[OpenSIPS-Users] OPENSIPS+ASTERISK Integration

Bogdan-Andrei Iancu bogdan at voice-system.ro
Fri Jun 11 16:41:49 CEST 2010


Hi Prem,

Premalatha Kuppan wrote:
> Hi Bogdan,
>
> Thanks for yor reply.
>
> Can you brief me about the "lookup(location) to send the call to the 
> callee device."
Please check the default opensips cfg that comes with the sources/packages.

Also listen the tutorial on the CFG - 
http://www.opensips.org/Resources/Webinars#toc8

Even more, read the docs related to user location (see module "registrar").
>
> Here, during the IVR the user will sent the callee #, i have to fetch 
> the corresponding IP addr of that callee from the database to route 
> the call. From asterisk, i can do that by connecting to mysql and 
> fetch the IP of that callee from the DB. I wonder how it can be done 
> at opensips.
You put the same question like last time - the answer is the same - if a 
device registered with OpenSIPS, OpenSIPS is the best place to send back 
calls to that device - querying from Asterisk the opensips DB is a dirty 
hack with a lot of potential problems.

Regards,
Bogdan

>
> If you could help me in briefing more on "lookup(location) to send the 
> call to the callee device" from opensips. I appreciate.
>
> Thanks,
> Prem
>
>
> On Fri, Jun 11, 2010 at 2:26 PM, Bogdan-Andrei Iancu 
> <bogdan at voice-system.ro <mailto:bogdan at voice-system.ro>> wrote:
>
>     Hi Prem,
>
>     Once asterisk did the IVR based auth, I suggest that asterisk should
>     route the SIP call back to OpenSIPS - and OpenSIPS will do the
>     lookup(location) to send the call to the callee device.
>
>     In OpenSIPS, based on source Ip, you can have 2 routing logics -
>     if call
>     comes from Asterisk -> outbound routing ; if not from asterisk ->
>     inbound logic.
>
>     Regards,
>     Bogdan
>
>     Premalatha Kuppan wrote:
>     > Yes, i have checked all these options. Everything is in correct
>     status.
>     >
>     > I doubt, connecting between vmware and with 64 bit RHEL. My set is
>     > running on vmware image. When i ping on from vmware where
>     > opensips+asterisk is running to the MySQL server vmware image, i see
>     > this problem. But when i set MySQL server on non-vmware, it works
>     > perferctly fine.
>     >
>     > i 'll update once i figure out the reason.
>     >
>     > Between, to route a call from OPENSIP to the destination. I need to
>     > fetch the destination details IP from MySQL DB.
>     > Meaning,
>     >
>     > User -------->
>     >
>     OPENSIPS(1)--------->ASTERISK(IVR)-------------->OPENSIPS(1)----------------->Destination
>     >
>     > Here, for authentication user enters the PIn and extension to
>     access,
>     > if the extension is present in the DB, Asterisk(IVR) will allow the
>     > user to access it. From this from asterisk, i can make query to
>     DB and
>     > fetch the neccessary details. Likewise is it possible to fetch the
>     > details(IP where the extension is connected and to be routed)
>     from DB
>     > from opensips.cfg to route the call to requested destination.
>     >
>     > Thanks,
>     > Prem
>     >
>     > On Thu, Jun 10, 2010 at 3:01 PM, Bogdan-Andrei Iancu
>     > <bogdan at voice-system.ro <mailto:bogdan at voice-system.ro>
>     <mailto:bogdan at voice-system.ro <mailto:bogdan at voice-system.ro>>>
>     wrote:
>     >
>     >     Are you sure the DB conn params are correct (like address of
>     myql
>     >     server) ?
>     >
>     >     Also are you sure the mysql server is remotely reachable (it
>     >     listens on
>     >     routable IP and there is no firewall in front)
>     >
>     >     Have you tried to connect to mysql by using directly the mysql
