Thanks, indeed that was it. So by default it is not enabled, Default value is
"0". and i did not realize this. But the error message is a bit odd, maybe it
should be removed?
Now it work, great. :)
BR
Max Mühlbronner
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Straße der Pariser Kom
Hi,
immediately after starting captagent, which transmits HEP to an opensips 2.3
configured as a HEP switch based on the hep switch tutorials, i receive tons of
messages:
Oct 17 17:03:43 XXX /usr/sbin/opensips[815]: ERROR:proto_hep:hep_udp_read_req:
failed to run hep callbacks
This
Google Voice. At least i can remember a blog post where someone figured that
out while it was in the beta phase...
Max Mühlbronner
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Web
ge each time.
I think it absolutely can be OpenSIPS server causing the TURN error because
OpenSIPS is in charge of creating the sessions in the first place, and if there
are hundreds of OpenSIPS processes running then there will be hundreds of
sessions created for each call.
On 10 August 2017 a
Hi,
I don't see hundreds or even thousands on the screenshot?
Also the Opensips server should not have any connection/relation to your TURN
Server, so i don't think Opensips could be the issue.
A quick google search for "TURN 437 Allocation Mismatch" suggests that it's a
TURN CLIENT <-->
Thanks for the hint. I will keep an eye on it.
BR
Max M.
Von: Bogdan-Andrei Iancu <bog...@opensips.org>
Gesendet: Montag, 3. Juli 2017 12:49:21
An: Max Mühlbronner; OpenSIPS users mailling list
Betreff: Re: AW: [OpenSIPS-Users] drouting (opensips
ag, 3. Juli 2017 12:18:42
An: OpenSIPS users mailling list; Max Mühlbronner
Betreff: Re: [OpenSIPS-Users] drouting (opensips 1.11.x) - maximum number of
gateways?
Hi Max,
Yes, a transaction cannot have more than 12 branches used. But this does not
limit how many GWs you can put in Dynamic Routi
I think i got it, default limit is 12 branches in config.h. Which corresponds
to my limit of 12 gateways, I will try and report back. :)
#define MAX_BRANCHES12 /*!< maximum number of branches
per transaction */
Max Mühlbronner
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Hi,
I've never noticed this until i came across it recently. I got a weird issue
with drouting, it turned out that even though the gatewaylist ("carrier")
contains a total of 20 gateways, only 12 are being used. (all gateways got the
same weight)
E.g. if all gws are rejecting the calls, it
Cool, thanks.
BR
Max Mühlbronner
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r
if (!rl_check("$avp(carriername)", "500", "RED")) {
Thanks.
BR
Max Mühlbronner
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ling list
Users@lists.opensips.org
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Handelsregister/Commercial regis
quot;)) ) || !goes_to_gw("1") ) {
Do failover if 444 reply or if 408 without any reply received
(internal 408).
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 01/20/2017 01:21 PM, Max Mühlbronner wrote:
Hi,
my scenario is a special setu
!goes_to_gw("1") ) {
t_on_failure("2");
t_relay();
exit;
}
if(!t_check_status("\d")) {
t_on_failure("2");
t_relay();
Hi,
@Miha: Are you sure that it does not automatically set the rtpproxies
for 200OK & ACK?
@Sasmita: According to the documentation it is not necessary to invoke
engage_rtp_proxy() for replies as this is handled by the dialog module.
"Function must only be called for the initial INVITE
, Apr 18, 2016 at 11:07 AM, Max Mühlbronner <m...@42com.com> wrote:
I just found this bug which turned into a feature request (from 2012)
someone else had exactly the same problem:
https://sourceforge.net/p/opensips/feature-requests/99/
@Bogdan, if for whatever reason the table is being
like mine and i would guess there are a lot of people with the
same problem: but they probably never noticed it, or never will.
Best Regards
Max M.
