Re: [OpenSIPS-Users] presence and dns

[Bogdan-Andrei Iancu]

Hi Slava,

1) NOTIFY'es

The presence NOTIFY requests are in dialog requests (part of the 
SUBSCRIBE dialog). And a sequential request is sent to a RR or Contact 
URI (from SUBSCRIBE) which typically hold IP addresses - so how do you 
get the @domain.com URI when sending the NOTIFY ? Do you have a SIP 
capture showing the SUBSCRIBE + 200 OK and the NOTIFY errors ?


2) BLF parsing errors - these are 3.0 specific, right ?

Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
  https://www.opensips-solutions.com
OpenSIPS Summit 2019
  https://www.opensips.org/events/Summit-2019Amsterdam/

On 02/12/2019 05:13 AM, Slava Bendersky wrote:


Hello Everyone,
I manage fix route DNS issue for presence NOTIFY
But I can't route presence NOTIFY with TLS transport on public leg
Text in bold missing ;tranposrt=tls
Do I need add local route and catch NOTIFY and add transport=tls ?

Feb 11 22:03:17 vprx00 /usr/sbin/opensips[1276]: 
INFO:presence:update_subscription: notify
Feb 11 22:03:17 vprx00 /usr/sbin/opensips[1276]: 
INFO:presence:send_notify_request: NOTIFY sip:1...@domain.com.tld via 
*sip:106@public_ip:18156;nat=yes*;ob on behalf of 
sip:1...@domain.com.tld for event presence, 
to_tag=656a154e9c9480ea0e92b7b8c389691c-5e5c, cseq=2
Feb 11 22:03:17 vprx00 /usr/sbin/opensips[1276]: 
INFO:core:probe_max_sock_buff: using snd buffer of 416 kb
Feb 11 22:03:17 vprx00 /usr/sbin/opensips[1276]: 
INFO:core:init_sock_keepalive: TCP keepalive enabled on socket 88
Feb 11 22:03:17 vprx00 /usr/sbin/opensips[1276]: 
ERROR:proto_tls:tls_conn_init: no TLS client domain found
Feb 11 22:03:17 vprx00 /usr/sbin/opensips[1276]: 
ERROR:core:tcpconn_new: failed to do proto 3 specific init for conn 
0x7f2be9aece58
Feb 11 22:03:17 vprx00 /usr/sbin/opensips[1276]: 
ERROR:core:tcp_conn_new: tcpconn_new failed
Feb 11 22:03:17 vprx00 /usr/sbin/opensips[1276]: 
ERROR:proto_tls:tls_sync_connect: tcp_conn_create failed, closing the 
socket
Feb 11 22:03:17 vprx00 /usr/sbin/opensips[1276]: 
ERROR:proto_tls:proto_tls_send: connect failed
Feb 11 22:03:17 vprx00 /usr/sbin/opensips[1276]: ERROR:tm:msg_send: 
send() to public_ip:5081 for proto tls/3 failed
Feb 11 22:03:17 vprx00 /usr/sbin/opensips[1276]: 
ERROR:tm:t_forward_nonack: sending request failed
Feb 11 22:03:17 vprx00 /usr/sbin/opensips[1276]: ERROR:tm:w_t_relay: 
t_forward_nonack failed
Feb 11 22:03:17 vprx00 /usr/sbin/opensips[1280]: 
INFO:core:probe_max_sock_buff: using snd buffer of 416 kb
Feb 11 22:03:17 vprx00 /usr/sbin/opensips[1280]: 
INFO:core:init_sock_keepalive: TCP keepalive enabled on socket 109
Feb 11 22:03:17 vprx00 /usr/sbin/opensips[1277]: 
ERROR:proto_tls:tls_accept: New TLS connection from public_ip:49523 
failed to accept
Feb 11 22:03:17 vprx00 /usr/sbin/opensips[1277]: 
ERROR:proto_tls:tls_read_req: failed to do pre-tls reading
Feb 11 22:03:17 vprx00 /usr/sbin/opensips[1276]: 
WARNING:presence:p_tm_callback: completed with status [500] and to_tag 
[656a154e9c9480ea0e92b7b8c389691c-5e5c], cseq [CSeq: 2]


volga629

*From: *"Slava Bendersky" 
*To: *"OpenSIPS users mailling list" 
*Sent: *Wednesday, February 6, 2019 10:58:38 AM
*Subject: *[OpenSIPS-Users] presence and dns

Hello Everyone,
How to avoid dns look up while NOTIFY is trying to reach end point for 
presence subscriber.


