I'm wanting to record the response sent back to the call originator, not
just the last winning response selected by tm module. There are cases where
opensips translates like a received 503 to a 500 or like if you don't trust
the response you received from a vendor that is not using sip codes
Luis,
forward() does not accept a variable for an argument.
instead use like:
$du = sip: + src_ip + :5070;
I think it needs URI format without the user portion.
see: http://www.opensips.org/Resources/DocsCoreVar17#toc34
There are a bunch of older functions that do not accept variables as
Have you checked if relay and dispatcher are actually running?
Have you run them in the foreground with debug turned on to see why
use_media_proxy returned an error?
I'm betting Brett is right that radius is at least primarily the problem.
I would disable it's use in opensips.cfg and
avps are per transaction. A branch is just part of a transaction.
Until the transaction is terminated the avp stays active. Be aware
that a call's avp is write-able in all branches and assigning a value
to an existing avp pushes the new value on the top of the avp array.
Calls do not go back to
Speaking of suggestions... (understandable probably won't make it in
1.7, more likely 2.0)
I've been thinking about variables in the script and a couple other things.
1. New variables for direct read/write access to value currently only
available by function (very beneficial for example with cdrs)
Toyima,
What I've done to deal with errors on a restart is to copy the config
to a temp, replace a couple things like fifo file and ports, start it
with foreground option, piping output to check for errors. The group
is backgrounded with and the init script waits for 2 sec. If bk
process is
Erik,
Just typed this up so probably has some mistakes. :-)
This form pulls all alias db results into an avp which are looped
though to do a lookup for each that only a user alias. You could do
the append branch mode for alias_db but then lookup would have to be
in branch route and Im not sure if
on the second
branch (192.168.131.129:5060 - 192.168.131.129:5060)...as i can see, the DB
alias is correctly fetching both users from DB, but only the first one is
resolved via lookup(location) (registration search).
any suggest?
Many thanks!
2011/5/18 Dave Singer dave.dorasin...@gmail.com
Toyima
Spencer,
Looks like you just need to move $rU=$oU; into branch_route[2]
Note that in failure route it will be reset to what it was before
changes done in branch route so you may want to do it in main route
and in branch route.
Dave
On Thu, May 19, 2011 at 4:02 PM, Spencer Thomason
Nick,
Have you verified there is no firewall on the opensips server with
iptables -L
I believe the dump pulls the packets before they go through iptables.
So what you capture may be just dropped by iptables and not reaching
opensips.
Though really that shouldn't be a problem since iptables
Toyima,
check:
http://www.opensips.org/html/docs/modules/1.6.x/alias_db.html#id250030
Dave
On Tue, May 17, 2011 at 6:44 AM, Toyima Dias toyim...@gmail.com wrote:
Is it possible to add the same alias to more than one subscriber? for
example, if an incoming call to the number 14882736524 is
Brett,
I believe you need the full path to the reply fifo.
Try:
echo -e :address_dump:/tmp/my_fifo\n\n /tmp/opensips_fifo
you may also have a permissions ( chmod a+w /tmp/my_fifo ) and/or SELinux
issue (very likely if SELinux is enabled, I posted a howto for SELinux in
the list a number of
:39 PM, Dave Singer dave.dorasin...@gmail.comwrote:
Brett,
I believe you need the full path to the reply fifo.
Try:
echo -e :address_dump:/tmp/my_fifo\n\n /tmp/opensips_fifo
you may also have a permissions ( chmod a+w /tmp/my_fifo ) and/or SELinux
issue (very likely if SELinux is enabled, I
Looks like you sent your request to the wrong list.
On Mon, May 9, 2011 at 12:24 AM, abid khan abidkhan...@gmail.com wrote:
Hello everybody,
I want to know main criteria or parameters which
i have to fulfill if i want to register* pjsip dialer* with openIMS core.
and
become unavailable. I've seen this with the opensipsctl script as well, where
the fifo just stops responding until you restart.
-Brett
On Mon, May 9, 2011 at 4:03 PM, Dave Singer dave.dorasin...@gmail.com wrote:
Are you actually able to get results out of the reply fifo?
