Hi Bodgan, Thank you very much. We got it to work. The problem is we have the
country edition loaded instead of the city edition. See you at cluecon.
Thank You
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Sent from the
I am getting unknown when using mmgeoip. I tried both the citylite version
and city version from maxmind. Thank You
version: opensips 1.7.2-notls (x86_64/linux)
if (mmg_lookup(lon:lat:cc:reg,$si,$avp(lat_lon))) {
xlog(GEOIP: $(avp(lat_lon)[0]) $(avp(lat_lon)[1])
I have the same problem with location and address tables before and i
increased the memory pool fixed it. Hope this will help you as well. Take a
look at http://www.opensips.org/Resources/DocsTsMem
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You need mysql-libs.x86_64.
This is what i have in my Centos to install OpenSIPS with MySQL
mysql.x86_64
mysql-devel.x86_64
mysql-libs.x86_64
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I don't see any error before. Will do the debug level 6 when i get to work in
few hour.
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http://opensips-open-sip-server.1449251.n2.nabble.com/ERROR-mediaproxy-tm-request-in-could-not-create-new-dialog-tp6817274p6820435.html
Sent from the OpenSIPS - Users
Do you have any suggestion on what is the easier way to do it or you want me
to attach everything from the debug log? I can't reproduce this problem on
my testing environment and it only happen on production. The debug log and
siptrace fill up very quickly.
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Sorry everyone. I been working on OpenSER and OpenSIPs for 5 years. This is
the first time i experienced so many problem upgrading.
1. Receiving ERROR:mediaproxy:__tm_request_in: could not create new dialog
on Production only not testing environment.
2. No Audio only on inbound to IP Phone
3.
I uploaded lvl 6 debug and SIP trace. I am not sure what is going on and am
very confused too. There is no audio and call drop only on inbound call to
IP Phone from OpenSIPs. What driving me crazy is that it happen on ATA, IP
Phone, and other Dialer but it doesn't happen to Zoiper Dialer (I tried
hi Saul,
thanks for your response. I am not calling create_dialog in the
opensips.cfg. I am using engage_mdeia_proxy() in the opensips.cfg.
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I think I figured out what is the problem.
IP Phone - Opensips - PSTN have no problem
PSTN - OpenSIPS - IP Phone will produce this error
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There got to be something I am missing for 1.7. on top of this error i have
no audio both way.
IP Phone - Opensips - PSTN have no problem
PSTN - OpenSIPS - IP Phone with no audio. The funny part is Linksys IP
Phone, Linksys PAP2, Pangolin Dialer all have the same problem no audio but
using
Thank you very much for all your response. The problem is fixed for me. It
was a configuration issue that someone sending lot of calls to my gateway
with a # at the end of the number caused a loop and used up all of my
memory. I sent a address incomplete now if none of the URI match. Thank You
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All of the sudden yesterday and today my OpenSIPS kept crashing every every
few hours before of out of mem messages. I tried to increase the PKG
4*1024*1024 and SHM to 256 and recompile and still having the same problem.
There is no core dump generated for the crash and there is out of memory for
Now I have the time to look into the log file more. The the memlog is very
big. What specific should I look into it?
WARNING:core:fm_malloc: Not enough free memory, will atempt defragmenation
ERROR:tm:new_t: out of mem
ERROR:tm:t_newtran: new_t failed
ERROR:tm:store_reply: failed to alloc'
I am using opensips 1.6.3 with very only few modules loaded. The OpenSIPS is
up for over a months with up any issue until now. I also tried to increase
the PKG to 4mb and SHM to 256 mb still crash.
loadmodule db_mysql.so
loadmodule sl.so
I even increased the SHM to 1024 still haven't the same problem. I believe
identified the problem now and will share with everyone once I confirm this
is the fix. Thank You
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Thank you very much for your response. I will look into the re-invite problem
now.
There is no BYE in the SIPTrace from the SIPTrace module associated to this.
The first Invite is received at 17:15:19 and the last ACK is at 17:16:05
from SIPTrace module. The dialog module send BYE at 17:19:05
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I apologized that I wasn't clear enough. The call is established but
terminated after some time. I can provide the SIP trace later today. Thank
You
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The call is established but terminated after some time.
Here is the SIP trace from siptrace module and debug 5 from Opensips. There
is no BYE in the SIPTrace. Debug 5 from Opensips did show BYE sent to caller
and to callee from dialog module.
INVITE sip:1510495x...@74.x.x.x. SIP/2.0
I am not a expert on this but would like to get some understand what is the
problem with my configuration.