On 07.04.2016 13:00, Max Mühlbronner wrote:
Hi,
I experienced something weird: I got two servers sharing the same
location table. (usrloc module
own socket/IP.
Is there any workaround for my situation? (Nat ping does not take the
socket/IP of the registered client into account?)
Best Regards
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E-Mail: m...@42com.com
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Fir
will check and fix asap.
Thanks and regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 17.02.2016 12:53, Max Mühlbronner wrote:
http://opensips.org/pub/opensips/1.11.6/opensips-1.11.6.tar.gz
It seems there is no 1.11.6-tls tarball source available
http://opensips.org/pub/opensips/1.11.6/opensips-1.11.6.tar.gz
It seems there is no 1.11.6-tls tarball source available. (This one is
"-notls", and there is no "tls" subdirectory included...)
I know i could fetch it from the repository, but i was just wondering
why the tarball does not
Hi,
if you want (non-system) logfiles to be rotated you should use
"logrotate" and you should create a configuration for opensips logfile. :)
http://opensips.org/pipermail/users/2010-December/015826.html
http://opensips.org/pipermail/users/2009-March/003774.html
Best Regards
Max M.
Hi,
not sure, but this might help:
http://www.opensips.org/html/docs/modules/1.11.x/auth_db.html#id293636
modparam("auth_db", "skip_version_check", 1)
Although in the long term, it's probably better to upgrade mysql.
BR
Max M.
On 28.12.2015 12:04, Husnain Taseer wrote:
Dear Users,
We
Sorry, skip_version_check() of auth_db module seems to be related to the
auth table (not the general mysql auth).
my fault.
BR
Max M.
On 28.12.2015 12:39, Max Mühlbronner wrote:
Hi,
not sure, but this might help:
http://www.opensips.org/html/docs/modules/1.11.x/auth_db.html#id293636
Hi,
seems to be a simple solution, without overhead/database/...
if(is_present_hf("P-Source-IP")){
$DLG_timeout = $(hdr(P-Source-IP));
}else{
$DLG_timeout = 3600;
}
But you could also save the information into a e.g. mysql/... database
and pull it from the db. (check out
Sorry, copy mistake.
if(is_present_hf("X-Timeout")){
$DLG_timeout = $(hdr(P-Source-IP));
On 28.09.2015 17:40, Max Mühlbronner wrote:
Hi,
seems to be a simple solution, without overhead/database/...
if(is_present_hf("P-Source-IP")){
$DLG_timeout
Hi,
without any additional modules, something like this works out of the box:
if ($rU==username_of_forbidden_phone) {
t_reply(486, Busy Here);
exit;
}
Just add this in the correct route block in your config, where the call
is going to user location / registered clients and
in Amsterdam for the OpenSIPS Summit !
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 07.05.2015 11:01, Max Mühlbronner wrote:
Comparing opensipsctl ps and the pid of the remaining process shows
it's the attendant process.
root@opensips1
-Andrei Iancu wrote:
Yes indeed, it looks like flushing dialog info into DB. How many
dialog do you have ongoing and how fast your db is ??
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 08.05.2015 13:05, Max Mühlbronner wrote:
Here
like this before?
BR
Max M.
On 05.05.2015 11:00, Max Mühlbronner wrote:
Hi,
when restarting Opensips it will shut down all the processes
immediately, but there is still a single process left hanging (cpu
load) which eventually exits after some time. It is not possible to
restart Opensips
then fails to start. Try adding - - retry option
to start-stop-daemon.
On May 7, 2015 4:01 AM, Max Mühlbronner m...@42com.com
mailto:m...@42com.com wrote:
Comparing opensipsctl ps and the pid of the remaining process
shows it's the attendant process.
root@opensips1:/etc/opensips
Hi,
when restarting Opensips it will shut down all the processes
immediately, but there is still a single process left hanging (cpu
load) which eventually exits after some time. It is not possible to
restart Opensips until this process is killed/quits.
root@opensips1:~# /etc/init.d/opensips
Hi,
i just ran into the same issue, and noticed the tarball
http://opensips.org/pub/opensips/1.11.4/src/opensips-1.11.4_src.tar.gz
(generated on 02. April 2015) is not fixed yet. Wanted to let you know,
as this might lead to problems for new users trying to install Opensips
1.11.x from
Hi,
I just noticed nortpproxy_str is listed in the documentation for both
Rtpproxy Nathelper.