>     >     client -
>     >     just to see if it works?
>     >
>     >     Regards,
>     >     Bogdan
>     >
>     >     Premalatha Kuppan wrote:
>     >     > Iam facing one more problem.
>     >     >
>     >     > Iam running my set up on 64bit processor. Opensips and
>     asterisk
>     >     on one
>     >     > PC and Mysql server on seperate PC.
>     >     >
>     >     > Iam running on RHEL enterprise version.
>     >     >
>     >     > Problem is that, its taking more than 2 mins to connect to
>     MySQL
>     >     > server when i start Opensips. Also user is not getting
>     >     registered. of
>     >     > 10 attmept, only 1 is getting successful. But even this is not
>     >     consistent.
>     >     >
>     >     > I have increased connection_timeout in /etc/my.cnf also.
>     But still
>     >     > facing the same problem.
>     >     >
>     >     > Any insight ?
>     >     >
>     >     >
>     >     > Jun  9 03:36:03 204548-4 /usr/local/sbin/opensips[7575]:
>     >     > DBG:db_mysql:db_mysql_connect: opening connection:
>     >     > mysql://xxxx:xxxx@MySQLserver/opensips
>     >     > Jun  9 03:36:03 204548-4 /usr/local/sbin/opensips[7576]:
>     >     > DBG:core:init_mod_child: type=PROC_TCP_MAIN, rank=-4,
>     module=acc
>     >     > Jun  9 03:36:03 204548-4 /usr/local/sbin/opensips[7576]:
>     >     > DBG:core:init_mod_child: type=PROC_TCP_MAIN, rank=-4,
>     module=auth_db
>     >     > Jun  9 03:36:03 204548-4 /usr/local/sbin/opensips[7576]:
>     >     > DBG:core:db_do_init: connection 0x79e858 not found in pool
>     >     > Jun  9 03:36:03 204548-4 /usr/local/sbin/opensips[7576]:
>     >     > DBG:db_mysql:db_mysql_connect: opening connection:
>     >     > mysql://xxxx:xxxx@MySQLserver/opensips
>     >     > Jun  9 03:39:12 204548-4 /usr/local/sbin/opensips[7567]:
>     >     > ERROR:db_mysql:db_mysql_connect: driver error(2003): Can't
>     >     connect to
>     >     > MySQL server on 'MySQLserver' (110)
>     >     > Jun  9 03:39:12 204548-4 /usr/local/sbin/opensips[7567]:
>     >     > ERROR:db_mysql:db_mysql_new_connection: initial connect failed
>     >     > Jun  9 03:39:12 204548-4 /usr/local/sbin/opensips[7567]:
>     >     > ERROR:core:db_do_init: could not add connection to the pool
>     >     > Jun  9 03:39:12 204548-4 /usr/local/sbin/opensips[7567]:
>     >     > ERROR:auth_db:child_init: unable to connect to the database
>     >     > Jun  9 03:39:12 204548-4 /usr/local/sbin/opensips[7567]:
>     >     > ERROR:core:init_mod_child: failed to initializing module
>     >     auth_db, rank 3
>     >     > Jun  9 03:39:12 204548-4 /usr/local/sbin/opensips[7567]:
>     >     > ERROR:core:main_loop: init_child failed for UDP listener
>     >     > Jun  9 03:39:12 204548-4 /usr/local/sbin/opensips[7341]:
>     >     > ERROR:db_mysql:db_mysql_connect: driver error(2003): Can't
>     >     connect to
>     >     > MySQL server on 'MySQL server' (110)
>     >     > Jun  9 03:39:12 204548-4 /usr/local/sbin/opensips[7341]:
>     >     > ERROR:db_mysql:db_mysql_new_connection: initial connect failed
>     >     > Jun  9 03:39:12 204548-4 /usr/local/sbin/opensips[7341]:
>     >     > ERROR:core:db_do_init: could not add connection to the pool
>     >     > Jun  9 03:39:12 204548-4 /usr/local/sbin/opensips[7341]:
>     >     > ERROR:usrloc:child_init: child(0): failed to connect to
>     database
>     >     > Jun  9 03:39:12 204548-4 /usr/local/sbin/opensips[7341]:
>     >     > ERROR:core:init_mod_child: failed to initializing module
>     usrloc,
>     >     rank 0
>     >     > Jun  9 03:39:12 204548-4 /usr/local/sbin/opensips[7341]:
>     >     > ERROR:core:main_loop: error in init_child for PROC_MAIN