I am getting call drop caused by the STT module on 1.6.3 but not on 1.6.2
every 180 seconds if I set modparam(sst, min_se, 180).
This is my configuration
# - Dialog params -
Thanks for your recommendation it work great :).
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Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
I have the same problem that I still haven't yet figured out. Media go both
direction but media relay unable to detect the RTP and conntrack time out
and disconnect the call for me. This only happen to me if use a http Tunnel
server.
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Using the permission module to check the source address and username first
before www_authorize should work.
http://www.opensips.org/html/docs/modules/1.6.x/permissions#id233458
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There is two way audio from UA to OpenSIPs to Asterisk, but there is no Audio
from Asterisk to OpenSIPs to UA after upgrading to OpenSIPS 1.6.2 with
Mediaproxy 2 from OpenSIPS 1.3 with Mediaproxy 1. Can someone please help me
out? I tried everything I can think of. I can see the Media session
I also tried to upgrade from 1.6.2 to 1.6.3 same problem.
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Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
Try to use max_contacts
http://www.opensips.org/html/docs/modules/1.6.x/registrar.html#id228388. Or
use save f flag
http://www.opensips.org/html/docs/modules/1.6.x/registrar.html#id228388
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I spend two day on this finally figured out the problem. I will post up more
detail later. Thank You
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Sent from the OpenSIPS -
Check the radiusclient.conf to make sure that your dictionary is mapped to
the right path. It be a good idea to check your radius log as well.
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I am using OpenSIPS 1.6.2 and followed the tutorial
http://www.opensips.org/Resources/DocsTutLoadbalancing to use the load
balancer module.
The Tutorial use $retcode0 for Service Full reply. I get $rectcode = 1
instead of 0. What is the correct retcode load_balance(id,resource) when
resource is
I figured it out.
This work all the time
if ( uri=~sip:92[1-9][0-...@.* ) {
load_balance(27,white);
} else if ( uri=~sip:3392[1-9][0-...@.* ) {
load_balance(27,grey); #
}
if ( $retcode 0 ) {
sl_send_reply(500,Service full);
exit;
}
This work sometime
if ( uri=~sip:92[1-9][0-...@.* ) {
I find the problem after sniffing the packet today. The problem is that the
somehow the tunnel server sent the RTP to different port and that Mediaproxy
2 doesn't support Asymmetric client anymore.
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k1028 wrote:
I find the problem after sniffing the packet today. The problem is that
the somehow the tunnel server sent the RTP to different port and that
Mediaproxy 2 doesn't support Asymmetric client anymore.
Please ignore my previous response. I triple looked into the sniffing and I
I upgraded OpenSIPS 1.3 with Mediaprxoy 1 to OpenSIPS 1.6 with Mediaproxy 2.
ATA-OpenSIPS is working well. ATA-Tunnel Server-OpenSIPS get conntrack
timeout via engage_media_proxy and use_media_proxy even call is connected
with 2 way audio. I spend two days looking into this and still can't figure
This happen to me before when i have a space at the end of the ip address
inserted into the database.
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Sent from the OpenSIPS - Users mailing list archive at
I discovered that my openser givign me a ERROR:mysql:db_mysql_submit_query:
driver error: there is no ''@'192.168.x.x' registered. This happen with
multiple servers with the same error message and source IP. Some reason the
db_mysql sending no user name and wrong source IP ''@'192.168.x.x' to
.
Bogdan-Andrei Iancu wrote:
Hi,
I think there is a error in your scriptthe $retcode returns the
return code of the last used function, but your LB function is much,
much above the retcode testing
Regards,
Bogdan
k1028 wrote:
I am playing with the Load_balancer module
You need to use db_extra in order to capture the username and callednumber to
log extra value that are not default. You also need to add the field in to
the database.
Look in acc module db_extra and Pseudo Variables
http://www.opensips.org/Resources/DocsCoreVar15#varpv to log extra
variables
I got it to work using
if ( !load_balance(40,pstn) {
sl_send_reply(500,Service FUll);
xlog(L_INFO,Service Full);
exit;
}
instead of
load_balance(40,pstn) {
if ($retcode0 ) {
sl_send_reply(500,Service full);
exit;
}
k1028 wrote:
I tried everything possible
I am playing with the Load_balancer module at this time. The retcode does not
return a negative value for me instead it return 18446744073709551614 when
it reach the pstn limit
I tried with pstn=1 and pstn=2 using 1 peer and 2 peer. All come back with
the same retcode.
I also tried my route
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