At first i tried setting it to (If empty string, no marker will be
added or checked.) for the nathelper module but it does not work/does
nothing.
But then i noticed: it works perfectly in the
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
when using force_tcp_alias(), after already setting
tcp_persistent_flag i am receiving errors:
/sbin/opensips[17128]: ERROR:core:tcpconn_add_alias: possible port
hijack attempt
/sbin/opensips[17128]: ERROR:core:tcpconn_add_alias: alias already
Hi,
nice, very interesting.
The link to the documentation is not working yet?
(http://www.opensips.org/html/docs/modules/1.10.x/sngtc.html)
Best Regards
Max M.
On 08/05/2013 04:55 PM, Liviu Chircu wrote:
Hello all,
The next OpenSIPS release has been improved with an
Hi,
SIP session-timers ( http://tools.ietf.org/html/rfc4028), which are
implemented via sst module in opensips.
Another commonly used method is to enable rtp timeout on the media
gateways, which does not depend on signaling but basically detects if
one leg does not send RTP anymore and will
Google:
inurl:cmterm-7941_7961-sip.8-5-4.zip
But its a russian site, not sure if this is legit?
Best Regards
On 03/18/2013 11:37 PM, Adam Baird wrote:
Hi all.
I have been tasked with performing a SIP interop with the Cisco 7941
model IP phone. I've failed to get it working with the
Sorry, wrong list :)
On 03/19/2013 12:24 PM, Max Mühlbronner wrote:
Google:
inurl:cmterm-7941_7961-sip.8-5-4.zip
But its a russian site, not sure if this is legit?
Best Regards
On 03/18/2013 11:37 PM, Adam Baird wrote:
Hi all.
I have been tasked with performing a SIP interop
just before creating the
transaction (by calling t_newtrans or t_relay or any t_* function that
creates transaction).
Thank you.
On Thu, Mar 14, 2013 at 5:15 PM, Max Mühlbronner m...@42com.com
mailto:m...@42com.com wrote:
Hi,
I am not sure about record-routing in combination
Regards
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10243 Berlin
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Hi,
drouting will choose the rule/entry based on the longest matching
prefix, which in your case is id2. But additionally you can assign a
different priority to each of the rules.(change prio webinterface/DB)
If prefix /3 has a priority of 2 and prefix 36 only has a priority/
of 1, even if
Developer
http://www.opensips-solutions.com
On 02/22/2013 11:17 AM, Max Mühlbronner wrote:
Hi,
Sorry to bother you directly, but it seems you were involed in this
problem.
http://lists.opensips.org/pipermail/users/2011-January/016473.html
Bug was closed:
http://sourceforge.net/tracker
Hi,
I just looked into this because i once had similar problems and this
caught my interest...
It seems like you have to change:
$this-serialize($prefix.['.preg_replace(/([\\\'])/, 1, $k).'],
$str);
to
$this-serialize($prefix.['.preg_replace(/([\\\'])/, 1,
$k).'],
Hi,
If the second server is started on-demand (e.g. keepalived) the dialogs
are loaded into memory from the DB (where the other opensips stored the
dialogs , by db_mode realtime..).
So there would be no need to use dlg_db_sync in this simple failover
scenario, right?
Best regards
Max
Hi,
maybe you need to adjust memory_limit in php.inf and restart apache. At
least this helped me a few times when i had the same problem with
dynamic routing/dr_rules table.
Max M.