Feb  5 15:15:42 vprx00 /usr/sbin/opensips[3889]: 
INFO:presence:update_subscription: notify
Feb  5 15:15:42 vprx00 /usr/sbin/opensips[3889]: 
INFO:presence:send_notify_request: NOTIFY sip:1...@domain.com via 
sip:106@158.69.151.88:58924;nat=yes;ob on behalf of 
sip:1...@domain.com for event presence, 
to_tag=656a154e9c9480ea0e92b7b8c389691c-0448, cseq=1
Feb  5 15:15:42 vprx00 /usr/sbin/opensips[3889]: 
CRITICAL:core:mk_proxy: could not resolve hostname: "domain.com"
Feb  5 15:15:42 vprx00 /usr/sbin/opensips[3889]: ERROR:tm:uri2proxy: 
bad host name in URI 
Feb  5 15:15:42 vprx00 /usr/sbin/opensips[3889]: 
ERROR:tm:t_forward_nonack: failure to add branches
Feb  5 15:15:42 vprx00 /usr/sbin/opensips[3889]: ERROR:tm:w_t_relay: 
t_forward_nonack failed



Also Pua Dialog Info producing the error of extra parameters

Feb  5 15:42:21 vprx00 /usr/sbin/opensips[4137]: 
ERROR:pua_dialoginfo:pack_cb_params: Failed to parse peer nameaddr 
[sip:105@10.30.100.41:5060#015#012]
Feb  5 15:42:21 vprx00 /usr/sbin/opensips[4137]: 
ERROR:pua_dialoginfo:dialoginfo_set: Failed to allocate parameters


Feb  5 18:27:20 vprx00 /usr/sbin/opensips[5451]: 
ERROR:pua_dialoginfo:pack_cb_params: Failed to parse peer nameaddr 
[sip:033003535...@eficart.colpbx.com#015#012]
Feb  5 18:27:20 vprx00 /usr/sbin/opensips[5451]: 
ERROR:pua_dialoginfo:dialoginfo_set: Failed to allocate parameters



volga629

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Re: [OpenSIPS-Users] Use of multiple outgoing TCP Trunks with B2B

[Bogdan-Andrei Iancu]

Hi Xaled,

First of all, you do not need b2b in order to do TCP conn reusage.

How the conns are managed in OpenSIPS : the conns are reused if the 
destination of the package matches the remote destination of an existing 
TCP conn. IF not, a new conn will be open.


In your case, the first tcp conns will be open when uac_registrant will 
fire the REGISTERs - if each registration/contact has a different 
destination, you will get one conn per contact.


In order to tunnel the INVITEs via the existing conns (open by 
registrations), you need to be sure to route/send the INVITE to the same 
destinations as the uac_registrant did for REGISTERs


Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
  https://www.opensips-solutions.com
OpenSIPS Summit 2019
  https://www.opensips.org/events/Summit-2019Amsterdam/

On 02/12/2019 03:02 PM, xaled wrote:

Hello,

We have a use case with some specific requirements on the TCP connection usage 
by the outgoing register trunk. Requirements are imposed by the service 
provider and cannot be modified by us.

a) Trunk must use TCP.
b) There has to be a registration over an established TCP connection.
c) INVITEs can be send only over established TCP connection that was previously 
used for successful registration.

I managed to get around this limitations by using uac_registrant and b2b 
modules. I also set tcp_connection_lifetime=3600 to have the TCP connection 
open for a pretty long time between possible SIP communications. B2B module 
reuses TCP connection established by uac_registrant and it works so far.

Now there is another limitation on this trunk and it is the number of parallel 
calls. We need more parallel calls then a single trunk is allowed to have. We 
can have additional trunks and multiply the capacity. I added additional trunk 
credentials to the uac_registrant DB and multiple registrations are 
successfully established.

Here come the problems:

1) outgoing INVITE does not reuse any of the established TCP connection. 
Instead the new TCP connection is established.
2) Even if INVITE would reuse one of the established TCP connections the 
credentials used by INVITE have to match the ones that were used during the 
registration.

Is there anything that can be done to correlate TCP connections and credentials 
between uac_registrant and B2B modules?

