Re: [OpenSIPS-Users] Load Balancing probing
Hello, Probing mode 2 means your destinations will be probed all the time. You need to manually catch failures within OpenSIPS failure_route and call lb_disable(). OpenSIPS will automatically activate back your failed destination once it properly responds to probing. You might also want to check out the probing_reply_codes [1] parameter. [1] http://www.opensips.org/html/docs/modules/devel/load_balancer.html#id250120 Regards, Vlad Paiu OpenSIPS Developer On 11/07/2011 09:14 AM, Schneur Rosenberg wrote: I'm trying to use load balancing, but I have a question I set the probing mode on 2, now my question is will opensips automatically disable the route if it does not probe or I need to do it manually with lb_disable() ? ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] topology_hiding in dialog module
Hello,
Glad that your issue with topology hiding is fixed in the latest
OpenSIPS stable release.
As far as I know, the issue with Mediaproxy TM is not yet fixed. Maybe
Saúl can provide more info about this.
Regards,
Vlad Paiu
OpenSIPS Developer
On 11/05/2011 10:14 AM, Jayesh Nambiar wrote:
Hi,
I just happened to check this same function after upgrading to 1.7
stable release and it works as expected. So I would really like to use
the stable version. Does anyone know if the issue related to
mediaproxy and tm is solved in the opensips 1.7 stable version? I had
same problem after moving my opensips 1.7 stable into production. Here
is the post I am referring to:
http://lists.opensips.org/pipermail/users/2011-October/019440.html
Is the above problem related to mediaproxy??
--- Jayesh
On Sat, Nov 5, 2011 at 11:56 AM, Jayesh Nambiar jayesh.v...@gmail.com
mailto:jayesh.v...@gmail.com wrote:
Hi All,
I am trying to use the topology_hiding feature in the dialog
module. The problem is opensips does not route the sequential
requests properly. I have tried using the match_dialog function as
described in the module docs:
if (has_totag() is_method(INVITE|ACK|BYE|UPDATE)) {
log(1, Method has To Tag);
if(match_dialog())
{
log(1, ACK Matched Earlier Dialog);
route(default_relay);
exit;
}
}
But the match_dialog function always returns false and opensips is
not able to relay the ACK by matching the dialog. I am definitely
calling create_dialog() on my intial INVITE. I can see the first
log message printed properly, but the second one inside the
match_dialog function is not printed which means match_dialog
returns false. Can anyone guide me to using the match_dialog
function appropriately and accurately so as to use this
light-weight topology-hiding mechanism!!
Any help is greatly appreciated !!
PS: I am using opensips-1.7-beta source as I had some serious
errors related to media-relay and tm module when I had moved to
opensips-1.7 stable version in production.
Thanks,
--- Jayesh
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[OpenSIPS-Users] 401 unauthorized
Hi guys, I'm having some troubles with a NAT client. I have 2 phones (one NATed and one with valid IP). Making SIP calls between them, everything works fine, the problem starts when I'm trying to receive a call from the PSTN, the NATed client is giving me 401 Uanuthorized, but in the valid IP one, everything is working. I got this from my ngrep. U *.*.*.*:5060 - *.*.*.*:55225 INVITE sip:1630751038@*.*.*.*:5062 SIP/2.0. Record-Route: sip:*.*.*.*;lr=on;did=5bf.4529821. Content-Type:application/sdp. To:sip:0151630751038@*.*.*.*. From:sip:1340101000@*.*.*.*;cpc=ordinary;tag=CC203030373433390080C9D0. Allow:INVITE,ACK,OPTIONS,BYE,CANCEL,REGISTER,INFO,UPDATE,PRACK,REFER,SUBSCRIBE,NOTIFY. Supported:100rel. Expires:120. Date:Mon, 07 Nov 2011 11:01:49 GMT. Call-ID:0200CC5230814000367E@TB007439_VOIP1.TB007439. CSeq:1 INVITE. Max-Forwards:69. Timestamp:8440272. User-Agent:TB007439. Contact:sip:1340101000@*.