On 11/15/2012 11:23 AM, Miguel J. López Valverde wrote:
Dear Opensips lists:
I've a trouble with the
Hi,
debug=3
log_stderror=yes
fork=no
This will write the messages straight to your console, so you can easily
spot any errors, fix restart until everything is fine. :)
Best Regards
Max M.
On 10/29/2012 02:56 PM, Christian Cambier wrote:
Hi.
I'm having problems getting started.
I
://lists.opensips.org/cgi-bin/mailman/listinfo/users
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10243 Berlin
E-Mail
Hi,
regarding asterisk as media-server, you could use the noanswer option for
playback(). Then it will signal audio via progress messages but will not
answer (200 OK) the call.
Best Regards
Max M.
Von: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] Im
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Firmenangaben/Company
Check which port rtpproxy is running on, to see if the port is reachable.
netstat -anp|grep rtpproxy
No available proxies, means opensips it not able to connect to your
Rtpproxy control ports or unix socket.
Best Regards
Max M.
On 09/27/2012 04:31 PM, Binan AL Halabi wrote:
hej spady,
is modified or otherwise when a gateway
goes down.
Regards,
MOUTOT Alexandre
a.mou...@alphalink.fr
+33 (0)6 62 91 95 14
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42com
Hi,
pri_prefix is related to dr_gateways table, it is like a techprefix
used by the carrier/gw. It is not matched, but added to the request uri
when sending out the call to the Gateway.
Did you mean prefix of dr_rules table? These prefixes are used to
match based on the request uri, but not
http://www.opensips.org/Resources/DocsTutLoadbalancing
The tutorial contains everything you need, documentation and example
opensips.cfg. There should be no big difference between 1.8 and older
versions.
Best Regards
Max M.
On 08/27/2012 02:04 PM, Engineer voip wrote:
Hello,
I have 2 GW
to get REGISTER instead of 128. How can I do that ?
Thanks for your help.
Regards,
Sebastien
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Straße
Hi,
i just noticed some strange problems while trying to use rtpproxy sets.
Opensips version 1.6.4-2.
if(is_method(INVITE) !has_totag()) {
switch ($Ri) {
case X.X.X.X:
$avp(s:rtpsets)=1;
I will open a bug report, thanks.
BR
Max M.
On 03/29/2012 05:39 PM, Bogdan-Andrei Iancu wrote:
Hi Max,
It sounds strange, especially that you are not doing something wrong -
better open a bug report on this.
Regards,
Bogdan
On 03/29/2012 04:54 PM, Max Mühlbronner wrote:
Hi,
i just
I was worried about the same thing, until i noticed the critical error
just means a client sends a bye (event 7) for a dialog in
progress.(state 2) when he should send a CANCEL instead.
So there is nothing to worry about, right!?
Best Regards
Max M.
On 03/14/2012 01:20 PM, Bogdan-Andrei
Hi,
20% might be true for general network usage, but not for Voice traffic.
Also it depends on the codecs used, 10% packetloss at VoIP means quality
will be degraded (codec could only compensate up to 5%).
http://www.voiptroubleshooter.com/problems/packetloss.html
g711 at 10% packetloss:
Hi,
the response from the Cisco suggests it has a problem with the traffic:
SIP/2.0 400 Bad Request - 'Invalid IP Address'.
You could try rtpproxy/mediaproxy, it seems like the Cisco GW is not able to
route to the network of the SDP IP (c=IN IP4 12.34.56.78.)?
BR
Max M.
-Ursprüngliche
If you are trying to use ICE you should indeed have a stun/turn server
in your settings (which also should be working!). Does opensips.org SIP
Service support ICE?
Best Regards
Max M.
Am 09.01.2012 17:01, schrieb Bo Shi:
Hi,
Thanks for reply!
Enclosed the log from pjsip.
I hide
I see, the problem seems to be your opensips.cfg. You should probably try
with the example opensips.cfg provided for the section of the book?