In our case It would be enough to have a random pick of registered trunk with 
established TCP connection and relevant credentials for every forwarded INVITE 
to use additional capacity given by additional trunks.

Thanks,
Xaled


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Re: [OpenSIPS-Users] locally generated replies

[Pasan Meemaduma via Users]

Hi Bogdan,

Thanks for the reply. writing inline replies


"Using t_on_reply() is harmless as it simply set a holder in the 
msg/transaction, nothing more. And a locally generated reply does NOT trigger 
on_reply_route - this route is triggered only by *RECEIVED* replies."

My problem was that server received 407 replies from its own source ip :/. I 
couldn't figure out what triggered it. Anyhow issue resolved after I removed 
fix_dialog_route() function which causes  BYE to send to a private ip of a 
NATed device. I have fix_nated_contact() before create_dialog() function but 
dialog data seems to have unmodified contact data which I believe is fixed in 
2.4 as you mentioned.

 "So the bottom problem for you is why the processes stuck ? do they stuck like 
consuming cpu (doing some crazy looping) or blocking in some I/O (no CPU usage) 
?"

All traffic processing workers were stuck in a loop and consume 100% CPU. I 
couldn't get much debug info like running opensipsctl trap when issue occured :/

Anyhow now I'm planning to upgrade thee system to 2.4 this year which has 
support till 2021.
my sip cluster is now  stable without fix_dialog_route() function.

On 02/13/2019 09:36 AM, Pasan Meemaduma via Users wrote:


>  
> Hi Guys,
> 
> 
> 
> 
> How's it possible locally generated replied triggering on_reply_route ? I ran 
> in to issue where all opensips process gets stuck in processing same call 
> replies and causing other traffic to get drop.
> 
> 
> 
> 
> 
> 
> 
>  /usr/sbin/opensips[27464]: Call: Reply from a NAT endpoint - S=408 
> D=Request Timeout F=sip:xxx@xxx T=sip:yyy
>  
> yy@x IP=a.b.c.d ID=asgasgasgas
> 
> 
> 
> Request process by opensips before this is an ACK request belong to the call 
> where I don't think It'll expect a reply. Could it be an issue If I call 
> t_on_reply on an ACK msg ?
> 
> 
> 
> 
> I'm trying to figure out where the bug in my opensips routing script.  It 
> causes all sip listerner processes to get stuck in a loop causing to generate 
> above message. IP a.b.c.d is the sip server IP which confuse me as locally 
> generated replies shouldn't trigger on_reply_route as per docs.
> 
> 
> 
> 
> 
> Any clue is welcome.
> 
> 
> 
> 
> I'm using opensips 2.3.6
> 
> 
> 
> 
> 
> 
> 
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Re: [OpenSIPS-Users] tls -> udp

[Bogdan-Andrei Iancu]

Hi John,

For (2), there is nothing special you need to do - the double_rr is by 
default enabled and you just need to do record_route() - opensips is 
automatically figure out that there is a protocol changing and do a 
double RR headers.


Regarding (1) and (3) - the proto info in RURI/destination is overriding 
the info in forced socket - if you have a explicit UDP destination, but 
a TLS socket forced, the socket will be discarded and a proper one (UDP) 
will be found.


Shortly said, you need only (1).

Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
  https://www.opensips-solutions.com
OpenSIPS Summit 2019
  https://www.opensips.org/events/Summit-2019Amsterdam/

On 02/13/2019 11:15 AM, John Quick wrote:

Hi Johan,

I've configured Proxy servers to do this a few times.
You should do the following:
1) For Requests going from TLS to UDP, change any occurrence of
"transport=tls" in the R-URI parameters. I use the following to do this:
subst_uri('/transport=tls/transport=udp/I');

2) Make sure OpenSIPS adds correct Record-Route headers.
Default behaviour in this case is to add 2 RR headers when you call
record_route(). Make sure double_rr has not been disabled in the modparam
section.
One header describes the TLS socket and the other header describes the UDP
socket. These are needed for sequential Loose-Routed requests later in the
dialogue.

3) Just before you relay the request over UDP, call the force_send_socket()
function. For example:
force_send_socket(udp:12.34.56.78:5060);

Hope this helps.

