Re: [OpenSIPS-Users] Drouting memory usage

[John Nash]

Actually crash happened shortly after we uploaded 11000 codes but looks
like it is not related to drouting. I see following message

CRITICAL:topology_hiding:build_encoded_contact_suffix: #012>>> pkg_free()
on non-pkg ptr 0x18 - aborting!#012#012It seems you have hit a programming
bug.#012Please help us make OpenSIPS better by reporting it at
https://github.com/OpenSIPS/opensips/issues#012

In log file I see following messages time to time
ERROR:core:pv_get_contact_body: failed to parse contact hdr

On Wed, May 11, 2016 at 11:29 PM, John Nash  wrote:

> I have been using drouting module with just 200 entries from 8 months
> yesterday we had need of adding around 11000 entries in rules table but
> after that opensips started to crash. I am currently using -m 2048 -M 1024
> isn't it enough memory?
>
> How can I anticipate memory usage?
>
> John
>
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[OpenSIPS-Users] Drouting memory usage

[John Nash]

I have been using drouting module with just 200 entries from 8 months
yesterday we had need of adding around 11000 entries in rules table but
after that opensips started to crash. I am currently using -m 2048 -M 1024
isn't it enough memory?

How can I anticipate memory usage?

John
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Re: [OpenSIPS-Users] Ubuntu 16.04 can no longer access trusty packages at apt.opensips.org

[Nick Altmann]

Hi, Jeff

I’ve changed GPG key to more secure and also I’ve added xenial builds.
Please, test.
We are not watching for new releases of ubuntu, but always glad to add them
by request.

--
Nick

2016-05-01 2:23 GMT+02:00 Jeff Pyle :

> Hello,
>
> First off, thanks to everyone who makes this repo possible.
>
> Ubuntu 15.10 works just fine with the trust repo at apt.opensips.org.
>  16.04, however, disables support for SHA1:
>
> W: http://apt.opensips.org/dists/trusty/Release.gpg: Signature by key
> 33FED9119AC17EB465F51BF001839BE75F2FBB7C uses weak digest algorithm (SHA1)
> E: Failed to fetch http://apt.opensips.org/dists/trusty/Release  No Hash
> entry in Release file
> /var/lib/apt/lists/partial/apt.opensips.org_dists_trusty_Release which is
> considered strong enough for security purposes
> E: Some index files failed to download. They have been ignored, or old
> ones used instead.
>
>
> I've tried disabling GPG checks with "APT::Get::AllowUnauthenticated
> "true";" and the like but it doesn't seem to help.
>
> Now that 16.04 is out, and it's LTS, I was curious when we might see a
> xenial repo available.
>
>
> - Jeff
>
>
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>
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Re: [OpenSIPS-Users] Re-Invite routed to private IP

[Barış ŞEKERCİLER]

You could click "unsubscribe" button.
Why are you swearing?