>     >     > Jun  9 03:39:12 204548-4 /usr/local/sbin/opensips[7341]:
>     >     > DBG:xlog:destroy: destroy module...
>     >     >
>     >     > On Wed, Jun 9, 2010 at 1:04 PM, ram <talk2ram at gmail.com
>     <mailto:talk2ram at gmail.com>
>     >     <mailto:talk2ram at gmail.com <mailto:talk2ram at gmail.com>>
>     >     > <mailto:talk2ram at gmail.com <mailto:talk2ram at gmail.com>
>     <mailto:talk2ram at gmail.com <mailto:talk2ram at gmail.com>>>> wrote:
>     >     >
>     >     >     depends on the design
>     >     >
>     >     >     you can have one point of authenticaion and transaction
>     >     >
>     >     >     Ram
>     >     >
>     >     >
>     >     >
>     >     >     On Tue, Jun 8, 2010 at 10:59 PM, Premalatha Kuppan
>     >     >     <premalatha at ngintech.com
>     <mailto:premalatha at ngintech.com> <mailto:premalatha at ngintech.com
>     <mailto:premalatha at ngintech.com>>
>     >     <mailto:premalatha at ngintech.com
>     <mailto:premalatha at ngintech.com> <mailto:premalatha at ngintech.com
>     <mailto:premalatha at ngintech.com>>>>
>     >     wrote:
>     >     >
>     >     >         Thanks a lot.
>     >     >
>     >     >         I have one question. If i route the call to
>     asterisk for
>     >     IVR (
>     >     >         in my case ivr is to authenticate the user to
>     access the
>     >     >         system), who will have the control, meaning who will
>     >     maintian
>     >     >         all the transactions and dailog. Is it
>     opensips/Asterisk ?
>     >     >
>     >     >         Thanks,
>     >     >         Prem
>     >     >
>     >     >         On Tue, Jun 8, 2010 at 9:50 AM, Gabriel Bermudez
>     >     >         <elgabo81 at gmail.com <mailto:elgabo81 at gmail.com>
>     <mailto:elgabo81 at gmail.com <mailto:elgabo81 at gmail.com>>
>     >     <mailto:elgabo81 at gmail.com <mailto:elgabo81 at gmail.com>
>     <mailto:elgabo81 at gmail.com <mailto:elgabo81 at gmail.com>>>> wrote:
>     >     >
>     >     >             Hi,
>     >     >
>     >     >             I basically use the load_balancer module to
>     dispatch to
>     >     >             different
>     >     >             asterisk servers
>     >     >
>     >     >             on the main route block after handling auth,
>     >     >             registrations, etc
>     >     >
>     >     >             if(db_is_user_in("Request-URI", "ccivr")) {
>     >     >                      xlog("The call will be redirect to
>     calling card
>     >     >             server");
>     >     >                      route(3);
>     >     >             }
>     >     >
>     >     >             # route for call handled by calling card servers
>     >     >             route[3] {
>     >     >                    # for INVITEs enable some additional helper
>     >     routes
>     >     >                    if (is_method("INVITE")) {
>     >     >                            t_on_branch("2");
>     >     >                            t_on_reply("2");
>     >     >                            t_on_failure("1");
>     >     >                            if(client_nat_test("15")) {
>     >     >                                    nat_keepalive();
>     >     >                            }
>     >     >                    }
>     >     >
>     >     >                    # prepare the message for the IVR
>     >     >
>     >     >                    # select less loaded IVR
>     >     >                    if(!load_balance("1", "ccivr")) {
>     >     >                            xlog("No IVR available !!!");
>     >     >                            sl_send_reply("503", "Service
>     >     Unavailable");
>     >     >                            exit;