John Quick
Smartvox Limited



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Re: [OpenSIPS-Users] locally generated replies

[Bogdan-Andrei Iancu]

Hi Pasan,

Using t_on_reply() is harmless as it simply set a holder in the 
msg/transaction, nothing more. And a locally generated reply does NOT 
trigger on_reply_route - this route is triggered only by *RECEIVED* replies.


So the bottom problem for you is why the processes stuck ? do they stuck 
like consuming cpu (doing some crazy looping) or blocking in some I/O 
(no CPU usage) ?


Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
  https://www.opensips-solutions.com
OpenSIPS Summit 2019
  https://www.opensips.org/events/Summit-2019Amsterdam/

On 02/13/2019 09:36 AM, Pasan Meemaduma via Users wrote:

Hi Guys,

How's it possible locally generated replied triggering on_reply_route 
? I ran in to issue where all opensips process gets stuck in 
processing same call replies and causing other traffic to get drop.



 /usr/sbin/opensips[27464]: Call: Reply from a NAT endpoint - 
S=408 D=Request Timeout F=sip:xxx@xxx T=sip:yyy

yy@x IP=a.b.c.d ID=asgasgasgas

Request process by opensips before this is an ACK request belong to 
the call where I don't think It'll expect a reply. Could it be an 
issue If I call t_on_reply on an ACK msg ?


I'm trying to figure out where the bug in my opensips routing script.  
It causes all sip listerner processes to get stuck in a loop causing 
to generate above message. IP a.b.c.d is the sip server IP which 
confuse me as locally generated replies shouldn't trigger 
on_reply_route as per docs.


Any clue is welcome.

I'm using opensips 2.3.6


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Re: [OpenSIPS-Users] opensips 2.4.4: bug in tls_mgm

[Vlad Patrascu]

Hi Johan,

The issue is probably caused by the fact that OpenSIPS tries to 
initialize a 'default' client and server domain and the certificate file 
is inexistent at the default path 
('/usr/local/opensips//etc/opensips/tls/cert.pem'). So even if you 
define your "sv_dom" custom server domain, OpenSIPS still tries to 
create default domains for fallback purposes. When you omit the 
'[sv_dom]' prefix for the domain settings in your second test, you 
overwrite the default ones (with a valid path for the certificate this 
time).


In conclusion, either specify your custom domain _and_ default domains 
separately (set the modparams multiple times) or make sure the 
certificate is found at the default path. Also, note that it's not 
necessary to define a custom server domain if you only intend to use a 
single one, as the default will match any socket.


Regards,

Vlad Patrascu
OpenSIPS Developer
http://www.opensips-solutions.com

On 02/17/2019 01:35 PM, johan de clercq wrote:


Hi,

I believe that I have found a bug in tls_mgm:

Using opensips default certificates:

/usr/local/opensips/etc/opensips/tls# ls -lu

total 24

-rw-r--r-- 1 root staff 2049 Feb 17 12:13 ca.conf

-rw-r--r-- 1 root staff 1048 Feb 17 12:13 README

-rw-r--r-- 1 root staff 1127 Feb 17 12:13 request.conf

drwxr-sr-x 4 root staff 4096 Feb 17 12:16 rootCA

drwxr-sr-x 2 root staff 4096 Feb 17 12:13 user

-rw-r--r-- 1 root staff  591 Feb 17 12:13 user.conf

/usr/local/opensips/etc/opensips/tls/rootCA# ls

cacert.pem  certs  index.txt  private serial

Tls params:

loadmodule "tls_mgm.so"

modparam("tls_mgm", "server_domain", "sv_dom=5.135.140.139:5061")

modparam("tls_mgm", "require_cert", "[sv_dom]0")

modparam("tls_mgm", "verify_cert", "[sv_dom]0")

modparam("tls_mgm", "tls_method", "[sv_dom]SSLv23")

modparam("tls_mgm", "certificate", 
"[sv_dom]/usr/local/opensips/etc/opensips/tls/rootCA/cacert.pem")


modparam("tls_mgm", "private_key", 
"[sv_dom]/usr/local/opensips/etc/opensips/tls/rootCA/private/cakey.pem")


modparam("tls_mgm", "ca_list", 
"[sv_dom]/usr/local/opensips/etc/opensips/tls/rootCA/cacert.pem")


 PROTO_TLS module

loadmodule "proto_tls.so"

modparam("proto_tls", "trace_destination", "hep_dest")

modparam("proto_tls", "trace_on", 1)