> On 29 Apr 2016, at 19:21, Ali Ariyayi  wrote:
> 
> fuck you
> fuck you
> 
> do not send mail for me
> 
> fuck you
> 
> fuck your parents 
> 
> sun of the bech
> 
> 
> On Friday, April 29, 2016 8:49 PM, Ali Ariyayi  wrote:
> 
> 
> fuck you
> fuck you
> 
> do not send mail for me
> 
> fuck you
> 
> fuck your parents 
> 
> sun of the bech
> 
> 
> On Friday, April 29, 2016 8:48 PM, Ali Ariyayi  wrote:
> 
> 
> fuck you
> fuck you
> 
> do not send mail for me
> 
> fuck you
> 
> fuck your parents 
> 
> sun of the bech
> 
> 
> On Friday, April 29, 2016 6:56 PM, Julian Santer  
> wrote:
> 
> 
> Ok. I read the OpenSips doc of force_rport again:
> Force_rport() adds the rport parameter to the first Via header. Thus, 
> OpenSIPS will add the received IP port to the top most via header in the SIP 
> message, even if the client does not indicate support for rport. This enables 
> subsequent SIP messages to return to the proper port later on in a SIP 
> transaction.
> 
> I misunderstand th "received IP port" maybe ;-)
> 
> But we didn't have this problems before changing the platform.
> 
> The Re-Invite enters here, and as I saw in syslog, the topology_hiding 
> matches and also the uac_test is be done:
> 
> if (has_totag())
> {
> if (topology_hiding_match())
> {
> if (nat_uac_test("127"))
> {
> xlog("L_INFO", "Topology hided, Contact fixed - LF_BASE");
> fix_nated_contact();
> }
> 
> t_relay();
> exit;
> }
> 
> Am 29.04.2016 um 15:56 schrieb Johan De Clercq:
> > I don;t think so : force_rport just adds the port on which you receive to 
> > the first via header.
> >
> > 2016-04-29 15:36 GMT+02:00 Julian Santer  > 
> > >>:
> >
> >We are using 2.1.2
> >So we made a "huge" version update and also changed the kind of working
> >
> >Am 29.04.2016 um 14:27 schrieb Johan De Clercq:
> >
> >What version do you use in your new install ?
> >
> >2016-04-29 13:12 GMT+02:00 Julian Santer  >   > >
> > 
> >  >
> >  Hi Johan,
> >
> >  we changed our platform 2 weeks ago from 1 OpenSips Proxy 
> > 1.6.4 to 3 OpenSips instances with topology hiding.
> >  Since we didn't have this problems earlier, I think it worked 
> > with the old platform.
> >  As we call force_rport() in the main_route, the private IP in 
> > the VIA header should be ignored, right?
> >  At the moment we call force_rport() in all our instances.
> >  I think, we should call force_rport() only on the edge server 
> > where we make the nat_handling, right?
> >
> >  Kind regards,
> >  Julian Santer
> >  Raiffeisen OnLIne
> >
> >
> >  Am 29.04.2016 um 10:10 schrieb Johan De Clercq:
> >
> >  Indeed.
> >
> >
> >  2016-04-29 9:49 GMT+02:00 Julian Santer 
> >  
> > >
> > 
> > >>
> >   >   > >  > 
> > > :
> >
> >  Hi Johan,
> >
> >  as the asterisk is not administrated by us, I have to 
> > ask the customer.
> >  As I understand you think the problem should be the 
> > private IP in the VIA header
> >  and this should be fixed with STUN, right?
> >
> >  Kind regards,
> >  Julian Santer
> >  Raiffeisen OnLine
> >
> >  Am 29.04.2016 um 09:17 schrieb Johan De Clercq:
> >
> >  do you use stun on the phones ?
> >
> >  2016-04-29 9:14 GMT+02:00 Julian Santer 
> > 
> > > >  > 
> >

Re: [OpenSIPS-Users] sipsak and sip_trace module

[Alain Bieuzent]

Hi,

Add this lines before load_balance test :

if (is_method("OPTIONS"))
{
sl_send_reply("200", "OK");
exit;
}


Regards




Le 10/05/2016 16:37, « Francjos »  a écrit :