*.*.*:5060. Via: SIP/2.0/UDP *.*.*.*;branch=z9hG4bKcef9.0ca191b4.0. Via:SIP/2.0/UDP *.*.*.*:5060;received=200.152.176.62;branch=z9hG4bKF6DBE15131EED68891B500A3455D133D;rport=5060. Content-Length: 203. . v=0. o=tb640 5 1 IN IP4 *.*.*.*. s=-. c=IN IP4 *.*.*.*. t=0 0. m=audio 61536 RTP/AVP 0 8 4 18 101 13. a=fmtp:4 bitrate=6300;annexa=no. a=rtpmap:101 telephone-event/8000. a=nortpproxy:yes. U *.*.*.*:55225 - *.*.*.*:5060 SIP/2.0 401 Unauthorized. To:sip:0151630751038@*.*.*.*;tag=7f18aab466db6f88i2. From:sip:1340101000@*.*.*.*;cpc=ordinary;tag=CC203030373433390080C9D0. Call-ID:0200CC5230814000367E@TB007439_VOIP1.TB007439. CSeq:1 INVITE. Via: SIP/2.0/UDP *.*.*.*;branch=z9hG4bKcef9.0ca191b4.0. Via:SIP/2.0/UDP *.*.*.*:5060;received=*.*.*.*;branch=z9hG4bKF6DBE15131EED68891B500A3455D133D;rport=5060. Record-Route: sip:*.*.*.*;lr=on;did=5bf.4529821. Timestamp:8440272. Server: Linksys/SPA942-5.1.15(a). WWW-Authenticate: Digest realm=*.*.*.*, nonce=178352f8, qop=auth, algorithm=md5. Content-Length: 0. . I have rtpproxy for NATed clients. Any clues? Eng.º Rodrigo Ferreira Supervisor de Telecomunicações Fone: (13) 4010-1037 Cel: (13) 9615-7774 VIPWay Serviços de Telecomunicações LTDA___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] RTPproxy + Opensips 1.7.0
Hi Razvan,
I added some INFO traces in the select_rtpp_node function in order
to get some clues about what happens (see enclosed file).
Here is what is displayed in my /var/log/messages when an INVITE is
received :
Nov 7 15:21:38 WWW_64Bits ./opensips[19884]: INFO :: (INVITE)
rtpproxy set 1
Nov 7 15:21:38 WWW_64Bits ./opensips[19884]:
INFO:rtpproxy:select_rtpp_node: entering select_rtpp_node
Nov 7 15:21:38 WWW_64Bits ./opensips[19884]:
INFO:rtpproxy:select_rtpp_node: rtpproxy node count = 1
Nov 7 15:21:38 WWW_64Bits ./opensips[19884]:
INFO:rtpproxy:select_rtpp_node: result rtpp_test = 0
Nov 7 15:21:38 WWW_64Bits ./opensips[19884]:
INFO:rtpproxy:select_rtpp_node: entering select_rtpp_node
Nov 7 15:21:38 WWW_64Bits ./opensips[19884]:
INFO:rtpproxy:select_rtpp_node: rtpproxy node count = 1
Nov 7 15:21:38 WWW_64Bits ./opensips[19884]:
INFO:rtpproxy:select_rtpp_node: result rtpp_test = 1
Nov 7 15:21:38 WWW_64Bits ./opensips[19884]:
ERROR:rtpproxy:force_rtp_proxy_body: no available proxies
This is really weird, if rtpp_test returns 1, it should mean that
the rtpproxy socket was found right ? Then why do we have an error
message saying that there are no available proxies ? I'm confused...
Best regards,
Sebastien
Le 04/11/2011 15:23, Razvan Crainea a crit:
Hi Sebastien,
I will try to replicate this scenario and see if I am getting the
same behaviour. I will get back to you later.
Regards,
--
Rzvan Crainea
OpenSIPS Developer
On 11/04/2011 04:20 PM, Sebastien CRUAUX wrote:
I also tried to enter the rtpproxy_sock parameters and the set
IDs in the nh_sockets table and to load the rtpproxy sets from
the database but it did not work either :(
Sebastien
Le 04/11/2011 11:52, Sebastien CRUAUX a crit:
Hi Razvan,
Yes I think I declared the rtpproxy sets correctly, unless
there is some new parameter in the new rtpproxy module that I
forgot :
# - rtpproxy params -
modparam("rtpproxy", "rtpproxy_sock", "1 ==
udp:localhost:12221")
modparam("rtpproxy", "rtpproxy_sock", "2 ==
udp:localhost:1")
Regards,
Sebastien
Le 04/11/2011 11:44, Razvan Crainea a crit:
Hi Sebastien,
Are you sure that when you declare the RTPProxy sets you
allocate them the set identifiers (1 and 2)? Can you send us
the rtpproxy_sock parameters declaration?
Regards,
--
Rzvan Crainea
OpenSIPS Developer
On 11/04/2011 12:27 PM, Sebastien CRUAUX wrote:
Hi,
I am currently migrating my old Opensips 1.6.2 to the new
Opensips 1.7.0 but I am facing some issues with the
configuration of rtpproxy.
The version of rtpproxy I am using is the commit
6b82ff914543d21ff9ddbb797b40a77516348308.