Step 3: Change the script to avoid authentication and loose routing for sipp
packets
(use the 0745_11_02.cfg script provided in the code bundle). In the
If i remember correctly building telephony systems with opensips (flavios
great book) suggests to disable record routing in opensips for testing with
sipp.
Best Regards
Max M.
-Ursprüngliche Nachricht-
Von: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] Im
Hi,
if i understand this correctly, this should be easy to solve if all your
domains point to one ip. The www_challenge function has a parameter
realm if you set this to your ip (where all your domains / subdomains
point to) and also add this IP as domain for every user in subscriber
table
Hello,
Thanks! Now i can also use opensips instead of kamailio as capture node
for Homer.
A great tool which makes life (work) a lot easier. Faster and way more
efficient than wireshark/ngrep or anything else i have used so far :)
Best Regards
Max M.
Am 14.12.2011 14:55, schrieb
Hi,
check out the LB documentation ( e.g.
http://www.opensips.org/html/docs/modules/1.6.x/load_balancer.html ),
there is a parameter for the load_balance function (algorithm) which
sets the load balancing to relative or absolute. As far as i know this
is the only way to manipulate the way
Hello,
a solution would be checking for the sourceip and using
force_send_socket() to set a different interface, which will be used by
t_relay.
if ($si=~^108\.109\.180\. || $si=~^10\.10\.10\. {
force_send_socket(udp:108.109.180.12:5060);
}
t_relay();
Best Regards
Max M.
Am
experiences with this new parameter?
Best Regards
Max M.
Am 03.10.2011 11:29, schrieb Max Mühlbronner:
Hello,
I have migrated one Opensips instance from version 1.6.4 to 1.7.0
(Database/Config/..) everything was working fine but after running it
for 24 hours under the same load (~2000 dialogs) I
the latest revision from the 1.7 svn branch ?
If not, I would advise to update, there have been some issues in the
RTPProxy module that lead to 100% CPU use.
Regards,
Vlad Paiu
OpenSIPS Developer
On 10/04/2011 11:01 AM, Max Mühlbronner wrote:
I found something, which sounds interesting:
/1.3.20
Hello,
I have migrated one Opensips instance from version 1.6.4 to 1.7.0
(Database/Config/..) everything was working fine but after running it for 24
hours under the same load (~2000 dialogs) I could see spikes of load caused
by opensips children processes.
This just goes for like 1
Hi,
what does too many mean? :)
Please explain a bit more, the e-mail before said removing un-wanted
Options packets which can be easily done, just check for options --
drop it. But why would you want to do this?
BR
Max M.
Am 29.09.2011 16:56, schrieb nguyen khue:
Hi Faisal,
Please
Hi,
could you post a trace, ngrep/wireshark? But the logs already show,
there is something wrong with the SIP Headers (bad via). It looks like
there are some characters missing. e.g. ntent should be Content ?
Best Regards
Max M.
Am 03.08.2011 16:27, schrieb Akib Sayyed:
Hi,
The actual forwarding is quite simple.
if(!is_method(NOTIFY)) {
rewritehostport(1.2.3.4:5060); --- IP/port of your IP-PBX
t_relay();
}
Best Regards
Max M.
Am 27.07.2011 11:53, schrieb spady:
Hi all, I am pretty new on OpenSIPS world so first of all sorry for dummy
Hi,
sorry typo:
if(is_method(NOTIFY)) {
BR
Max M.
Am 28.07.2011 12:36, schrieb Max Mühlbronner:
Hi,
The actual forwarding is quite simple.
if(!is_method(NOTIFY)) {
rewritehostport(1.2.3.4:5060); --- IP/port of your IP-PBX
t_relay();
}
Best Regards
Max M.
Am
Hi,
\r\n is just the line break (new line), like br in html.
difference between both functions is the location where the header will
be added.
append_hf(txt) - Appends 'txt' as header*after the last* header field.
append_hf(txt, hdr) - Appends 'txt' as header*after first
Hi,
quick guess, dont know why, but could it be related to insert_hf?