I removed the passphrase:

mv etc/tls/rootCA/private/cakey.pem 
etc/tls/rootCA/private/cakey.pem.protected


openssl rsa -in etc/tls/rootCA/private/cakey.pem.protected -out 
etc/tls/rootCA/private/cakey.pem


and then tried to run opensips from cmdline : ./opensips -f 
/usr/local/opensips/etc/opensips/opensips.cfg


syslog output:

Feb 17 12:22:01 ns3012072 ./opensips[28673]: 
ERROR:tls_mgm:load_certificate: unable to load certificate file 
'/usr/local/opensips//etc/opensips/tls/cert.pem'


Feb 17 12:22:01 ns3012072 ./opensips[28673]: 
ERROR:tls_mgm:init_tls_domains: Failed to init TLS domain 'default'


Feb 17 12:22:01 ns3012072 ./opensips[28673]: ERROR:core:init_mod: 
failed to initialize module tls_mgm


Feb 17 12:22:01 ns3012072 ./opensips[28673]: ERROR:core:main: error 
while initializing modules


Feb 17 12:22:01 ns3012072 ./opensips[28673]: CRITICAL:core:sig_usr: 
segfault in attendant (starter) process!


Feb 17 12:22:01 ns3012072 kernel: [ 4024.678398] opensips[28673]: 
segfault at 7fcb76dbf850 ip 7fcb76546f69 sp 7ffe803ac150 error 
4 in libcrypto.so.1.1[7fcb763df000+265000]


Next I tried with:

loadmodule "tls_mgm.so"

#modparam("tls_mgm", "server_domain", "sv_dom=5.135.140.139:5061")

modparam("tls_mgm", "require_cert", "0")

modparam("tls_mgm", "verify_cert", "0")

modparam("tls_mgm", "tls_method", "SSLv23")

modparam("tls_mgm", "certificate", 
"/usr/local/opensips/etc/opensips/tls/rootCA/cacert.pem")


modparam("tls_mgm", "private_key", 
"/usr/local/opensips/etc/opensips/tls/rootCA/private/cakey.pem")


modparam("tls_mgm", "ca_list", 
"/usr/local/opensips/etc/opensips/tls/rootCA/cacert.pem")


 PROTO_TLS module

loadmodule "proto_tls.so"

modparam("proto_tls", "trace_destination", "hep_dest")

modparam("proto_tls", "trace_on", 1)

and then opensips starts…

Can you please explain what I am doing wrong ?

cid:F3100D46-F00D-4610-87ED-3E91DA790A82

Johan De Clercq, Managing Director
Democon bvba - Ooigemstraat 41 - 8780 Oostrozebeke

Tel +3256980990– GSM +32478720104



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Re: [OpenSIPS-Users] OpenSIPS 2.4.4, db_postgres.so number of connections

[Bogdan-Andrei Iancu]

Hi Alexey,

Couple of notes on how the DB conns are handled in the OpenSIPS

1) each process has its own connections, even if to the same DB

2) if multiple modules need to connect to the same DB, from the same 
process, the conn is shared (between all modules using the same DB).


Now, depending on the exact opensips revision you have, you may run a 
version with a DB bug in clusterer module (this was fixed in the latest 
versions). Try removing the clusterer module and see if you have one 
connection per process per DB.


Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
  https://www.opensips-solutions.com
OpenSIPS Summit 2019
  https://www.opensips.org/events/Summit-2019Amsterdam/

On 02/26/2019 09:47 AM, Alexey Kazantsev via Users wrote:

Hi list!

OpenSIPS 2.4.4 (x86_64/linux).
CentOS 7.6.1810.

# we increased this value from default "2"
modparam("db_postgres", "max_db_queries", 20)
modparam("db_postgres", "exec_query_threshold", 6)

modparam("acc","db_url", "postgres://user:PaSs@pg-server/db")
modparam("avpops",  "db_url", "postgres://user:PaSs@pg-server/db")
modparam("dispatcher", "db_url", "postgres://user:PaSs@pg-server/db")
modparam("clusterer","db_url", "postgres://user:PaSs@pg-server/db")


My colleague noticed an interesting behavior with PG DB connection.

'netstat -tulpan | grep %pg-server-ip%' showed only 1 ESTABLISHED connection.