When I start Opensips, the two sets of rtpproxies I
configured are detected :
INFO:rtpproxy:rtpp_test:
rtp proxy udp:localhost:12221 found, support
for it enabled
INFO:rtpproxy:rtpp_test:
rtp proxy udp:localhost:1 found, support
for it enabled
However, when an INVITE is received by Opensips it seems
rtpproxy is not found, consequently the SDP body is not
rewritten :
INFO
:: (INVITE) rtpproxy set 1
ERROR:rtpproxy:force_rtp_proxy_body:
no available proxies
More information about my configuration :
- my Opensips/rtpproxy server has 2 IP addresses, one
opened on the internet, one internal used to communicate
with my VoIP/PSTN gateway
- I have 2 sets of rtpproxies : the 1st one is in bridge
mode for VoIP to PSTN or PSTN to VoIP calls, the 2nd one
only listens on the external IP and is used for SIP to SIP
calls
./rtpproxy -u seb -l 172.17.1.126 172.17.1.131 -s
udp:localhost 12221 -m 18000 -M 18020
./rtpproxy -u seb -l 172.17.1.131 -s udp:localhost
1 -m 18021 -M 18030
- below is the part of my opensips.cfg file which
handles the INVITE requests (I just replaced
[OpenSIPS-Users] Loadbalancing timeout for lb_disable()
Hi
I'm not yet experienced enough in opensips, so please bear with me.
I set up a opensips to loadbalance 2 asterisk servers, in the failure
route I placed
if (t_check_status((408)) t_local_replied(all))
{
lb_disable();
# Try to load balance once again
if ( load_balance(2,pstn,1) )
{
t_on_failure(1);
t_relay();
}
else
{
t_reply(503,Service Unavailable);
}
}
My loadbalancing probing is set to 1 in the database, now when I shut
Asterisk Opensips will keep on trying again and again because asterisk
is not replying, how can I have Opensips realize that it timed out and
create a 408 timeout that will trigger the lb_disable()?
S. Rosenberg
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Re: [OpenSIPS-Users] Loadbalancing timeout for lb_disable()
Your setup looks pretty much like this guy except for your first IF
statement
http://opensips-open-sip-server.1449251.n2.nabble.com/Load-balancer-probe-mode-1-bug-td4245085.html#a5715729
Put this before your first IF statement and then try again
xlog(L_INFO, -- BEFORE LB IF Statement: Call [$rm] ru[$ru] fu[$fu]
si[$si] \n);
On , Schneur Rosenberg rosenberg11...@gmail.com wrote:
Hi
I'm not yet experienced enough in opensips, so please bear with me.
I set up a opensips to loadbalance 2 asterisk servers, in the failure
route I placed
if (t_check_status((408)) t_local_replied(all))
{
lb_disable();
# Try to load balance once again
if ( load_balance(2,pstn,1) )
{
t_on_failure(1);
t_relay();
}
else
{
t_reply(503,Service Unavailable);
}
}
My loadbalancing probing is set to 1 in the database, now when I shut
Asterisk Opensips will keep on trying again and again because asterisk
is not replying, how can I have Opensips realize that it timed out and
create a 408 timeout that will trigger the lb_disable()?
S. Rosenberg
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Re: [OpenSIPS-Users] RTPProxy + Call_control
Hi Nick, Try what this message says http://www.mail-archive.com/users@lists.opensips.org/msg17830.html Regards, 2011/11/4 Nick nick_ch...@ezmobo.com ** Hello About Call Control : You need to setup the following components: - OpenSIPS callcontrol module, version =1.5 == It's OK. - Call Control application (this software) - CDRTool rating engine, version =6.7.0 == - The RatingEngine IP address - The mysql connection details to cdrtool, radius, opensisps, siptrace and mediaproxy MySQL databases, which are described in the following sections of this document - The mediaDispatcher address This is a cdrtool documtn. http://download.dns-hosting.info/CDRTool/doc/INSTALL.txt But, I don't used radius, mediaproxy. How does cdrtool work??? When I input command start-stop-daemon --start --quiet --pidfile /var/run/ratingEngine.pid --exec /var/www/CDRTool/scripts/ratingEngine.php It always display error. Database error for query select `value` from memcache where `key` = 'destinations': ()Database DB_opensips error: Table 'opensips.trusted_peers' doesn't exist (1146) select * from trusted_peers Database error for query select * from billing_profiles order by name: ()Database error for query select *, UNIX_TIMESTAMP(startDate) as startDateTimestamp, UNIX_TIMESTAMP(endDate) as endDateTimestamp from billing_rates_history order by name ASC,destination ASC,startDate DESC: ()Database error for query select * from billing_holidays order by day: ()Database error for query select * from billing_enum_tlds: ()PHP Fatal error: Call to undefined function posix_setsid() in /var/www/CDRTool/library/rating_server.php on line 81 How to solve this problem?? I want to used opensips, call_control, rtpproxy, mysql and cdrtool, But I don't want to used mediaproxy and radius. I need billing system with opensips. Thanks for your support. Nick On 2011年11月04日 10:06, Nick wrote: Hello So, I used rtpproxy + callcontrol + CDRTOOL . Is it OK??? In Document, I don't this part. Thanks for your support. Nick On 2011年11月04日 07:08, Saul Ibarra Corretge wrote: On Nov 2, 2011, at 5:08 AM, Nick wrote: Hello I see document. In opensips, it has call_control this module. But it can't support rtpproxy, only support media-proxy If I want to a billing system for opensips. Can you give me a another suggest?? CallControl doesn't require MediaProxy to operate, you may use any other media relaying software. Regards, ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Loadbalancing timeout for lb_disable()
Ok it was working it just took a very long time until it timed out.