It's been a long
Bret,
try:
ps aux --forest | grep opensips
that will show you parent and child processes in tree diagram and if any
opensips master process is actually responsible for so many of them.
I agree that it is probably being spawned many times with a cron job, loop
in the init script or a service like
Jeff,
Sounds like the bye is not getting matched with the dialog. There are some
settings for dialog module for matching fallback that you probably want to
look at.
Also there are a number of threads that have discussed matching problems
that may be helpful.
Dave
On Tue, May 3, 2011 at 2:46 PM,
Jan,
I'm assuming you actually meant $tU which is the user portion of the To URI.
If all you are trying to do is test if they are numeric then regex is the
best way to test it.
Something like:
if ( ! $tU =~ ^[0-9]*$ ) {
... deal with non numeric user ...
}
Dave
On Wed, May 4, 2011 at 2:07
Marcello,
Might check into getting an IP with BGP (Border Gateway Protocol) or some
other IP routing protocol. Not all ISPs offer these services and it is
usually offered on a subnet of addresses not a single address. If your
providers/contacts can't do this or don't have good answers you might
Jeff,
I would be interested in working with you. I've been integrating
OpenSIPS in hosted VOIP environment for 4 years. Unfortunately I will
be laid off the end of this week, Apr 22 (I'm the last one to go).
You can look at my resume on Linkedin (
http://www.linkedin.com/in/davesingercando ) or
I have a server running 1.6.3. In failure route I add a Diversion and
the RURI is a DNS SRV record and sometimes one of the servers times
out and opensips auto tries the second server from the SRV records but
without the diversion header in the new message
Is this proper/expected behavior or a
Cinthia,
The CSeq needs to be changed before it hits opensips. So you could do
something where if the cancel is not matched, lookup the right cseq
using a memcache lookup (that you stored during the accepted invite)
and send it out to it self with the modified CSeq. I wouldn't be
surprised if
installed in the same machine)
Checking these files db.inc.php ; local.inc.php; globals.php in the CP
config directory /var/www/opensips-cp/config
I don't see anywhere to config CP listening port !
Please help me to figure out, thanks!
2011/3/24 Dave Singer dave.sin...@wideideas.com
Duong,
Yea
/3/24 Dave Singer dave.sin...@wideideas.com
Duong,
The MI commands are for opensips and ps does NOT take any arguments.
The easiest way for it to connect when on the same box is with fifo.
Here is a snippet from the boxes.global.inc.php:
/* DEFINITION OF BOXES (servers
contained records of
online users
(not the presence module as you suggested)
Please tell me what are the next steps to check ?
Thanks!
2011/3/25 Dave Singer dave.sin...@wideideas.com
Duong,
Sorry. I forgot I modified my CP and those options.
You need to have it:
$boxes[$box_id]['mi']['conn
!
--
Message: 1
Date: Tue, 22 Mar 2011 21:06:21 -0700
From: Dave Singer dave.sin...@wideideas.com
Subject: Re: [OpenSIPS-Users] Opensips Control Pannel does not take
effect into Opensips Server
To: OpenSIPS users mailling list
.
Date: Mon, 21 Mar 2011 08:45:03 -0700
From: Dave Singer dave.sin...@wideideas.com
Subject: Re: [OpenSIPS-Users] Opensips Control Pannel does not take
effect into Opensips Server
To: OpenSIPS users mailling list users@lists.opensips.org
Message-ID
Duong,
Some modules, like drouting, you have to click a reload module button
in the control pannel.
However, I think your problem is that you haven't implemented the
needed stuff in opensips.cfg script.
CP only provides an interface to managing/accessing the databases
tables. You still have to
JP,
The two end points have to agree on an RTP protocol.
You may look at the SDP in the initial INVITEs to see if T.38 is offered and
if it is, remove it.
Also if the re-INVITE contains G711 along with the T.38 you should be able
to just remove the T.38.
That should accomplish the same effect as
Igor,
I'm pretty sure that won't work at all. You would need to use the
transformation ip.resolve.
I don't know if it returns multiple IPs but if it does you would want to
assign it to an avp that can handle multiple.