>hello ,
>
>I wish 
>
>I'm using two opensips servers in redundancy mode, i would like one  be able
>to check if the other is still aliveand vice versa. I use a sipsak to
>perform this.
>But when i execute the commande  "sipsak -s sip:opensips@ where xxx in
>the ip address of the other server, i get the following:
>
>
>Server 10.0.0.5 gives:
>.
>
>
>U 2016/05/10 13:57:12.402720 10.0.0.5:47413 -> 10.0.0.4:5060
>OPTIONS sip:opensips@10.0.0.4 SIP/2.0.
>Via: SIP/2.0/UDP 10.0.0.5:47413;branch=z9hG4bK.3fc50b66;rport;alias.
>From: sip:sipsak@10.0.0.5:47413;tag=4c5c78b9.
>To: sip:opensips@10.0.0.4.
>Call-ID: 1281128633@10.0.0.5.
>CSeq: 1 OPTIONS.
>Contact: sip:sipsak@10.0.0.5:47413.
>Content-Length: 0.
>Max-Forwards: 70.
>User-Agent: sipsak 0.9.6.
>Accept: text/plain.
>.
>
>
>U 2016/05/10 13:57:12.405823 10.0.0.4:5060 -> 10.0.0.5:47413
>SIP/2.0 500 Service full.
>Via: SIP/2.0/UDP
>10.0.0.5:47413;received=10.0.0.5;branch=z9hG4bK.3fc50b66;rport=47413;alias.
>From: sip:sipsak@10.0.0.5:47413;tag=4c5c78b9.
>To: sip:opensips@10.0.0.4;tag=61890dad1e908c702027bf054a266115.c3bd.
>Call-ID: 1281128633@10.0.0.5.
>CSeq: 1 OPTIONS.
>Server: OpenSIPS (2.1.2 (x86_64/linux)).
>Content-Length: 0.
>.
>
>
>
>
>Server 10.0.0.4 gives:
>
>
>
>
>root@front-1:/etc/opensips# ngrep -d eth0 -t -W byline "$1" port 5060 -q
>interface: eth0 (10.0.0.0/255.255.255.0)
>filter: (ip or ip6) and ( port 5060 )
>
>U 2016/05/10 13:57:12.406208 10.0.0.5:47413 -> 10.0.0.4:5060
>OPTIONS sip:opensips@10.0.0.4 SIP/2.0.
>Via: SIP/2.0/UDP 10.0.0.5:47413;branch=z9hG4bK.3fc50b66;rport;alias.
>From: sip:sipsak@10.0.0.5:47413;tag=4c5c78b9.
>To: sip:opensips@10.0.0.4.
>Call-ID: 1281128633@10.0.0.5.
>CSeq: 1 OPTIONS.
>Contact: sip:sipsak@10.0.0.5:47413.
>Content-Length: 0.
>Max-Forwards: 70.
>User-Agent: sipsak 0.9.6.
>Accept: text/plain.
>.
>
>
>U 2016/05/10 13:57:12.406735 10.0.0.4:5060 -> 10.0.0.5:47413
>SIP/2.0 500 Service full.
>Via: SIP/2.0/UDP
>10.0.0.5:47413;received=10.0.0.5;branch=z9hG4bK.3fc50b66;rport=47413;alias.
>From: sip:sipsak@10.0.0.5:47413;tag=4c5c78b9.
>To: sip:opensips@10.0.0.4;tag=61890dad1e908c702027bf054a266115.c3bd.
>Call-ID: 1281128633@10.0.0.5.
>CSeq: 1 OPTIONS.
>Server: OpenSIPS (2.1.2 (x86_64/linux)).
>Content-Length: 0.
>
>
>
>I guess the error "500 Service full" is related to a bloc in the script
>where i define the routing logic for the load balancing:
>
>
>
>
>
>## Freeswitch
>route[2] {
>
>xlog("L_NOTICE","[$pr:$fU@$si:$sp]: This is Media-Server Route Use
>Load-balancer NOW!!\n");
> if (!load_balance("1","calls")) {
> sl_send_reply("500","Service full");
>exit;
>}
>xlog("L_NOTICE","[$pr:$fU@$si:$sp]: Selected destination
>Media-Server : $du\n");
>}
>
>In need more explanation how i can fix that.
>Is sipsak a tool of checking the status of the servers or are there othor
>tools?
>
>My end goal is to put in place high availability with the two servers.
>Ideas please!
>
>
>
>
>--
>View this message in context: 
>http://opensips-open-sip-server.1449251.n2.nabble.com/sipsak-and-sip-trace-module-tp7602933.html
>Sent from the OpenSIPS - Users mailing list archive at Nabble.com.
>
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Re: [OpenSIPS-Users] sipsak and sip_trace module

[Bogdan-Andrei Iancu]

Change your routing script to be sure that the OPTIONS request does not 
hit the load balancing logic. Maybe you should reply to it using the 
options module:

http://www.opensips.org/html/docs/modules/2.1.x/options.html#id293442

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 11.05.2016 10:31, Nduwayezu, Joselyne wrote:

Hello Bogdan,

Thank you for your reply. How can i fix it then?

NDUWAYEZU Joselyne

2016-05-11 9:28 GMT+02:00 Bogdan-Andrei Iancu >:


Hi,

Your OPTIONS hits the load_balance() function which returns false
in this case as only INVITEs are supposed to be handled by this
function.

Best regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 10.05.2016 17:37, Francjos wrote:

hello ,

I wish

I'm using two opensips servers in redundancy mode, i would
like one  be able
to check if the other is still aliveand vice versa. I use a
sipsak to
perform this.
But when i execute the commande  "sipsak -s sip:opensips@
where xxx in
the ip address of the other server, i get the following:


Server 10.0.0.5 gives:
.


