Re: [OpenSIPS-Users] Dispatcher or Load Balancer Module??
Hi Abdul, I was working on the Load Balancer task a few days back using OpenSIPS I was successful to implement it, will you please explain a little bit more about your No. of INVITES so that we can proceed accordingly. Regards, Faisal Rehman From: M.Abdulaziz malduwa...@ksu.edu.sa To: users@lists.opensips.org Sent: Thursday, November 3, 2011 4:56 AM Subject: Re: [OpenSIPS-Users] Dispatcher or Load Balancer Module?? Can any one help please. -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Dispatcher-or-Load-Balancer-Module-tp6954965p6957454.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ 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] Dispatcher or Load Balancer Module??
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 the Loadbalancing works. But i dont know if this is what you are looking for? BR Max M. Am 03.11.2011 00:56, schrieb M.Abdulaziz: Can any one help please. -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Dispatcher-or-Load-Balancer-Module-tp6954965p6957454.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- 42com Telecommunication GmbH Straße der Pariser Kommune 12-16 / D-10243 Berlin E-mail: m...@42com.com Homepage: www.42com.com Firmenangaben/Company information: Handelsregister/Commercial register: Amtsgericht Berlin HRB 99071 B Umsatzsteuer-ID/VAT-ID: DE223812306, Geschäftsführer/CEO: Thomas Reinig, Alexander Reinig Diese E-Mail enthält Informationen von 42com Telecommunication GmbH. Diese sind möglicherweise vertraulich und ausschließlich für den Adressaten bestimmt. Sollten Sie diese elektronische Nachricht irrtümlicherweise erhalten haben, so informieren Sie uns bitte unverzüglich telefonisch oder per E-Mail. This message is intended only for the use of the individual or entity to which it is addressed. If you have received this message in error, please delete the message and notify us immediately. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Dispatcher or Load Balancer Module??
Max, Faisal Thank you very much for contributing The idea that I want to implement is to store # of current invites going to each distenation (which is an opensips proxy server) in a table in the load balancer machine then based on the # of invites I want to make the route (the destination with the least # of invites is the best one to forward calls to). I know that this is not the best way to balance the load for sip networks but I have to implement this method. As far as I understand from the load balancer module it depends on the load per resource which is a different thing than what I want to do. But is there any way to adjust this module or the dispatcher module or you suggest me to begin writing my design from the scratch using for example Java language? Thank you -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Dispatcher-or-Load-Balancer-Module-tp6954965p6958599.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Change routing after 200OK
Hello,
So in the onreply_route, do you want to route just the 200 Ok to another
provider, or the initial Invite ?
Both cases would be kind of bogus, because from the 1st provider's point
of view the dialog was established, so you would first need to terminate
the call on their side.
Can't you determine if the destination supports RFC2833 prior to
actually routing the Invite ?
Regards,
Vlad Paiu
OpenSIPS Developer
On 11/02/2011 06:31 PM, mick...@winlux.fr wrote:
Hi List !
I'm looking for a solution to a problem : some providers don't support
RFC2833 (Payload 101), but my architecture only support this payload.
So I'd like to route the call to another provider if the first one, to who
I've sent the call, doesn't support RFC2833.
I'd like to use the REPLY_ROUTES with a code like this :
if(t_check_status(200))
{
if(has_body(application/sdp))
{
if ! (codec_exists_re(telephone-event))
{
// ROUTE TO ANOTHER PROVIDER
}
}
}
The best way is to use use_next_gw(), but it's not possible in a
REPLY_ROUTE.
Have you got any idea, or solution ?
Feel free to ask me if you have some questions !
Thanks,
Best regards,
Mike
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Re: [OpenSIPS-Users] Dispatcher or Load Balancer Module??
Max, Faisal Thank you very much for contributing The idea that I want to implement is to store # of current invites going to each distenation (which is an opensips proxy server) in a table in the load balancer machine then based on the # of invites I want to make the route (the destination with the least # of invites is the best one to forward calls to). I know that this is not the best way to balance the load for sip networks but I have to implement this method. As far as I understand from the load balancer module it depends on the load per resource which is a different thing than what I want to do. But is there any way to adjust this module or the dispatcher module or you suggest me to begin writing my design from the scratch using for example Java language? Thank you -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Dispatcher-or-Load-Balancer-Module-tp6954965p6958603.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Dispatcher or Load Balancer Module??