An example to try:
$var(fqdn) = some.server.com; # Not sure if you can do a
openSIPS failed to start.
*From:* users-boun...@lists.opensips.org [mailto:
users-boun...@lists.opensips.org] *On Behalf Of *Dave Singer
*Sent:* Friday, February 25, 2011 4:14 AM
*To:* OpenSIPS users mailling list
*Subject:* Re: [OpenSIPS-Users] running opensips under another user
Igor,
Very close. Notice the ( between $ and var and ) after var name:
$var(fs_addr) = $(var(fs_addr_s){ip.resolve}{ip.ntop});
see syntax near top of http://www.opensips.org/Resources/DocsCoreVar16
Dave
On Thu, Feb 24, 2011 at 10:36 AM, Igor Solovyov i...@choochee.com wrote:
Hi All,
I try
method to get resolved ip by string?
Igor.
On Thu, Feb 24, 2011 at 11:06 PM, Dave Singer
dave.sin...@wideideas.comwrote:
Igor,
Very close. Notice the ( between $ and var and ) after var name:
$var(fs_addr) = $(var(fs_addr_s){ip.resolve}{ip.ntop});
see syntax near top of http
Anton,
Have you checked the selinux logs?
Search the list about selinux. A couple months back I posted what I ran into
and how to solve it fairly simply.
Dave.
On Thu, Feb 24, 2011 at 6:23 AM, Duane Larson duane.lar...@gmail.comwrote:
And you're positive that the /var/run/opensips directory is
Ronald,
The only time I've seen it be null in failure route is if there was no reply
received,
you might add a check to see if it was a local timeout:
if ( t_local_replied(all) ) {
xlog(did not get any response);
} else {
xlog($(replyci): $C(rx)failure route: $(replyrs)
$(replyrr)$C(xx)\n);
I tested setting up acc module to use the cdr_flag with dialog module.
Works nicely until opensips is restarted while there is an open call.
After an opensips restart, calls that were started before the restart do not
get an entry in the DB. Not even an old style BYE record.
I verified that the
/pipermail/users/2010-November/015448.html) to
another person and seems that he never solve the situation. Any help will be
appreciated.
2011/2/18 Dave Singer dave.sin...@wideideas.com
Toyima,
It is looking for those executable files. It looks like it first tries
db_dump then because that fails
Jason,
That is very strange behavior. If there is no matching transaction I
think there is no way to catch the message as reply and failure routes
are triggered by tm module.
I'm not sure since I haven't setup a stateless config.
I'm curious too if there is something that can be done here.
Dave
Dave,
2011/2/17 Dave Singer dave.sin...@wideideas.com
Toyima,
Have you configured it to connect to your database?
the file opensipsctlrc is already configured to connect to my database
where i'm storing domains, locations, subscribers, etc etc...the DB_PATH
parameter commented and i
Toyima,
This is the right list. You just have to take into account that no one
is getting paid to answer your questions in the list. And sometimes
that means waiting a couple days.
Best tactic I've seen to keep putting it in front of people is to
respond to your own thread with a little update of
Juri,
If you are looking to get opensips1 itself to respond to a
proxy_challange of opensips2 your looking at a headache. Any way you
go you'll have to change the way things work to make a special case
for opensips1 on opensips2.
My suggestion would be to, where you are about to do the
Toyima,
On Wed, Feb 16, 2011 at 5:55 AM, Toyima Dias toyim...@gmail.com wrote:
2011/2/16 Dave Singer dave.sin...@wideideas.com
Toyima,
This is the right list. You just have to take into account that no one
is getting paid to answer your questions in the list. And sometimes
that means
NULL
Regards
Juri Nysschen
-Original Message-
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Dave Singer
Sent: Monday, February 14, 2011 10:16 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] FW: CANCELs
Toyima,
The client does NOT have to use NAPTR or SRV. It can use regular A or
CNAM records as well or even just an IP. (unless it is a half baked
client)
SRV and NAPTR are special DNS records that can be used tell the client
what services are available and where and how to connect to them. If
your
, 2011, at 10:45 PM, Dave Singer wrote:
Adrian,
Probably want to only respond to registers that are to valid user
accounts, drop the rest, as they start scanning with like 100, 101,
., 5000, etc
Dave
On Fri, Feb 11, 2011 at 6:25 AM, Adrian Vasile y...@opennet.ro wrote:
Hi Dave
.
