[OpenSIPS-Users] [SERVICE] OpenSIPS Alerts/Notifications Service

2012-06-07 Thread Bogdan-Andrei Iancu

Hello all,

A new service is available for the OpenSIP project - the '''OpenSIPS 
Alerts Service"



1.  Overview
=

The OpenSIPS Alerts Service is a way to help you keep your OpenSIPS 
installation updated, safer and secure. Via this service you will 
receive realtime email alerts/notifications with a full and detailed 
description of issues and bugs that were found and fixed in OpenSIPS code.


More on http://www.opensips.org/Resources/AlertsMain#toc1


2.  Who and Why should use it
==

The OpenSIPS Alerts Service is addressed mainly to user and operators 
(production systems) of OpenSIPS based platforms - shortly, it is 
addressed to anyone who whats to run and keep an up to date, safe and 
secure OpenSIPS installation
The information provided by the service is critical to operators because 
it is realtime, it is comprehensive and it is brought under his attention.


More on http://www.opensips.org/Resources/AlertsMain#toc2


3.  Examples
=

See http://www.opensips.org/Resources/AlertsMain#toc3


4.  Why is it paid


The OpenSIPS Alerts Service is a paid service. You do not pay for 
getting access to information on bugs and fixes. What you pay for is the 
realtime notification, translation and extensive documentation of the fixes.
The income from this service will be invested back into the project, 
like supporting developers for their work (especially focused on 
OpenSIPS 2.0 development) and covering expenses for running the project 
(server hosting, webinar program, etc).


More on http://www.opensips.org/Resources/AlertsMain#toc4


5.  Subscribe to alerts


To subscriber to the OpenSIPS Alerts Service, see 
http://www.opensips.org/Resources/AlertsSubscribe



I hope you all will find this service useful!



Best regards,

--
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com


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[OpenSIPS-Users] No Voice Comm in Conference call

2012-06-07 Thread Arjun Shankar K S
Hi All, 

Greetings to everyone !!! 

I have set up opensips and RTP Proxy in two different hosts since I have 
opensips in a Natted environment where RTP Proxy refused to budge. 

Now I have installed RTP Proxy in a direct public IP. Normal calls between 2 
Client is working great !! 

During conference call, the calls get connected but there is no voice 
communication between any of them and soon the client who was connected last, 
gets disconnected. 

I could not find much support regarding this issue. Any support is sincerely 
appreciated. 

In my opensips.cfg, I have made the following config changes for RTP in 
different host, 

--Nat Params- 
modparam("usrloc","nat_bflag", 6) 
modparam("nathelper","rtpproxy_sock", "udp:rtp_proxy_publicIP:7890") 
modparam("nathelper","natping_interval", 30) 
modparam("nathelper","ping_nated_only", 0) 
modparam("nathelper","sipping_bflag", 7) 
modparam("nathelper","sipping_from", "sip:pinger@PROXY_IP") 
modparam("registrar","received_avp", "$avp(i:42)") 
modparam("nathelper","received_avp", "$avp(i:42)") 

I am running my RTP Proxy using the following command, 

./rtpproxy -l rtp_proxy_publicIP -s udp:*:7890 -F 


Thanks, 
Arjun 
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[OpenSIPS-Users] drouting do_routing not randomizing when sort=1

2012-06-07 Thread Alejandro Recarey
Hi all,

I am using opensips 1.6.2 and am using drouting module like this:

  do_routing("1","1");

According to the manual this means "group 1" and "randomize members in
each group". My group has the following gateways:

  13,14

When I run opensips, the calls only go to gateway 13. I thought that
by randomizing, sometimes it would take "13,14" and sometimes "14,13".
Am I understanding this wrong?

Alex

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[OpenSIPS-Users] Getting opensips: Unknown Object Identifier (Sub-id not found: (top) -> opensips)

2012-06-07 Thread Ahmed Munir
Hi,

After configuring SNMP + OpenSIPS as described in the document;
http://www.opensips.org/html/docs/modules/devel/snmpstats.html#id250252 and
http://www.kamailio.org/dokuwiki/doku.php/utils:kamailio-and-snmp. When I
try to run the 'snmapwalk localhost openser/opensips' on shell, I'm getting
this message as listed below;

opensips: Unknown Object Identifier (Sub-id not found: (top) -> opensips)

Please assist me at earliest, to resolve this issue.


-- 
Regards,

Ahmed Munir Chohan
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[OpenSIPS-Users] Dialogs not getting removed upon receiving BYE

2012-06-07 Thread Dinusha Kannangara
Hi,

I'm new to opensips, and I'm having trouble with dialogs created from the
load_balancing module not being removed upon receiving the BYE when the
call is terminated.
I'm using opensips version 1.7.2 and have followed this example as well (
http://www.opensips.org/Resources/DocsTutLoadbalancing
).

