12 hours with some
heavy loads and have noticed that there are many "zombie" media
sessions listed when using the CDRTool page to view current MP
sessions.
These calls no longer exist in OpenSIPS dialog table nor the media
servers (Asterisk boxes)
Jeff,
Right on the money... that's the way I ended up approaching the issue.
Just put a "mapping table" of sorts using if else's in the OpenSIPSs
config. Not the most elegant solution, but it works.
Now to tackle my next problem... "zombie" MP sessions.
AF.
On Mon, Jun 21, 2010 at 8:14 AM, Je
I've been experimenting with Mediaproxy 2.4.2 on a couple servers and
have hit some kind of snafu regarding the IPs that the relays take on
a machine.
Machine A
eth0 - 209.x.x.100
lo1: loopback IP 1 - 64.x.x.120 (MP Dispatcher)
Machine B
eth0 - 209.x.x.200
lo1: loopback IP 1 - 64.x.x.220 (MP Rel
1.6.2-tls compiles fine on my CentOS 5.3 Box.
On Thu, Mar 11, 2010 at 4:14 PM, erik pepermans wrote:
> Hi,
>
> When compiling the perl module on version 1.6.2-notls on Centos :
>
> make prefix=/ all
>
> I receive :
>
> ...
> make[1]: Entering directory `/usr/src/opensips-1.6.2-notls/modules/perl'
I am using CDRTool and am noticing that certains calls from one
particular user get rated as a "Free Call".
The user makes a call to a PSTN number (probably his own) and upon
answer, hangs up immediately (probably some kind of system monitoring
call). Since the call is answered (200 OK) and hung-
The docs still say MP requires Python 2.4 or higher... but while
running this 2.4.1 on Python 2.4 I get the below popping up
occasionally. Seems to be related to the use of the all() command in
the dispatcher.py script. AFAIK this command is only available in
Python 2.5+.
Feb 28 04:06:22 pbx med
Thanks!
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Went and DLed the latest MediaProxy (2.4.0) and upon running
./setup.py build end up with the following error:
gcc: mediaproxy/interfaces/system/_conntrack.c: No such file or directory
This existed in the 2.3.10 .tar.gz archive.
If I copy the missing file(s) from the 2.3.10 archive, it further
c
I've been fooling around with OpenSIPS and regex pattern matching on the R-URI.
For example, this code allows for any number of numerical characters:
if(uri =~ "^sip:[0-9]+@")
{
whatever
}
What I am trying to do is prefix a value if the inbound number comes
in with an E164 format. I tried the
Bogdan,
Thanks for the info. I load the RPID with the modparam("auth_db",
"load_credentials", "rpid") and put it into $avp(s:rpid).
As long as OpenSIPS is in forked mode, it works fine. But when I was
running it in non-forked mode is when I saw the retention behavior.
Seems the RPID would stick
Been pulling out what's left of my hair on this one...
I'm trying to get the siptrace module to record calls only using the
trace_dialog() command at the initial invite. When I put the command
in the invite route, everything works as it should.
Now I am trying to get OpenSIPs to use the avp_trac
wrote:
> Depends on your db_mode for the registrar.
>
> On 01/14/2010 03:07 PM, Alan Frisch wrote:
>
>> Using OpenSIPs 1.6.1 and have noticed a peculiar issue.
>>
>> If a subscriber has a value in the RPID column, then that value is
>> changed to NULL it s
Using OpenSIPs 1.6.1 and have noticed a peculiar issue.
If a subscriber has a value in the RPID column, then that value is
changed to NULL it seems that OpenSIPs still retains the previous
number.
For example:
If RPID column is NULL (initially) ---> if ($avp(s:rpid))) returns FALSE
If RPID colum
Looking at deploying a test bed with a couple servers.
Just wondering which is more CPU intensive... the server running
OpenSIPS or the servers proxying the media (no codec translation)?
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Baz,
Many thanks... your explanation really cleared it up for me. So I
guess in the example I cited, the fact that $du is empty is okay
(since it is a direct routing to the gateway).
AF.
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I used the "SIP Proxy/Registrar with Offnet-Termination and
Accounting" configuration from the Sipwise Wizard. Rather than
gumming up the list with what it spits out there, here is the relevant
bits when issuing a call to one of my gateways:
###
I've been using the scripts generated by the Sipwise Wizard to learn the LCR
module and OpenSIPS in general. Just sending requests from OpenSIPS to two
Asterisk test servers I set up.
While I get the general concept of URI's, I am trying to discern exactly
what a Destination URI is within the Ope
Robert,
That worked like a charm. Thanks!
On Fri, Mar 27, 2009 at 2:02 AM, Robert Borz wrote:
> Hi Alan,
>
> this issue sounds a bit like the one I had some time ago... I also started
> with a sample configuration from the sipwise wizard.
>
> See http://lists.opensips.org/pipermail/users/2009
>Try to send a BYE inmediatelly after answering the call. Does the error also
>occur?
Indeed, I tried having the calling end hang up almost immediately
after it connected to the gateway as well as 1 minute connection...
both yield the same result.
I can suppress the message by editing the releva
I'm still learning OpenSIPS and have come a long way from a few weeks back!
Right now, I am using one of the scripts from the Sipwise OpenSER script
generator. With some tweaks to bring the script up to OpenSIPS standards I
am now using a basic PSTN gateway script (LCR) from the site. The script
Been playing with the Load Balancer module this weekend (thanks again to
Bogdan for fixing the MySQL bug I posted a few weeks back) and was wondering
how one might be able to do failover with this module?
I'm only new to OpenSIPs but am trying to figure this one out... With the
LCR module, it is
I've been experimenting with OpenSIPs new Load Balancer module and have run
into a bit of a snag. Used the configuration from the Tutorial on Load
Balancer, though needed some lines added as it seems incomplete (ie.
load_balancer module needs to be added to the loadmodule section).
When I attempt
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