The key is finding the reason why the transaction id disappears or at least
be able to put it back when delivering the CANCEL downstream.
Regards
Juri Nysschen
-Original Message-
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Dave Singer
Juri,
You say it only happens after a do_routing(). To be clear, you mean
that you used do_routing on the invite and not that you already tried
do_routing for the cancel earlier in the script. (I'm not sure it
would make any difference)
The hack to store the destination in a variable is one you
REGISTER requests from anywhere and
the rest check the source ip.
Great ideea.
I will implement it as soon as possible.
Thanks,
Adrian Vasile
y...@opennet.ro
On Feb 10, 2011, at 10:41 PM, Dave Singer wrote:
Adrian,
I was just thinking about the implementing no response for INVITE
Adrian,
There are lots of people out there with servers doing sip scans to see
if an ip will respond to a sip ping (NOTIFY or OPTIONS message). Then
they will either try to send register and/or invites for all sorts of
numbers trying to get a hit. Of course the invites are not actual
calls so if
is required for initial incoming INVITE
requests from the SIP proxy.
On Feb 10, 2011, at 6:57 PM, Dave Singer wrote:
Adrian,
There are lots of people out there with servers doing sip scans to see
if an ip will respond to a sip ping (NOTIFY or OPTIONS message). Then
they will either try to send
opensips reboots and other situations where the
cache is lost or unavailable. Like a memcached server fails.
Advantage to using external memcached vs local cache would be that
cache would not be cleared on opensips restart.
Dave
On Thu, Feb 10, 2011 at 11:16 AM, Dave Singer dave.sin
Cinthia,
I'm fairly new to OCP myself, just got it running about a month ago.
Since no one with more experience responded, I'll help if I can.
Sounds like you have it all setup and working as designed.
Looking at my system that I have connected to two servers, it looks
like the dialog module
mato848,
The default/sample config has a section for registration. you need to
basically use that. just copy it to your script. Look for
www_authorize.
Registrations are stored with the usrloc module and auth is done by
auth and auth_db modules so they need to be configured as well like
the
Jeff,
I checked nathelper code in opensips 1.6.4 and looks like adding the
oldip header to the SDP is not there anymore.
So you shouldn't need to modify the source like I had to previously to
accomplish complete topology hiding.
Also it doesn't really matter if you use rtp_proxy or media proxy
Anton,
My bet is that since the first one was dropped, a transaction was
never created and thus the transaction cleanup doesn't happen either.
If you want it to it do the automatic cleanup it might work to create
a transaction prior to the reply, then use t_reply and finally drop.
Dave
On Tue,
Kyle,
use t_replicate(sip:second server);
Look at the tm module docs for a more complete picture.
Then modify the message. One branch will go out with the message the
way it was before the t_replicate and one will have the changes after
the t_replicate.
If you need more replicated destinations,
, Dave Singer dave.sin...@wideideas.com
wrote:
Don't know what tools you are familiar with so here are some
suggestions for what they're worth.
Appreciate the input!
Am familiar with all - but included output below - always happy to
have another set of eyes on things ;-)
what is listening
, Feb 3, 2011 at 4:32 AM, Stefano Pisani
stefano.pis...@omnianet.it wrote:
Hi Dave
you could try
if ($si == $hdr(X-src-ip)){...}
Il 03/02/2011 12:59, Bogdan-Andrei Iancu ha scritto:
Hi Dave,
Unfortunately does not work with variables.