Currently I'm testing it on two sip phones which are registered on the
opensips SIP Router (at times I'm using one sip phone as both the caller
and callee, rewriting the destination uri (automatically done through the
load_balance() method) as my IP to route the call back to the same sip
phone). Either way the dialog is not destroyed on bye and Wireshark output
shows that I'm receiving the 200 OK to both sides and the call gets cut
without any problem.

So, am I missing something on the config file ? Here's my config file

*opensips.cfg*

### Global Parameters #

debug=6
log_stderror=yes
log_facility=LOG_LOCAL0

fork=yes
children=4

/* uncomment the following lines to enable debugging */

/* uncomment the next line to disable TCP (default on) */
#disable_tcp=yes

/* uncomment the next line to enable the auto temporary blacklisting of
   not available destinations (default disabled) */
#disable_dns_blacklist=no

/* uncomment the next line to enable IPv6 lookup after IPv4 dns
   lookup failures (default disabled) */
#dns_try_ipv6=yes

/* uncomment the next line to disable the auto discovery of local aliases
   based on revers DNS on IPs (default on) */
#auto_aliases=no
/* uncomment the following lines to enable TLS support  (default off) */
#disable_tls = no
#listen = tls:your_IP:5061
#tls_verify_server = 1
#tls_verify_client = 1
#tls_require_client_certificate = 0
#tls_method = TLSv1
#tls_certificate = "//etc/opensips/tls/user/user-cert.pem"
#tls_private_key = "//etc/opensips/tls/user/user-privkey.pem"
#tls_ca_list = "//etc/opensips/tls/user/user-calist.pem"

/* default db_url to be used by modules requiring DB connection;
   uncomment it if you use any module requiring DB connectivity */
#db_default_url="mysql://opensips:opensipsrw@localhost/opensips"


port=5060

/* uncomment and configure the following line if you want opensips to
   bind on a specific interface/port/proto (default bind on all available)
*/
#listen=udp:192.168.1.2:5060




### Modules Section 

#set module path
mpath="//lib64/opensips/modules/"

/* uncomment next line for MySQL DB support */
loadmodule "db_mysql.so"
loadmodule "signaling.so"
loadmodule "sl.so"
loadmodule "tm.so"
loadmodule "rr.so"
loadmodule "maxfwd.so"
loadmodule "usrloc.so"
loadmodule "registrar.so"
loadmodule "textops.so"
loadmodule "mi_fifo.so"
loadmodule "uri.so"
loadmodule "acc.so"
loadmodule "avpops.so"
loadmodule "dialog.so"
loadmodule "load_balancer.so"
/* uncomment next lines for MySQL based authentication support
   NOTE: a DB (like db_mysql) module must be also loaded */
loadmodule "auth.so"
loadmodule "auth_db.so"
/* uncomment next line for aliases support
   NOTE: a DB (like db_mysql) module must be also loaded */
#loadmodule "alias_db.so"
/* uncomment next line for multi-domain support
   NOTE: a DB (like db_mysql) module must be also loaded
   NOTE: be sure and enable multi-domain support in all used modules
 (see "multi-module params" section ) */
#loadmodule "domain.so"
/* uncomment the next two lines for presence server support
   NOTE: a DB (like db_mysql) module must be also loaded */
#loadmodule "presence.so"
#loadmodule "presence_xml.so"


# - setting module-specific parameters ---


# - mi_fifo params -
modparam("mi_fifo", "fifo_name", "/tmp/opensips_fifo")


# - rr params -
# do not append from tag to the RR (no need for this script)
modparam("rr", "append_fromtag", 0)


# - registrar params -
/* uncomment the next line not to allow more than 10 contacts per AOR */
#modparam("registrar", "max_contacts", 10)
# - usrloc params -
modparam("usrloc", "db_mode",   0)
/* uncomment the following lines if you want to enable DB persistency
   for location entries */
#modparam("usrloc", "db_mode",   2)
#modparam("usrloc", "db_url",
#   "mysql://opensips:opensipsrw@localhost/opensips")


# - uri params -
modparam("uri", "use_uri_table", 0)


# - acc params -
/* what sepcial events should be accounted ? */
modparam("acc", "early_media", 1)
modparam("acc", "report_cancels", 1)
/* by default ww do not adjust the direct of the sequential requests.
   if you enable this parameter, be sure the enable "append_fromtag"
   in "rr" module */
modparam("acc", "detect_direction", 0)
/* account triggers (flags) */
modparam("acc", "failed_transaction_flag", 3)
modparam("acc", "log_flag", 1)
modparam("acc", "log_missed_flag", 2)
/* uncomment the following lines to enable DB accounting also */
modparam("acc", "db_flag", 1)
modparam("acc", "db_missed_flag", 2)

#load_balance---#
modparam("load_balancer", "db_url","mysql://root:123@localhost/opensips")