[OpenSIPS-Users] SBC Edge
I am struggling mightily. I just installed OpenSIPS, but very confused. I've deployed hundreds of Asterisk boxes, but this is challenging.Ideally, I want to setup OpenSIPS to handle far end NAT traversal and forward REGISTER requests to my switch (Happens to be a Metaswitch). In the commercial world, SBC's allow end user devices to use their public IP as a outbound proxy, and the switch sees the IP address of the proxy as it's registered location.Is this possible? I have no idea how to do that.joe ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Manage keep-alives
Hi Jorge,
If these keep-alives are SIP requests, you need to answer, otherwise the
sender will do retransmission :)
So, in the beginning of the script, you can do:
if (is_method("OPTIONS") && uri==myself && $rU==NULL) {
send_reply("200","ok");exit;}
That means -> if you get an OPTIONS with the IP of your server in RURI
and without username part -> it is a ping
Regards,
Bogdan
On 03/01/2012 01:56 PM, Jorge Ortea wrote:
Hi,
my question is how manage the keep-alives messages ??
I understand that the keep-alives only serve to keep open NAT.
Now, to avoid manage this messages in my routing logic, i am
discarding this messages, is this correct? or i should respond.
I too have detected that when my cisco telephones don't receive
response they keep sending keep-alives.
Thanks.
Regards.
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--
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
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Re: [OpenSIPS-Users] Can you drop a 487 Request Terminated?
Hi,
any call must have a final response (rejected or answered)...So you can
discard the 487 from GW, but you need to provide an anwer back to
callerI guess you do not want to let the guy in dead air :P
Regards,
Bogdan
On 03/01/2012 07:11 PM, discodo...@aol.com wrote:
Thanks. So other than that there is no way to just discard/drop the
487 response from the gateway and not pass it to caller. Like send the
reply to a black hole?
-Original Message-
From: Vlad Paiu
To: users
Sent: Thu, Mar 1, 2012 1:41 am
Subject: Re: [OpenSIPS-Users] Can you drop a 487 Request Terminated?
Hello,
You can arm a failure_route for the Invites to the gateways, and
inside the failure_route change the status code to whatever you want
by doing
t_reply("500","Cannot Route");
or whatever message that you want.
Regards,
Vlad
Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com
On 03/01/2012 08:33 AM, discodo...@aol.com wrote:
When I setup a call then cancel the call I am getting a 487 from my
gateway that is relayed to the client.
I don't wish to show the 487 to the client. Is it possible to drop
the reply for the 487?
I am hoping someone could let me know if this is possible.
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Users mailing list
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___
Users mailing list
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--
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
___
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Re: [OpenSIPS-Users] Can you drop a 487 Request Terminated?
Thanks. So other than that there is no way to just discard/drop the 487
response from the gateway and not pass it to caller. Like send the reply to a
black hole?
-Original Message-
From: Vlad Paiu
To: users
Sent: Thu, Mar 1, 2012 1:41 am
Subject: Re: [OpenSIPS-Users] Can you drop a 487 Request Terminated?
Hello,
You can arm a failure_route for the Invites to the gateways, andinside
the failure_route change the status code to whatever you wantby doing
t_reply("500","Cannot Route");
or whatever message that you want.
Regards,
Vlad
Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com
On 03/01/2012 08:33 AM, discodo...@aol.com wrote:
When I setup a call then cancel the call I am getting a 487 from
my gateway that is relayed to the client.
I don't wish to show the 487 to the client. Is it possible to
drop the reply for the 487?
I am hoping someone could letme know if this is possible.
___
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
[OpenSIPS-Users] Manage keep-alives
Hi, my question is how manage the keep-alives messages ?? I understand that the keep-alives only serve to keep open NAT. Now, to avoid manage this messages in my routing logic, i am discarding this messages, is this correct? or i should respond. I too have detected that when my cisco telephones don't receive response they keep sending keep-alives. Thanks. Regards. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] OpenSIPs-CP -Pro blems with ERROR:mi_fifo:mi_fifo_server:
Hi, It does not work because "Matching string length" has to be integer or, in case you use a REGEX you can leave it blank, because it will be ignored anyway. For the permissions module you have to specify the mask using the number of bits, just like you did (24 or 32, or what ever you need). The warning you get is because you have specified a mask like 24 and most probably you have used an IP that does NOT look like this xxx.xxx.xxx.0. So, OpenSIPS expect this kind of IP when you use a 24 bits mask. But don't worry, it's just a warning, it will still work fine. Regards, Alex On 02/29/2012 07:35 PM, Alexandre Keller wrote: I might be doing something wrong. I'm kind of new on OpenSIPs, I've only worked with Asterisk for the past 6 years. Here is my Rule, it's a testing rule. *Dialplan ID: 1* *Rule Priority: 1* *Matching Operator: REGEX* *Matching Regular Expression: 123** *Matching String Length: ** *Substitution Regular Expression: BLANK* *Replacement Expression: BLANK* *Attributes: NONE* When a push ADD button: Failed to issue query, error message : MDB2 Error: syntax error Another strange thing is when I add a Gateway on PERMISSIONS page. I fill the netmask field with 255.255.255.0, but it does not save the informations. I tried filling with 24 (bits), and it saved just fine. Is it correct? When I APPLY CHANGES TO SERVER the following message appears on syslogd: Feb 29 14:33:25 vm-opensips /usr/sbin/opensips[4281]: WARNING:core:mk_net: invalid network address/netmask combination fixed... As I said before, I must be doing something wrong. Thanks again. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Can you drop a 487 Request Terminated?