Regards,
Bogdan
Dave Singer wrote:
Wow I missed
need to re-read the docs
On Sun, Feb 6, 2011 at 6:48 PM, Dave Singer
dave.sin...@wideideas.comwrote:
Isn't
$var(my_uri) = 5...@example.com;
$ru = sip: + $var(my_ruri);
the same as
rewriteuri(sip:5...@example.com);
just that using $ru you can use it just like assigning to other vars
like you
On Mon, Feb 7, 2011 at 7:05 PM, Ovidiu Sas o...@voipembedded.com wrote:
On Mon, Feb 7, 2011 at 9:14 PM, Chris Stone axi...@gmail.com wrote:
Sorry all for the last message - too quick on the Send button.
On Mon, Feb 7, 2011 at 6:48 PM, Henk Hesselink opensips-us...@voipro.nl
wrote:
Hi
would have to modify
the OpenSIPs core to make it happen. Not worth the time or effort for this
reason.
On Tue, Feb 8, 2011 at 3:10 AM, Dave Singer dave.sin...@wideideas.com
wrote:
The 100 Trying is sent out to the originating server/user
automatically when you send the invite off to the next
Isn't
$var(my_uri) = 5...@example.com;
$ru = sip: + $var(my_ruri);
the same as
rewriteuri(sip:5...@example.com);
just that using $ru you can use it just like assigning to other vars
like you are doing.
On Sun, Feb 6, 2011 at 6:36 PM, Duane Larson duane.lar...@gmail.com wrote:
Avpops
Tyler,
Just went through the OpenSIPS default script webminar =
http://www.opensips.org/html/docs/video/webinar005/
And while the audio at the beginning is bad (and very end), it is only
just a little bit and it is because it was coming through a bad
connection to the seminar where the webinar
-Andrei Iancu bog...@opensips.org wrote:
Hi Dave,
do : if (src_ip==myself) {}
Regards,
Bogdan
Dave Singer wrote:
Is there any way to check if the source IP/port is one that opensips
is listening on or one ? something like if (sip:$si:$sp == myself) {
...bla; bla;}
Thanks
Dave
The best place to start is http://www.opensips.org/
In the left column of the web page there is a section titled Resources
with links to many very helpful resources. Your using the mailing list
so you probably already have seen them to get here.
So. Where are you getting stuck? We need specifics
Toyima,
I posted in the list last week on the thread multiple
use_media_proxy() calls my notes on getting media proxy installed on
centos 5.5.
To compile opensips just do the yum installs mentioned in that thread,
get the opensips source, unpack, compile and install according to docs
on install
Is there any way to check if the source IP/port is one that opensips
is listening on or one ? something like if (sip:$si:$sp == myself) {
...bla; bla;}
Thanks
Dave
___
Users mailing list
Users@lists.opensips.org
Toyima,
Asterisk is a pain for making conference work because of its
dependence on the dahdi kernel driver for timing.
I like the FreeSwitch implementation. FreeSwitch has a very active irc
at irc.freenode.net.
Dave
On Thu, Jan 27, 2011 at 12:21 AM, Toyima Dias toyim...@gmail.com wrote:
Anca,
Thanks for that clarification!
So just to be sure I'm clear on this. $Ts rounds down ( truncates ) the
current second. So using $avp(s:start_time) = $Ts.$Tsm; would give something
like 12343253.543233 and always be accurate?
Further (standard ACC [without dialog]) you could just put $Tsm in the
Be careful when using $Tsm. I read a thread the other day that was just
talking about it and that it returns milliseconds since midnight, not epoch.
So if that is true you will need to somehow handle calls that cross midnight
for duration at least and start/answer/end of call if including the
Adjust STARTOPTIONS to include -p /var/run/opensips.pid ( replace with the
path to where you want the pid file )
There are other options you may want to adjust for starting opensips. Note
that some options are overridden if set in the config file like listen
address. Here is the list of options
Banged my head for a while with why I couldn't get fifo working for the
Control Panel
I was getting sorry -- cannot open write fifo.
Hope this can help other and maybe even make it into the docs.
Found two problems.
1. Apache process couldn't use /tmp/opensips_fifo because the permissions
were
It seems VAD would also be a potential problem with NAT and state
full packet inspection on firewalls. It would see a better practice for the
UA to send keep alive RTP packets. That would solve both dead call detection
and potential NAT issues. That would be nice RFC change.
In my situation I have
is ending. What
about other cases?
