thanks. Let me try to test.
wilddra...@sina.com
发件人： Liviu Chircu
发送时间： 2014-10-16 06:12
收件人： users
主题： Re: [OpenSIPS-Users]回复：Re: About "Not enough free memory, no more shm
memory and out of mem" Error
Find your "opensipsctlrc" file (common dirs: /etc/opensips or
/usr/local/opensips/etc), g
Find your "opensipsctlrc" file (common dirs: /etc/opensips or
/usr/local/opensips/etc), go to the bottom and change:
# STARTOPTIONS=
into
STARTOPTIONS="-m64 -M8"
Best regards,
Liviu Chircu
OpenSIPS Developer
http://www.opensips-solutions.com
On 16.10.2014 01:00, wilddra...@sina.com wrote:
op
opensipsctl start
i have not set any param related to mem.
- 原始邮件 -
发件人：Miha
收件人：OpenSIPS users mailling list
主题：Re: [OpenSIPS-Users] About "Not enough free memory, no more shm memory and
out of mem" Error
日期：2014年10月15日 20点27分
来自新浪邮箱手机网页版
__
Hi, all!
OpenSIPS 1.11.3[1], 1.10.3[2] as well as 1.8.6[3] (missed it in the
initial announce :)) have just been released!
All versions are stable and contain the latest bug fixes. You can
download them from the links below.
Enjoy the new releases and thank you all for your valuable contributi
former 1.12, current 2.1.1 is devel version and never released as stable.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 15.10.2014 19:51, Satish Patel wrote:
You mean say 1.12.x isn't stable yet?
On Wed, Oct 15, 2014 at 12:48 PM, Bogdan-And
I wouldn't recommended as the trunk GIT branch is the development one,
so , by definition, not stable.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 15.10.2014 19:46, Satish Patel wrote:
can i use this GIT branch in production?
On Wed, Oct 1
Is it because the packet MTU is too big? It can't handle size bigger than 1472
bytes?
The last line looks strange to me.
a=candidate:2 2 UDP 1694498814 124.193.138.210 5060 typ srflx raddr
192.168[|sip]
At 2014-10-15 22:50:43, "Bogdan-Andrei Iancu" wrote:
The trace definitely shows linp
Hi,
OpenSIPS is not ISUP aware. But if you have the ISUP payload, you can
encapsulate it and make OpenSIPS to send a generic SIP request with the
ISUP payload.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 13.10.2014 11:19, Лытаев Антон Вик
Hi,
First, please use separate email threads for separate topics .
Now, going back to your question - take a look at the dispatcher module:
http://www.opensips.org/html/docs/modules/1.11.x/dispatcher.html
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions
Hi Satish,
In devel branch, there was an error on handling the default value for
returned value. The M10 (explicitly setting the limit to 10) is a hack,
it has to work without it.
This error was fixed in the mean while on GIT, see :
https://github.com/OpenSIPS/opensips/pull/357
So, if you
Great!!!
After putting "FM10" now it working! what the hack is FM10?
On Wed, Oct 8, 2014 at 6:22 AM, Bogdan-Andrei Iancu
wrote:
> Hi Satish,
>
> Could you add as extra flag to ds_select_xxx() the "M10" ? That will make
> the overall flags string "FM10".
>
> Try with that flags and let me kn
The trace definitely shows linphone not answering to the INVITE handled
via mediaproxy (INVITE goes to linphone but nothing back).
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 15.10.2014 17:32, george wu wrote:
//part 2
103.24.228.158.
*On October 21st, 2014, Las Vegas will be the the "OpenSIPS" city!*
*Pete Kelly - **Sourcevox - */"//Solving Business Problems with
OpenSIPs: Least Cost Routing"/
» In addition to standard modules, OpenSIPS comes with a powerful
"Turing complete" scripting language which truly allows you to do
Hi, Bogdan:
Is this a bug? When the server relay invite message to the callee in the last
line:
a=candidate:2 2 UDP 1694498814 124.193.138.210 5060 typ srflx raddr
192.168[|sip]
At 2014-10-15 22:31:41, "george wu" wrote:
Hi, Bogdan:
The following is the tcpdump with mediaproxy on.