>     >     >                    };
>     >     >
>     >     >                    if(!t_relay()) {
>     >     >                            sl_reply_error();
>     >     >                            exit;
>     >     >                    }
>     >     >
>     >     >                    exit;
>     >     >
>     >     >             }
>     >     >
>     >     >
>     >     >             Hope it helps
>     >     >
>     >     >             Regards,
>     >     >
>     >     >             2010/6/8 ram <talk2ram at gmail.com
>     <mailto:talk2ram at gmail.com>
>     >     <mailto:talk2ram at gmail.com <mailto:talk2ram at gmail.com>>
>     <mailto:talk2ram at gmail.com <mailto:talk2ram at gmail.com>
>     >     <mailto:talk2ram at gmail.com <mailto:talk2ram at gmail.com>>>>:
>     >     >             > I forgot the link
>     >     >             >
>     >     >             > i did this work some time back
>     >     >             >
>     >     >             > Lost the link, google it
>     >     >             >
>     >     >             > Opensips+asterisk+a2b
>     >     >             >
>     >     >             > Ram
>     >     >             >
>     >     >             > On Mon, Jun 7, 2010 at 4:07 PM, Premalatha
>     Kuppan
>     >     >             <premalatha at ngintech.com
>     <mailto:premalatha at ngintech.com>
>     >     <mailto:premalatha at ngintech.com
>     <mailto:premalatha at ngintech.com>> <mailto:premalatha at ngintech.com
>     <mailto:premalatha at ngintech.com>
>     >     <mailto:premalatha at ngintech.com
>     <mailto:premalatha at ngintech.com>>>>
>     >     >             > wrote:
>     >     >             >>
>     >     >             >> Can you pleae guide me how to do this ?
>     >     >             >>
>     >     >             >> On Mon, Jun 7, 2010 at 4:04 PM, Douglas Lane
>     >     >             <doug at wd.co.za <mailto:doug at wd.co.za>
>     <mailto:doug at wd.co.za <mailto:doug at wd.co.za>>
>     >     <mailto:doug at wd.co.za <mailto:doug at wd.co.za>
>     <mailto:doug at wd.co.za <mailto:doug at wd.co.za>>>> wrote:
>     >     >             >>>
>     >     >             >>> Hi Premalatha,
>     >     >             >>>
>     >     >             >>> Perhaps have a look at SEMS for this.
>     >     >             >>>
>     >     >             >>> Thanks
>     >     >             >>> Doug
>     >     >             >>>
>     >     >             >>>
>     >     >             >>> On 2010/06/07 12:30 PM, Premalatha Kuppan
>     wrote:
>     >     >             >>>
>     >     >             >>> Thanks Sebastian.
>     >     >             >>>
>     >     >             >>> I have followed up this link and tried
>     extending the
>     >     >             opensips.cfg file to
>     >     >             >>> route call to Asterisk.
>     >     >             >>>
>     >     >             >>> I doubt/not clear that after IVR (meaning
>     when the
>     >     >             user is authenticated
>     >     >             >>> through IVR) who will handle all the
>     >     transactions and
>     >     >             dialog (OPENSIPS or
>     >     >             >>> Asterisk ) ?
>     >     >             >>>
>     >     >             >>> I want Opensips to handle all the
>     transactions.
>     >     >             >>>
>     >     >             >>> Any insight on this ?
>     >     >             >>>
>     >     >             >>> Thanks,
>     >     >             >>> Prem
>     >     >             >>>
>     >     >             >>> On Mon, Jun 7, 2010 at 3:15 PM, Schumann
>     Sebastian
>     >     >             >>> <Sebastian.Schumann at t-com.sk
>     <mailto:Sebastian.Schumann at t-com.sk>