I always used append_hf which adds the header after the last header
field. I never tried insert_hf, append_hf worked fine for me.
BR
Max M.
Am 11.07.2011 15:47, schrieb n...@uni-petrol.com:
Forgot to add, that problem
:13 PM, Max Mühlbronner wrote:
Hi,
yes, i tried several times. I should have mentioned the dr_rules has
quite a few (about 100k entries). I did notice it on a production
server running 1.6.2 but also was able to replicate the same behavior
when testing with the same Database/dr_rules on 1.6.4-tls
Hi,
i just came across another weird issue. There are two Gateways in
routing, both got the same IP, but two different Gateway types as
defined in opensips-cp. (type 7 / type 8)
I got some checks in my script to check for the gw Type and do some
action depending on the type of Gw used in
Hello,
Opensips seems to not route my requests while reloading the drouting
rules from Database. Probably the DB operations are blocking the
remaining operations?
Any idea if this is normal behavior, or misconfiguration on my side?
Does anyone know a solution for reloading while still
Hello,
the avpops is missing the db connection / db_url modparam.
http://www.opensips.org/html/docs/modules/devel/avpops.html#id249134
modparam(avpops,db_url,mysql://user:passwd@host/database)
Best Regards
Max M.
Am 14.06.2011 17:02, schrieb Tiberiu Breana:
Hello.
I want to use some
to
configure one if I want to use avp_db_query?
Thanks.
On 15 June 2011 13:35, Max Mühlbronner m...@42com.com
mailto:m...@42com.com wrote:
Hello,
the avpops is missing the db connection / db_url modparam.
http://www.opensips.org/html/docs/modules/devel/avpops.html#id249134
Hello,
Logan did you have any luck? I am also looking into this because we had
some issues with vst/vsf parameters and maybe this could be a solution. :)
Is it okay to set force_dialog and also execute create_dialog?
BR
Max M.
Am 17.05.2011 19:35, schrieb Logan:
I don't think so
Hi,
The sems example uses variables which are setup at serweb.
(email/language/...)
These should be set in opensips config (maybe taken from database/
whatever...)
There are variables in opensips containing some of the values you could
use like the request domain / from uri / ...
Hi,
i would suggest doing sip-traces on asterisk (sip debug) and opensips
(ngrep) while watching the corresponding log messages of both servers
(asterisk/opensips). Most of the time it´s difficult to find a problem
by looking at it from just one side.
BR
Max M.
Am 06.05.2011 05:53,
Hi,
You just need to raise the debug level (debug=4) and you will receive
lots of debugging messages in your logs.
debug= in config, or set the debug level on the fly via fifo mi commands:
opensipsctl fifo debug
opensipsctl fifo debug 4
Best Regards
Max M.
Am 04.05.2011 11:35,
Hi,
yeah, sounds like the right solution.
Also stumbled upon this problem sometime ago, and kept wondering what
was going on. Now it is very clear to me.
BR
Max M.
Am 14.04.2011 13:02, schrieb Bogdan-Andrei Iancu:
Hi all,
Following some discussions on the value of db_url, we end up with
Hi,
it´s simple, if you are runnin both opensips and opensips-cp on the same
server you need to configure: /config/boxes.global.inc.php
$boxes[$box_id]['mi']['conn']=/tmp/opensips_fifo;
change it to:
$boxes[$box_id]['mi']['conn']=127.0.0.1:8000;
but if they are running on the same server,
Uh, typo...
change $boxes[$box_id]['mi']['conn']=127.0.0.1:8000; to
$boxes[$box_id]['mi']['conn']=/tmp/opensips_fifo;
Sorry :)
Max M.