//part 2
103.24.228.158.5060 > 124.193.138.210.6001: SIP, length: 1472
INVITE sip:test2@192.168.1.3:5080 SIP/2.0
Record-Route:
Via: SIP/2.0/UDP 103.24.228.158:5060;branch=z9hG4bK57b9.3415f0e2.0
Via: SIP/2.0/UDP
192.168.1.3:5070;received=124.193.138.210;branch=z9hG4bK.FohJ-P
Hi, Bogdan:
The following is the tcpdump with mediaproxy on.
I can't see any problem. That means it is linphone problem which can't
understand ice?
Actually as I said before, if I use tcp/tls, ice actually works on linphone.
What should I do more to test it?
George Wu
/// part 1
124.193.
Hi, Bogdan:
server 103.24.228.158.5060
caller 124.193.138.210.5758
callee 124.193.138.210.6001
1) first I enable mediaproxy,
2) invite, tcpdump, callee not reply.
3) kill media-relay
4) invite, tcpdump, callee replied
So it must be something with the mediaproxy.
what should I dump further so tha
George, you can get all that info by using ngrep or tcpdump to see the
actual traffic.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 15.10.2014 14:25, george wu wrote:
Hi, Bogdan:
1)
I can find the ip from the pinging message, but can't get
Hi, Bogdan:
I will try ngrep and tcpdump.
But I am quite sure it is mediaproxy problem, since I can print out all
the debug message by linphonec.
If I enable mediaproxy, the linphone doesn't print any thing since it doesn't
get message from opensips.
Once I kill mediaproxy, the linphone gets
Hi, Bogdan:
I will try ngrep and tcpdump.
But I am quite sure it is mediaproxy problem, since I can print out all
the debug message by linphonec.
If I enable mediaproxy, the linphone doesn't print any thing since it doesn't
get message from opensips.
Once I kill mediaproxy, the linphone gets an
I have not set any mem param in opensips.cfg
and start opensip use "opensipsctl start" only.
wilddra...@sina.com
From: Miha
Date: 2014-10-15 20:27
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] About "Not enough free memory, no more shm memory
and out of mem" Error
how do you
Can anybody share your mediaproxy configuration?
I am using mediaproxy to work with ice.
I modify the script from rtpproxy. Finally it turns out
it breaks some invite relay logic.
The route logic configuration is very hard. The original rtpproxy is generated
from menuconfig.
Since there is no op
Hi George,
Not sure if a media relay process has anything to do with the ability to
send traffic to an UAC - do you actually see with ngrep/tcpdump the
request (on the network level) sent by opensips to the UAC ?
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips
how do you start opensips, how much private mem and shared mam do you
asigne?
brM
On 15/10/2014 14:25, wilddra...@sina.com wrote:
Hi, My friends,
When I do a stress testing for opensips(version: 1.11.2TLS) , I
received the following errors. How to tuning it to solve these problem.
Hi, My friends,
When I do a stress testing for opensips(version: 1.11.2TLS) , I received
the following errors. How to tuning it to solve these problem.
anyone can help me, thanks.
opensips[1362]: ERROR:dialog:dialog_update_db: could not add another dialog to
db
opensips[1358]: WARN
Hi, Bogdan:
I think I have found the problem.
I am using mediaproxy. If I kill that proxy.
suddenly the uac can get the message.
So it is quite obvious that my mediaproxy setting is not correct.
Just I don't know how to fix it. I modify it from my old rtpproxy setting.
George
Hi, My friends,
When I do a stress testing for opensips(version: 1.11.2TLS) , I received
the following errors. How to tuning it to solve these problem.
anyone can help me, thanks.
opensips[1362]: ERROR:dialog:dialog_update_db: could not add another dialog to
db
opensips[1358]: WARN
Hi Kaan,
I'm glad the first part is properly working now.
Your ACK (as it is for a 200 OK) should be routed based on Route headers
(record route and loose route mechanism). TO better understand this,
please take a look to this webinar:
http://www.opensips.org/Documentation/Webinars#toc12
Hi George,
It looks like you use the add_rcv_param() function when processing the
REGISTER.
That is not the correct way of fixing the private registrations, but
rather via the fix_nated_register() function :
http://www.opensips.org/html/docs/modules/1.11.x/nathelper.html#id294034
Use that fun
Hi George,
If your OpenSIPS fails to reach the UAC is because of two reasons:
- NAT pinhole is closed - but if pinging is done, it shouldn't be
- opensips is trying to contact UAC via wrong IP:port - can you
confirm that when calling the UAC, OpenSIPS sends the INVITE to same IP
and por
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