Hello,
You can arm a failure_route for the Invites to the gateways, and inside
the failure_route change the status code to whatever you want by doing
t_reply("500","Cannot Route");
or whatever message that you want.
Regards,
Vlad
Vlad Paiu
OpenSIPS Developer
http://www.opensips-solutions.com
On 03/01/2012 08:33 AM, discodo...@aol.com wrote:
When I setup a call then cancel the call I am getting a 487 from my
gateway that is relayed to the client.
I don't wish to show the 487 to the client. Is it possible to drop
the reply for the 487?
I am hoping someone could let me know if this is possible.
___
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] Packet Loss and its Solution
I don't have anything related to these troubleshooting and tweaking readily available in my mind, I already wrote the pseudo routine/steps to resolve the issue. As Sir Bogdan has stated openSIPS dont deal with RTPs so you can safely conclude right away that its not OpenSIPS causing the packet losses. I will still insist on focusing/troubleshooting the 1st three layers on your network or on server. On Thu, Mar 1, 2012 at 12:04 PM, Faisal Rehman wrote: > Hi Sammy, > > Sure I will paste the stats here but will you please tell me some > important steps to be performed on software side. Here I am talking about > the OS, kernel upgrade & advanced system administration level stuff that > can reduce the packet loss immediately. Through the final conclusions I > have become to know that its not the OpenSIPS problem but it regards with > physical, software & network upgrades. > > Regards, > > Faisal Rehman > -- > *From:* Sammy Govind > *To:* OpenSIPS users mailling list > *Sent:* Tuesday, February 28, 2012 10:38 AM > *Subject:* Re: [OpenSIPS-Users] Packet Loss and its Solution > > Hi Faisal, > > Can you copy/paste the stats of "ifconfig eth*N*" on which traffic is > terminating. I just wanted to see the error and dropped packets on physical > interface. > Hardware and physical connectivity plays major role in packet > losses.On*Asterisk server > * Jitter options might help you but this is media-proxy, I assume from > the interface you are viewing, the packets are shown as lost. So could it > be heavy media traffic flowing through the interface and media proxy is > unable to use much CPU processing power to process all the RTPs ? > > Before troubleshooting the software application I suggest start digging > the networking interfaces and tweak the eth*N and *related properties of > kernel to maximize the throughput. > > This would be just how I'd go with this kind of problem. > > Regards. > Sammy > > > On Tue, Feb 28, 2012 at 3:45 AM, Muhammad Danish Moosa < > danishmo...@gmail.com> wrote: > > Helo Bogdan > > only signalling packets can be lost on opensips? > > But rtp streams are faster and frequent and have high impact on voice > quality. He seems to ask packet losses on rtp packets. Even if the problem > is identified what are the clues to solve the problem? > > > > On Tue, Feb 28, 2012 at 3:14 AM, Bogdan-Andrei Iancu > wrote: > > ** > Hi Faisal, > > Let me comment a bit on the loss at software - packages can be discarded > at TCP/IP stack level (by kernel) if no application is reading the data > (or no reading as fast as the data comes). > > You can check on this (if opensips is able to process all incoming > traffic, without having the kernel to discard data because of full > buffering on sockets) via some statistics from the NET class : > http://www.opensips.org/Resources/DocsCoreStats17#toc17 > > An overall idea over the load in opensips (if you have idle processes or > not) can be monitored via the LOAD stats: > http://www.opensips.org/Resources/DocsCoreStats17#toc14 > > Regards, > Bogdan > > > On 02/27/2012 04:02 PM, Faisal Rehman wrote: > > Hi Everyone, > > I am facing huge packet loss in my server, so I am here to share with > you some of the output of packets losses that you can see in attached > image. Secondly I have few questions that I want to discuss with: > > 1. How can we reduce the packet loss to a minimum in an asterisk server, > I mean I just want to know more detailed info about packet loss reduction. > 2. I am calculating packet loss following that link > http://www.linuxjournal.com/article/9398 where there is written that 20% > loss is acceptable, but if you see the attached image what will be your > conclusions about packet loss here? > 3. At last but not least I just want to know the responsibilities of the > software & the network, I mean how much software or network is responsible > for packet loss? > 4. What are the best possible ways to reduce the packet loss to a minimum > extent? > > > Regards, > > Faisal Rehman > > > ___ > Users mailing > listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > -- > Bogdan-Andrei Iancu > OpenSIPS Founder and Developerhttp://www.opensips-solutions.com > > > ___ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > -- > Muhammad Danish Moosa > > " The core of mans' spirit comes from new experiences. "___ Christopher > McCandless > > ___ > 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 mai