thank you
On Mon, Nov 7, 2011 at 10:59 PM, duane.lar...@gmail.com wrote:
Your setup looks pretty much like this guy except for your first IF
statement
http://opensips-open-sip-server.1449251.n2.nabble.com/Load-balancer-probe-mode-1-bug-td4245085.html#a5715729
Put this before your first IF statement and then try again
xlog(L_INFO, -- BEFORE LB IF Statement: Call [$rm] ru[$ru] fu[$fu]
si[$si] \n);
On , Schneur Rosenberg rosenberg11...@gmail.com wrote:
Hi
I'm not yet experienced enough in opensips, so please bear with me.
I set up a opensips to loadbalance 2 asterisk servers, in the failure
route I placed
if (t_check_status((408)) t_local_replied(all))
{
lb_disable();
# Try to load balance once again
if ( load_balance(2,pstn,1) )
{
t_on_failure(1);
t_relay();
}
else
{
t_reply(503,Service Unavailable);
}
}
My loadbalancing probing is set to 1 in the database, now when I shut
Asterisk Opensips will keep on trying again and again because asterisk
is not replying, how can I have Opensips realize that it timed out and
create a 408 timeout that will trigger the lb_disable()?
S. Rosenberg
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Re: [OpenSIPS-Users] topology_hiding in dialog module
Hi Saul,
Is there at least a temporary fix available for this problem. Something
like, if I compile the mediaproxy module of 1.6.4 version and load it in
the 1.7 modules directory, should it work?? Just a thought.
Thanks,
--- Jayesh
Hello,
Glad that your issue with topology hiding is fixed in the latest
OpenSIPS stable release.
As far as I know, the issue with Mediaproxy TM is not yet fixed. Maybe
Sa?l can provide more info about this.
Regards,
Vlad Paiu
OpenSIPS Developer
On 11/05/2011 10:14 AM, Jayesh Nambiar wrote:
Hi,
I just happened to check this same function after upgrading to 1.7
stable release and it works as expected. So I would really like to use
the stable version. Does anyone know if the issue related to
mediaproxy and tm is solved in the opensips 1.7 stable version? I had
same problem after moving my opensips 1.7 stable into production. Here
is the post I am referring to:
http://lists.opensips.org/pipermail/users/2011-October/019440.html
Is the above problem related to mediaproxy??
--- Jayesh
On Sat, Nov 5, 2011 at 11:56 AM, Jayesh Nambiar jayesh.v...@gmail.com
mailto:jayesh.v...@gmail.com wrote:
Hi All,
I am trying to use the topology_hiding feature in the dialog
module. The problem is opensips does not route the sequential
requests properly. I have tried using the match_dialog function as
described in the module docs:
if (has_totag() is_method(INVITE|ACK|BYE|UPDATE)) {
log(1, Method has To Tag);
if(match_dialog())
{
log(1, ACK Matched Earlier Dialog);
route(default_relay);
exit;
}
}
But the match_dialog function always returns false and opensips is
not able to relay the ACK by matching the dialog. I am definitely
calling create_dialog() on my intial INVITE. I can see the first
log message printed properly, but the second one inside the
match_dialog function is not printed which means match_dialog
returns false. Can anyone guide me to using the match_dialog
function appropriately and accurately so as to use this
light-weight topology-hiding mechanism!!
Any help is greatly appreciated !!
PS: I am using opensips-1.7-beta source as I had some serious
errors related to media-relay and tm module when I had moved to
opensips-1.7 stable version in production.
Thanks,
--- Jayesh
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[OpenSIPS-Users] conecting sipp to opensips
i want to set basic connection of one person calling to other plez i want to finish my project fast ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