Hello, If you configure all your destinations to have the same resources, then overall the only thing that will count for load-balancing will be actual number of calls per destination, so the total number of Invites per destination. This should meet your needs. Regards, Vlad Paiu OpenSIPS Developer On 11/03/2011 12:57 PM, M.Abdulaziz wrote: Max, Faisal Thank you very much for contributing The idea that I want to implement is to store # of current invites going to each distenation (which is an opensips proxy server) in a table in the load balancer machine then based on the # of invites I want to make the route (the destination with the least # of invites is the best one to forward calls to). I know that this is not the best way to balance the load for sip networks but I have to implement this method. As far as I understand from the load balancer module it depends on the load per resource which is a different thing than what I want to do. But is there any way to adjust this module or the dispatcher module or you suggest me to begin writing my design from the scratch using for example Java language? Thank you -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/Dispatcher-or-Load-Balancer-Module-tp6954965p6958599.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ 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
[OpenSIPS-Users] setting 1st opensip client server call
m doing it 4 1st time i m final yr engg student doing it as my proj plz help i dont know wr to download dependencies nd save them tn instalation commands are with me plz ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] [NEW ] New OpenSIPS Control Panel Release
Hi all, We have a new release for the OpenSIPS Control Panel. The new release (4.1) fixes most of the bugs that the old version had and introduces two new tools: - *TViewer* (aka Table Viewer) which allows you to handle generic DB tables (operations like read, add, remove, update). This new tool becomes very helpful if you want to manage some extra tables other than the ones already handled by Control Panel's tools. - *ACL Management* - this tool controls the grp table and allows you to have some predefined groups (which you can define this in the tool's config file) and assign users to these groups from the web interface. The new release can be downloaded from here https://sourceforge.net/projects/opensips-cp/files/latest/download?source=files. Best regards, Alex Ionescu ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] [NEW] OpenSIPS pkg memory configurable from command line (-M)
Very nice. I think that will help a lot of people when those type of issues come up. Seems like a lot of emails are sent to the mailing list about that. I think I saw an email a couple of days from someone so hopefully they are watching their emails. On , Ovidiu Sas o...@voipembedded.com wrote: Hello all, The trunk version of OpenSIPS has the ability to configure the amount of PKG memory via the command line (there's no need to recompile opensips in order to change the size of pkg memory). The switch that controls the size of the pkg mem is 'M': -M nr Size of pkg memory allocated in Megabytes Regards, Ovidiu Sas ___ 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
[OpenSIPS-Users] Missing calls in CDR
Hi guys, Im trying to figure out why there are missing calls in our CDR. We are not using any other app but opensip to populate the CDRs but at the end of the day, we are like missing 1,000 calls in our CDRs, We generate lots of traffic and I'm thinking if query_buffer_size=5 query_flush_time=10 Could be the reason why. If we get a lot of traffic, should the query_flush_time be higher or lower? OR should we change buffer_size instead? Would love to hear any other ideas from you guys TIA ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Hung dialogs
I've run into an issue where *very* rarely, dialogs will hang, that is,
the dialog will continue to exist even though the call never connected. In
the example I managed to track down, the proxy should reject the call
(because a limit of concurrent calls has been reached) and at first, it
seems like it's going to... but then it sends a 100 Trying and continues to
route the call to the destination. From there, everything spirals and the
dialog doesn't die until it hits the global timeout.
The relevant portion of my script looks like:
# Check gateway calls total
if ($avp(custgwtotal_limit) != 0 $avp(custgwtotal_limit) != NULL
$avp(custgwtotal_limit) != )
{
get_profile_size(custgwtotal,$rd,$avp(custgwtotal_count));
if (!$avp(custgwtotal_count) $avp(custgwtotal_limit))
{
sl_send_reply(486,Endpoint Session Limit);
exit;
}
}
set_dlg_profile(custgwtotal,$rd); #count total calls to customer's IP
I've attached a pcap example of what the signaling looks like when the
dialog hangs. For this example, the output in dlg_list looks like:
dialog:: hash=629:349156783
state:: 3
user_flags:: 0
timestart:: 1320336333
timeout:: 1320357957
callid:: did2.139.1120741
from_uri:: sip:16034299966@64.136.174.30
to_uri:: sip:16038869119@184.106.218.8
caller_tag:: 1ae26ee84c61df6b84baef371d2bf5e4
caller_contact:: sip:184.106.219.203:5060;transport=udp
callee_cseq:: 0
caller_route_set::
caller_bind_addr:: udp:184.106.218.8:5060
callee_tag:: as58f4e19a
callee_contact:: sip:16038869119@71.168.70.47
caller_cseq:: 2
callee_route_set::
callee_bind_addr:: udp:184.106.218.8:5060
Any help would be appreciated.
Thanks,
Ryan
hung_call.pcap
Description: Binary data
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Re: [OpenSIPS-Users] Hung dialogs
Have you looked at the following
http://www.opensips.org/html/docs/modules/devel/dialog.html#id293895
OpenSIPS will send OPTIONS pings to whoever you want to and if they don't
answer back because they don't recognize the Dialog then opensips will send
a BYE to both caller and callee and kill the dialog.
Does that help any?
On , Ryan Revels r...@revelous.net wrote:
I've run into an issue where very rarely, dialogs will hang, that is,
the dialog will continue to exist even though the call never connected.
In the example I managed to track down, the proxy should reject the call
(because a limit of concurrent calls has been reached) and at first, it
seems like it's going to... but then it sends a 100 Trying and continues
to route the call to the destination. From there, everything spirals and
the dialog doesn't die until it hits the global timeout.
The relevant portion of my script looks like:
# Check gateway calls total
if ($avp(custgwtotal_limit) != 0 $avp(custgwtotal_limit) != NULL
$avp(custgwtotal_limit) != )
{
get_profile_size(custgwtotal,$rd,$avp(custgwtotal_count));
if (!$avp(custgwtotal_count)
{
sl_send_reply(486,Endpoint Session Limit);
exit;
}
}
set_dlg_profile(custgwtotal,$rd); #count total calls to customer's IP
I've attached a pcap example of what the signaling looks like when the
dialog hangs. For this example, the output in dlg_list looks like:
dialog:: hash=629:349156783
state:: 3
user_flags:: 0
timestart:: 1320336333
timeout:: 1320357957
callid:: did2.139.1120741
from_uri:: sip:16034299966@64.136.174.30
to_uri:: sip:16038869119@184.106.218.8
caller_tag:: 1ae26ee84c61df6b84baef371d2bf5e4
caller_contact:: sip:184.106.219.203:5060;transport=udp
callee_cseq:: 0
caller_route_set::
caller_bind_addr:: udp:184.106.218.8:5060
callee_tag:: as58f4e19a
callee_contact:: sip:16038869119@71.168.70.47
caller_cseq:: 2
callee_route_set::
callee_bind_addr:: udp:184.106.218.8:5060
Any help would be appreciated.
Thanks,
Ryan
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Re: [OpenSIPS-Users] RTPProxy + Call_control
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, -- Saúl Ibarra Corretgé AG Projects ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