U 2016/05/10 13:57:12.402720 10.0.0.5:47413
 -> 10.0.0.4:5060 
OPTIONS sip:opensips@10.0.0.4 
SIP/2.0.
Via: SIP/2.0/UDP
10.0.0.5:47413;branch=z9hG4bK.3fc50b66;rport;alias.
From: sip:sipsak@10.0.0.5:47413;tag=4c5c78b9.
To: sip:opensips@10.0.0.4 .
Call-ID: 1281128633@10.0.0.5 .
CSeq: 1 OPTIONS.
Contact: sip:sipsak@10.0.0.5:47413
.
Content-Length: 0.
Max-Forwards: 70.
User-Agent: sipsak 0.9.6.
Accept: text/plain.
.


U 2016/05/10 13:57:12.405823 10.0.0.4:5060
 -> 10.0.0.5:47413 
SIP/2.0 500 Service full.
Via: SIP/2.0/UDP

10.0.0.5:47413;received=10.0.0.5;branch=z9hG4bK.3fc50b66;rport=47413;alias.
From: sip:sipsak@10.0.0.5:47413;tag=4c5c78b9.
To: sip:opensips@10.0.0.4

;tag=61890dad1e908c702027bf054a266115.c3bd.
Call-ID: 1281128633@10.0.0.5 .
CSeq: 1 OPTIONS.
Server: OpenSIPS (2.1.2 (x86_64/linux)).
Content-Length: 0.
.




Server 10.0.0.4 gives:




root@front-1:/etc/opensips# ngrep -d eth0 -t -W byline "$1"
port 5060 -q
interface: eth0 (10.0.0.0/255.255.255.0
)
filter: (ip or ip6) and ( port 5060 )

U 2016/05/10 13:57:12.406208 10.0.0.5:47413
 -> 10.0.0.4:5060 
OPTIONS sip:opensips@10.0.0.4 
SIP/2.0.
Via: SIP/2.0/UDP
10.0.0.5:47413;branch=z9hG4bK.3fc50b66;rport;alias.
From: sip:sipsak@10.0.0.5:47413;tag=4c5c78b9.
To: sip:opensips@10.0.0.4 .
Call-ID: 1281128633@10.0.0.5 .
CSeq: 1 OPTIONS.
Contact: sip:sipsak@10.0.0.5:47413
.
Content-Length: 0.
Max-Forwards: 70.
User-Agent: sipsak 0.9.6.
Accept: text/plain.
.


U 2016/05/10 13:57:12.406735 10.0.0.4:5060
 -> 10.0.0.5:47413 
SIP/2.0 500 Service full.
Via: SIP/2.0/UDP

10.0.0.5:47413;received=10.0.0.5;branch=z9hG4bK.3fc50b66;rport=47413;alias.
From: sip:sipsak@10.0.0.5:47413;tag=4c5c78b9.
To: sip:opensips@10.0.0.4

;tag=61890dad1e908c702027bf054a266115.c3bd.
Call-ID: 1281128633@10.0.0.5 .
CSeq: 1 OPTIONS.
Server: OpenSIPS (2.1.2 (x86_64/linux)).
Content-Length: 0.



I guess the error "500 Service full" is related to a bloc in
the script
where i define the routing logic for the load balancing:





## Freeswitch
route[2] {

 xlog("L_NOTICE","[$pr:$fU@$si:$sp]: This is
Media-Server Route Use
Load-balancer NOW!!\n");
  if (!load_balance("1","calls")) {
  sl_send_reply("500","Service full");
 exit;
 }
 xlog("L_NOTICE","[$pr:$fU@$si:$sp]: Selected destination
Media-Server : $du\n");
}

In need more explanation how i can fix that.
Is sipsak a tool of checking the status of the servers or are
  

Re: [OpenSIPS-Users] sipsak and sip_trace module

[Bogdan-Andrei Iancu]

Hi,

Your OPTIONS hits the load_balance() function which returns false in 
this case as only INVITEs are supposed to be handled by this function.


Best regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com

On 10.05.2016 17:37, Francjos wrote:

hello ,

I wish

I'm using two opensips servers in redundancy mode, i would like one  be able
to check if the other is still aliveand vice versa. I use a sipsak to
perform this.
But when i execute the commande  "sipsak -s sip:opensips@ where xxx in
the ip address of the other server, i get the following:


Server 10.0.0.5 gives:
.