When we configured 5432 port explicitly for the 'acc' module (which uses the DB
most actively) ...

modparam("acc", "db_url", "postgres://user:PaSs@pg-server:5432/db")

... we saw several dozens of connections.


And the load on the workers also decreased after that.
Is it a bug?


---
BR, Alexey
http://alexeyka.zantsev.com/
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Re: [OpenSIPS-Users] Presence

[Bogdan-Andrei Iancu]

Hi Mark,

Who is generating the PUBLISH ? the end-device ? or some pua module from 
opensips ?


Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
  https://www.opensips-solutions.com
OpenSIPS Summit 2019
  https://www.opensips.org/events/Summit-2019Amsterdam/

On 02/09/2019 06:06 PM, Mark Thomas wrote:
Alright I've gotten a little farther now with presence. My problem now 
is whenever a publish is being sent out it sends it to DNS and not 
routed using the location table. Any input on this?



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Re: [OpenSIPS-Users] SDP manipulation & rtpengine

[Bogdan-Andrei Iancu]

Hmmm, indeed, passing SDP as variable to rtpengine is possible only with 
3.0 , my badtoo many versions.


This is the commit you are looking for:
https://github.com/OpenSIPS/opensips/commit/a27797d04ba418eb3ac4c2d6dadd0cdf7f3c17b6#diff-069ca6b59f3521936eca7bf3684ae63d

maybe you can grab the module from 3.0 right after that commit and copy 
it in 2.4 tree - it may compile.


Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
  https://www.opensips-solutions.com
OpenSIPS Summit 2019
  https://www.opensips.org/events/Summit-2019Amsterdam/

On 03/06/2019 10:47 PM, Pat Burke wrote:


Thanks Bogdan,


I see that it is part of 3.0, but not a part of 2.4.  Is this a module 
we can pull back to 2.4?



Regards,
*Pat Burke*

Voxtelesys | solutions to grow your business
__
Direct: (402) 403-5121 |   Cell: (402) 443-8929 |   Email: 
p...@voxtelesys.com 

1801 23rd Avenue North |  Suite 217 |  Fargo, North Dakota 58102


-Original Message-
From: "Bogdan-Andrei Iancu" mailto:bog...@opensips.org>>
To: "OpenSIPS users mailling list" mailto:users@lists.opensips.org>>, "Pat Burke"
mailto:p...@voxtelesys.com>>
Date: 03/05/19 09:45
Subject: Re: [OpenSIPS-Users] SDP manipulation & rtpengine

Hi Pat,

What you can do is to grab the SDP from the msg into a variable,
to do whatever fixes/change you have to directly in the variable
and push the body via variable to rtpengine.

Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
   https://www.opensips-solutions.comOpenSIPS Summit 2019
   https://www.opensips.org/events/Summit-2019Amsterdam/

On 02/27/2019 11:30 PM, Pat Burke wrote:


Hello:


I am using trying to manipulate parts of the SDP body before
calling rtpengine_offer / rtpengine_answer. However, any changes
made via textops functions such as subst_body, replace_body,
replace_body_all, etc. do not seem to impact the SDP that is sent
to rtpengine.

In my particular case, rtpengine fails to parse the SDP because
of an extra carriage return line feed sent in the SDP. Is there a
way to send rtpengine manipulated SDP, rather than just the SDP
sent in the request?

*_Use case:_*
if (subst_body("/(^a=.*\r\n)\r\n/\1/g")) {
xlog("L_INFO", "bad SDP --- duplicate CRLF");
}

rtpengine_offer(" ... options ... ");


Regards,
*Pat Burke*

Voxtelesys | solutions to grow your business

__
Direct: (402) 403-5121 | Cell: (402) 443-8929 | Email:
p...@voxtelesys.com 
1801 23rd Avenue North | Suite 217 | Fargo, North Dakota 58102


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[OpenSIPS-Users] [Blog] Achieving service redundancy in two steps with unified clustering in OpenSIPS 3.0

[Bogdan-Andrei Iancu]

A hot backup means redundancy, redundancy means more uptime, more uptime 
means a better SLA, a better SLA means happier customers and more money.