Am 10.03.2011 15:58, schrieb Max Mühlbronner:
Hi,
it´s simple, if you are runnin both opensips and opensips-cp on the
same server you need
the change, now i'm getting:
Array ( [0] = sorry -- cannot open write fifo [1] = sorry --
cannot open write fifo )
On Thu, Mar 10, 2011 at 9:59 AM, Max Mühlbronner m...@42com.com
mailto:m...@42com.com wrote:
Uh, typo...
change $boxes[$box_id]['mi']['conn']=127.0.0.1:8000
http
Hello,
i want to perform a check on the VIA Headers, basically to compare if
the source ip ($si) is included in one of the VIA Headers. Already tried
several things, but it seems like there is no way to check for a
variable in a regexpression, maybe someone got a solution or tried
something
to use the dr_rules table (for the mapping
of users to groups), I would manually do the query (with
avp_db_query() ) from the script (to get the group id) and only for
certain values of the group ID I will do do_routing(group_id)
Regards,
Bogdan
Max Mühlbronner wrote:
Hi,
sorry maybe i did
,
Bogdan
Max Mühlbronner wrote:
Hello,
regarding opensips-cp and drouting i came across a small problem,
maybe someone already tried something similar and wants to share his
knowledge :)
|
opensips-cp -- Drouting / Settings, Gateway Types / Group ID´s is
what i am talking about
Hello,
regarding opensips-cp and drouting i came across a small problem, maybe
someone already tried something similar and wants to share his knowledge :)
|
opensips-cp -- Drouting / Settings, Gateway Types / Group ID?s is what
i am talking about.
|
Is there any function to check for the
Hi,
you could use the pseudo variable $ml (*$ml* - reference to SIP message
length ). or if needed: $cl (content-length header)
Hope this helps.
Max M.
Am 19.10.2010 15:48, schrieb Dmitry Kravchenko:
Hi!
Those values - uri and myself can only be used for comparisons/tests,
you can
Hi,
Opensips offers a modular setup, where the core provides just SIP
routing/registrar functions and you can add additional features by
loading modules for the purpose you mentioned. e.g. using b2bua module
for b2bua functionality.
Mediaproxy/mediaserver/Class5 PBX features would need
of retries before timeout
So, to avoid too much waiting, tthe rtpproxy_timeout and
rtpproxy_retr should have small values - this will avoid blocking.
Regards,
Bogdan
Max Mühlbronner wrote:
Hello everyone,
i have experienced some strange Problem using multiple instances of
rtpproxy via
Hello everyone,
i have experienced some strange Problem using multiple instances of
rtpproxy via rtpproxy_sock. (Opensips 1.6.2) If one rtpproxy is
disabled, opensips will still try to re-enable/reconnect to the same
rtpproxy later. The problem is, opensips is not responding correctly at
Hello,
@Inaki:
it sounds to me like he does not route through both, but only one (maybe
one of the servers is some failover node/Backup?)
And if he is registering on one server alone, it is working fine but not
on the other one. If this is the case, then i would agree there should
not be a
? we are planning a new release
on 1.6 branch for next week and I really want to have this fixed.
Regards,
Bogdan
Max Mühlbronner wrote:
Hello Bogdan,
too bad, but the problem continues after update.
opensips-dev:/tmp/opensips# gdb /sbin/opensips core.opensips.sig11.9272
GNU gdb 6.8
if i still got access to it.
Thanks
Max M.
Bogdan-Andrei Iancu schrieb:
Hi Max,
What you mean by a missing SDP in the progress ? you mean a 183
without SDP ? so a force_rtp_proxy on something without SDP may lead in
crash?
Regards,
Bogdan
Max Mühlbronner wrote:
Hello,
Very
there with this version.
Thanks and regards,
Bogdan
Max Mühlbronner wrote:
Hi,
Thanks for the hint. But I really dont understand how this happened
because i thought i did initially check out the 1.6 branch via svn!?
(and not 1.5) But maybe i mixed up something..
/usr/src/OPENSIPS-SVN/opensips_1_6
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