U 2016/05/10 13:57:12.402720 10.0.0.5:47413 -> 10.0.0.4:5060
OPTIONS sip:opensips@10.0.0.4 SIP/2.0.
Via: SIP/2.0/UDP 10.0.0.5:47413;branch=z9hG4bK.3fc50b66;rport;alias.
From: sip:sipsak@10.0.0.5:47413;tag=4c5c78b9.
To: sip:opensips@10.0.0.4.
Call-ID: 1281128633@10.0.0.5.
CSeq: 1 OPTIONS.
Contact: sip:sipsak@10.0.0.5:47413.
Content-Length: 0.
Max-Forwards: 70.
User-Agent: sipsak 0.9.6.
Accept: text/plain.
.


U 2016/05/10 13:57:12.405823 10.0.0.4:5060 -> 10.0.0.5:47413
SIP/2.0 500 Service full.
Via: SIP/2.0/UDP
10.0.0.5:47413;received=10.0.0.5;branch=z9hG4bK.3fc50b66;rport=47413;alias.
From: sip:sipsak@10.0.0.5:47413;tag=4c5c78b9.
To: sip:opensips@10.0.0.4;tag=61890dad1e908c702027bf054a266115.c3bd.
Call-ID: 1281128633@10.0.0.5.
CSeq: 1 OPTIONS.
Server: OpenSIPS (2.1.2 (x86_64/linux)).
Content-Length: 0.
.




Server 10.0.0.4 gives:




root@front-1:/etc/opensips# ngrep -d eth0 -t -W byline "$1" port 5060 -q
interface: eth0 (10.0.0.0/255.255.255.0)
filter: (ip or ip6) and ( port 5060 )

U 2016/05/10 13:57:12.406208 10.0.0.5:47413 -> 10.0.0.4:5060
OPTIONS sip:opensips@10.0.0.4 SIP/2.0.
Via: SIP/2.0/UDP 10.0.0.5:47413;branch=z9hG4bK.3fc50b66;rport;alias.
From: sip:sipsak@10.0.0.5:47413;tag=4c5c78b9.
To: sip:opensips@10.0.0.4.
Call-ID: 1281128633@10.0.0.5.
CSeq: 1 OPTIONS.
Contact: sip:sipsak@10.0.0.5:47413.
Content-Length: 0.
Max-Forwards: 70.
User-Agent: sipsak 0.9.6.
Accept: text/plain.
.


U 2016/05/10 13:57:12.406735 10.0.0.4:5060 -> 10.0.0.5:47413
SIP/2.0 500 Service full.
Via: SIP/2.0/UDP
10.0.0.5:47413;received=10.0.0.5;branch=z9hG4bK.3fc50b66;rport=47413;alias.
From: sip:sipsak@10.0.0.5:47413;tag=4c5c78b9.
To: sip:opensips@10.0.0.4;tag=61890dad1e908c702027bf054a266115.c3bd.
Call-ID: 1281128633@10.0.0.5.
CSeq: 1 OPTIONS.
Server: OpenSIPS (2.1.2 (x86_64/linux)).
Content-Length: 0.



I guess the error "500 Service full" is related to a bloc in the script
where i define the routing logic for the load balancing:





## Freeswitch
route[2] {

 xlog("L_NOTICE","[$pr:$fU@$si:$sp]: This is Media-Server Route Use
Load-balancer NOW!!\n");
  if (!load_balance("1","calls")) {
  sl_send_reply("500","Service full");
 exit;
 }
 xlog("L_NOTICE","[$pr:$fU@$si:$sp]: Selected destination
Media-Server : $du\n");
}

In need more explanation how i can fix that.
Is sipsak a tool of checking the status of the servers or are there othor
tools?

My end goal is to put in place high availability with the two servers.
Ideas please!




--
View this message in context: 
http://opensips-open-sip-server.1449251.n2.nabble.com/sipsak-and-sip-trace-module-tp7602933.html
Sent from the OpenSIPS - Users mailing list archive at Nabble.com.

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[OpenSIPS-Users] APT and YUM repositories important news (GPG changed)

[Nick Altmann]

Hello everyone!

GPG key signing RPM and APT repositories has changed from yesterday to
increase the security.

The new key is *0x049AD65B*.

For YUM users:
1. Remove old opensips-yum releases or nightly package:
*> *rpm -qa|grep opensips-yum|xargs yum -y erase
2. Add it again from yum repository web page:
http://yum.opensips.org/packages.php
3. Accept the new key 0x049AD65B.

For APT users:
Add new key.
> apt-key adv --keyserver keyserver.ubuntu.com --recv-keys 049AD65B

You may also expect some problems accessing APT repository during this day.

Sorry for the inconvenience.

--
Nick
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