Building redundancy is a must when moving your service into production. 
How can you achieve this with OpenSIPS 3.0? Well, thanks to the new 
unified clustering support, you are only 2 steps away from getting it done.


https://blog.opensips.org/2019/03/07/achieving-service-redundancy-in-two-steps-with-unified-clustering-in-opensips-3-0/

Enjoy it,

--
Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
  https://www.opensips-solutions.com
OpenSIPS Summit 2019
  https://www.opensips.org/events/Summit-2019Amsterdam/


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Re: [OpenSIPS-Users] [RELEASE] OpenSIPS 2.4.5 minor release

[Răzvan Crainea]

Hello to you all!

I am happy to inform you that we've just released OpenSIPS 2.4.5, the 
latest release of the OpenSIPS 2.4 stable branch.
We strongly encourage you to update your setup to the latest release as 
soon as possible, to make sure you get all the bug fixes we've done for 
this version.


Happy hacking!

On 3/4/19 4:38 PM, Răzvan Crainea wrote:

Greetings!

We are glad to announce you that we are planning a new minor release of 
OpenSIPS 2.4.5 for this Thursday, 7th of March 2019.
If there are any outstanding issues that you are facing with OpenSIPS 
2.4.4, please post them asap on our issue tracker.


[1] https://github.com/OpenSIPS/opensips/issues

Happy hacking!


--
Răzvan Crainea
OpenSIPS Core Developer
  http://www.opensips-solutions.com
Meet the OpenSIPS team at the next OpenSIPS Summit:
  https://www.opensips.org/events

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Re: [OpenSIPS-Users] locally generated replies

[Bogdan-Andrei Iancu]

Hi Pasan,

Have you checked if there is traffic on looping on your OpenSIPS ? maybe 
due bad routing, OpenSIPS is keep sending to itself some traffic, 
causing the loops and the load.


Regards,

Bogdan-Andrei Iancu

OpenSIPS Founder and Developer
  https://www.opensips-solutions.com
OpenSIPS Summit 2019
  https://www.opensips.org/events/Summit-2019Amsterdam/

On 03/07/2019 03:17 PM, Pasan Meemaduma wrote:

Hi Bogdan,

Thanks for the reply. writing inline replies


"Using t_on_reply() is harmless as it simply set a holder in the msg/transaction, 
nothing more. And a locally generated reply does NOT trigger on_reply_route - this route 
is triggered only by *RECEIVED* replies."

My problem was that server received 407 replies from its own source ip :/. I 
couldn't figure out what triggered it. Anyhow issue resolved after I removed 
fix_dialog_route() function which causes  BYE to send to a private ip of a 
NATed device. I have fix_nated_contact() before create_dialog() function but 
dialog data seems to have unmodified contact data which I believe is fixed in 
2.4 as you mentioned.

  "So the bottom problem for you is why the processes stuck ? do they stuck like 
consuming cpu (doing some crazy looping) or blocking in some I/O (no CPU usage) ?"

All traffic processing workers were stuck in a loop and consume 100% CPU. I 
couldn't get much debug info like running opensipsctl trap when issue occured :/

Anyhow now I'm planning to upgrade thee system to 2.4 this year which has 
support till 2021.
my sip cluster is now  stable without fix_dialog_route() function.

On 02/13/2019 09:36 AM, Pasan Meemaduma via Users wrote:


   
Hi Guys,





How's it possible locally generated replied triggering on_reply_route ? I ran 
in to issue where all opensips process gets stuck in processing same call 
replies and causing other traffic to get drop.







 /usr/sbin/opensips[27464]: Call: Reply from a NAT endpoint - S=408 
D=Request Timeout F=sip:xxx@xxx T=sip:yyy
   
yy@x IP=a.b.c.d ID=asgasgasgas




Request process by opensips before this is an ACK request belong to the call 
where I don't think It'll expect a reply. Could it be an issue If I call 
t_on_reply on an ACK msg ?




I'm trying to figure out where the bug in my opensips routing script.  It 
causes all sip listerner processes to get stuck in a loop causing to generate 
above message. IP a.b.c.d is the sip server IP which confuse me as locally 
generated replies shouldn't trigger on_reply_route as per docs.





Any clue is welcome.




I'm using opensips 2.3.6







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[OpenSIPS-Users] registrar lookup function

[Pasan Meemaduma via Users]

Hi Guys,
Would someone able to shed some light on the method filtering flag (m) in 
registrar lookup function do  ? I'm having trouble understanding its purpose. 
Can somone give me an example of how it works ?
registrar Module

| 
| 
|  | 
registrar Module


 |

 |

 |




Thank you.
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