On Fri, 19 Nov 2010 09:27:38 -0800
Dave Singer dave.sin...@wideideas.com wrote:
You can use a combination of the following dialog profiles (grouping
calls)
functions for this:
get_profile_size(profile,[value],size)
http://www.opensips.org/html/docs/modules
You can use a combination of the following dialog profiles (grouping calls)
functions for this:
get_profile_size(profile,[value],size)http://www.opensips.org/html/docs/modules/1.6.x/dialog.html#id294302
Check
if a prifile is at limit.
Learning opensips is not like learning a programming language like C, ruby,
puthon, etc. It is very important to get an understanding of how sip flows
and opensips works (get the big picture) then get into the details of how to
manipulate things.
Checkout the past webinars on the Webinars
I ran into the same problem with one of our carriers. The way I did it, with
advice from bogdan, was to set the fr_inv_timer_avp to 6 sec (so long
because of some calls to cell phone systems have long delays) when sending
to the particular carrier then in a reply route special to that carrier,
On my production servers, both 1.6.2 and 1.6.3 versions of opensips, almost
every time I restart opensips it creates a core dump. Since I'm giving it
2GB shared mem, it takes a little while to write the core to disk and start
running again.
So a couple questions.
1. Am I giving it more memory than
compressed from 272MB to 11MB and is at this
linkhttp://viper.wideideas.net/mediaproxy.log.gz
.
Thanks for any help.
Dave
On Wed, Oct 20, 2010 at 11:43 PM, Saúl Ibarra Corretgé s...@ag-projects.com
wrote:
Hi,
On 10/20/2010 11:14 PM, Dave Singer wrote:
I have one server where mediaproxy is having
I have one server where mediaproxy is having serious problems.
In the logs I have a lot of:
media-dispatcher[6067]: error: failed to end dialog: 500 Operation failed
and a lot of of the following but much less than the above:
media-dispatcher[6067]: error: Got `remove' command from OpenSIPS
Julien,
I have been catching it in failure_route and sending it on up with this:
if (t_check_status(^503$)) {
t_reply(503, Service Unavailable);
exit;
}
I think you could use t_reply(503, $(replyrr)); (note the use of
reply to indicate the reply
Setting setenforce 0 is only active for the current running session of the
server. The problem will be back after the reboot.
While that is fine to temporarily do that to see if SELinux is the thing
blocking, it is generally very bad to use that to solve the problem and you
definitely wouldn't
I ran into out of memory and adjusted the shared memory opensips uses with
the -m option.
It tells opensips how much shared memory to use in MB. eg:
.../sbin/opensips -m 500 -l eth0
gives it 500MB of shared memory and listen on eth0
I believe by default it uses 100MB if -m is not specified.
Dave
You may not have monit running as a service. Check the current status of
monit:
monit status
It will tell you if it is running.
If it is not start it with:
/etc/init.d/monit start
I primarily use centos and fedora but I think that will work with debian
systems too.
I think that the default
server load.
A simple patch to report the shutdown duration of each module can be
made if you consider it helpful - but I suspect the DB ops.
Regards,
Bogdan
Dave Singer wrote:
Sometimes when I'm do a restart on opensips (using init.d script, with
some customization to handle
If this goes through to everyone on the list just ignore the paragraph below
and look at the meat of the problem below it in the forward.
This is/was my first post so I'm not sure exactly sure how things work. It
has been a few days and I haven't seen it come out among the other emails
from the
http://www.opensips.org/html/docs/modules/1.6.x/avpops.html#id228513 second
example.
I believe the main problem is the : (colon)
I'm also not sure about your var $ruri.
This should work:
avp_db_load($ru/username, s);
On Mon, Aug 23, 2010 at 2:55 AM, Sujeev suppo...@meewadaya.com wrote:
Hello
Sometimes when I'm do a restart on opensips (using init.d script, with some
customization to handle this problem) opensips takes quite a bit of time,
like 30 - 50 seconds to stop. Other times it is very quick. I'm using 1.6.2
and 1.6.3.
The servers are fairly busy, less then 200 calls per sec,
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