>     >     <mailto:Sebastian.Schumann at t-com.sk
>     <mailto:Sebastian.Schumann at t-com.sk>>
>     >     >             <mailto:Sebastian.Schumann at t-com.sk
>     <mailto:Sebastian.Schumann at t-com.sk>
>     >     <mailto:Sebastian.Schumann at t-com.sk
>     <mailto:Sebastian.Schumann at t-com.sk>>>> wrote:
>     >     >             >>>>
>     >     >             >>>> Hi Prem
>     >     >             >>>>
>     >     >             >>>> There is a good tutorial at
>     >     >             >>>>
>     >     http://www.opensips.org/Resources/DocsTutAsterisk It
>     >     >             does exactly what you
>     >     >             >>>> need I assume.
>     >     >             >>>>
>     >     >             >>>> For details in writing and extending basic
>     >     >             configuration, you can find
>     >     >             >>>> also the linked documentation there.
>     >     >             >>>>
>     >     >             >>>> Best regards
>     >     >             >>>> Sebastian
>     >     >             >>>>
>     >     >             >>>> > -----Original Message-----
>     >     >             >>>> > From: users-bounces at lists.opensips.org
>     <mailto:users-bounces at lists.opensips.org>
>     >     <mailto:users-bounces at lists.opensips.org
>     <mailto:users-bounces at lists.opensips.org>>
>     >     >             <mailto:users-bounces at lists.opensips.org
>     <mailto:users-bounces at lists.opensips.org>
>     >     <mailto:users-bounces at lists.opensips.org
>     <mailto:users-bounces at lists.opensips.org>>> [mailto:users-
>     <mailto:users->
>     >     <mailto:users- <mailto:users->>
>     >     >             <mailto:users- <mailto:users-> <mailto:users-
>     <mailto:users->>>
>     >     >             >>>> > bounces at lists.opensips.org
>     <mailto:bounces at lists.opensips.org>
>     >     <mailto:bounces at lists.opensips.org
>     <mailto:bounces at lists.opensips.org>>
>     >     >             <mailto:bounces at lists.opensips.org
>     <mailto:bounces at lists.opensips.org>
>     >     <mailto:bounces at lists.opensips.org
>     <mailto:bounces at lists.opensips.org>>>] On Behalf Of
>     >     >             Premalatha Kuppan
>     >     >             >>>> > Sent: Monday, 07. June 2010 11:35
>     >     >             >>>> > To: OpenSIPS users mailling list
>     >     >             >>>> > Subject: [OpenSIPS-Users] OPENSIPS+ASTERISK
>     >     Integration
>     >     >             >>>> >
>     >     >             >>>> > Hi,
>     >     >             >>>> >
>     >     >             >>>> > Can anyone guide me in building the
>     Opensips and
>     >     >             Asterisk Integration.
>     >     >             >>>> >
>     >     >             >>>> > I want to Use OpenSIPS as SIP PROXY
>     (i.e all
>     >     >             transactions and dialogs
>     >     >             >>>> > should be
>     >     >             >>>> > handled by Opensips) and Asterisk to do
>     only IVR
>     >     >             functionality.
>     >     >             >>>> >
>     >     >             >>>> > I appreciate if anyone can guide me in
>     writing a
>     >     >             routing logic from
>     >     >             >>>> > Opensips to
>     >     >             >>>> > Asterisk for IVR and futher call flow
>     OPENSIPS to
>     >     >             handle.
>     >     >             >>>> >
>     >     >             >>>> > Thanks,
>     >     >             >>>> > Prem
>     >     >             >>>> >
>     >     >             >>>>
>     >     >             >>>>
>     >     >             >>>>
>     _______________________________________________
>     >
>


-- 
Bogdan-Andrei Iancu
www.voice-system.ro




More information about the Users mailing list