Hey Razvan,
I tried the following on Opensips:
subscribe_event("E_ACC_EVENT","
rabbitmq:vhostuser:p...@host.cloudamqp.com/vhostuser/queuename");
their URL string is in this format :
amqp://vhostusert:p...@host.cloudamqp.com/vhostuser
When Opensips tries to connect to this queue it sends
/
Hello,
I have problem with "topology_hiding_match()" and WSS.
Zoiper ---(send BYE)--> Opensips --(can not relay to)---> SIP.JS
Here part ot script:
if (has_totag()) {
if (topology_hiding_match()) {
t_relay();
exit;
}
...
Opensips receive BYE. When execute "t_relay()" give follow ERRORS:
INFO:
Robert, what Bogdan says is essentially correct. The OpenSIPS itself is not
using any threads AFAIK, therefore python module code is kept as simple as
possible. Now back to the original question: we use quite a lot of python
code in our routing and some of the python modules that are running are
ac
Hi Achintha,
If you are using usrloc with DB persistence and you reloaded / restarted
your OpenSIPS, this is a known issue (ticket #1094 [1]). The logs look
almost identical - let me know if this is not the case, and we'll take a
better look at it.
Regards,
[1]: https://github.com/OpenSIPS/
I agree with Ben Newlin.
Pradee – are you trying to simulate a ‘broken call’? Eg one that stops working,
as opposed to a BYE indicating that it has ended ‘cleanly’?
Ben Cropley
Sent from my Windows 10 phone
From: Newlin, Ben
Sent: 21 April 2017 12:50
To: OpenSIPS users mailling list
Subject: R
What you are asking to do breaks the protocol rules of SIP. Neither Opensips
nor any other SIP proxy will do it.
I am not sure which you actually wanted as the result as CANCEL and 480 are not
the same thing, but you cannot convert a BYE into either one.
A CANCEL has to refer to an in-progress
Hi Peter,
In TCP, you cannot control the port used to fire a new TCP connection -
this is selected by the kernel. Still, the UAS side will reply back to
the originating port (as the trace shows). The fact the uac_registrant
does not see the 401 is not necessarily because of the TCP connection
Hi,
I'm using opensip as a load balancer in front of Freeswitch nodes.
Opensips Version 1.11 is used for a reason and can't change the version.
When an INVITE comes from mobile to Freeswitch through Opensips, call will
be answered. Then at the end of call, Freeswitch sends BYE to Opensips.
I wa
Hi,
I'm using opensip as a load balancer in front of Freeswitch nodes.
Opensips Version 1.11 is used for a reason and can't change the version.
When an INVITE comes from mobile to Freeswitch through Opensips, call will
be answered. Then at the end of call, Freeswitch sends BYE to Opensips.
I
hi all,
i configured opensips 2.3 beta on centos 7
then i configured mid-registrar module and freeswitch server as main
registrar.
first user registered properly but opensips is crashed when second user
registration with following console-log
Apr 21 08:29:41 [17198] DBG:core:udp_read_re
thank you, i added this to my opensips.cfg file and it started successfully.
Lets see if it works.
From: "Nabeel"
To: "users"
Sent: Friday, April 21, 2017 2:23:52 AM
Subject: Re: [OpenSIPS-Users] Ghost calls 1001
In case the call is attempted via your server, you can add the following to
Sending a 200 ok will notify the hacker that a sip server exists on the
IP/port, simply ignoring the request is best.
On Apr 21, 2017 12:20 PM, "johan de clercq" wrote:
> Another approach is sending 200 ok and then exit().
>
>
>
> *From:* Users [mailto:users-boun...@lists.opensips.org] *On Behal
Hi, Tito!
I've just made a free cloudamqp account for testing and used the new
rabbitmq module to send a message in the queue. The message was not
published initially due to the fact that I was using the "immediate"
flag, (I was receiving NOT_IMPLEMENTED, probably because the free
account lac
In case the call is attempted via your server, you can add the following to
opensips.cfg to block sip scanners:
if($ua=~"friendly-scanner") {
xlog("L_ERROR", "Auth error for $fU@$fd from $si method $rm
user-agent (friendly-scanner)\n");
drop();
exit;
}
if($ua=~"sipv
Hi Bogdan,
Yes you are correct, it is a retransmission.
I've specifed port 5060 as the forced_socket in the registrant table but
opensips (1.2.3.4 below) doesn't use it. Is there a way to force
uac_registrant to use 5060, or alternatively make the uac_registrant module
listen for replies on whate
Another approach is sending 200 ok and then exit().
From: Users [mailto:users-boun...@lists.opensips.org] On Behalf Of Schneur
Rosenberg
Sent: Friday, April 21, 2017 11:00 AM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] Ghost calls 1001
User agent variable is stored in
User agent variable is stored in $ua do a if and drop()
Regarding iptables do something like this
https://community.freepbx.org/t/stop-sipvicious-friendly-scanner/28580
On Apr 21, 2017 10:12 AM, "Uzair Hassan" wrote:
> Is there any documentation I could read to understand the process you jus
Hi Pete,
Looking to the logs, I see only one line :
Apr 21 09:58:56 dev01 /usr/local/opensips_2.2/sbin/opensips[4766]:
DBG:uac_registrant:reg_tm_cback: tm [0x7f2ceb70b2f0] notification cb for
FAKED_REPLY [408] reply at [1492761536]
The "reg_tm_cback" is the TM callback used to report back to
Hi Rodrigo,
After doing the lookup(location), you can do :
if ( isbflagset("NAT") ) {}
Best regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
OpenSIPS Summit May 2017 Amsterdam
http://www.opensips.org/events/Summit-2017Amsterdam.html
On 04
Hi Marty,
As a high level blueprint, it looks ok - I suggest using in OpenSIPs the
new FreeSWITCH integration for better load balancing :
https://blog.opensips.org/2017/03/01/freeswitch-driven-routing-in-opensips-2-3/
and using topo hiding in OpenSIPS to avoid exposing the FS IP addresses.
Be
Hi Robert,
The only question I can answer is 1) - OpenSIPS it is a multi-process
application (and not using threads).
How the python module is design (from threading perspective), I do not
know - maybe Maxim, the author of this module can help with this.
Regards,
Bogdan-Andrei Iancu
Open
As an extra note here . When logging via syslog, the time and pid part
are formated by syslog itself. When logging to stderr, opensips emulates
the syslog formating for consistency reasons.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
OpenS
Hi Sasmita,
Sharing the contacts themselves is not really enough - as you have WS
endpoints, they use TCP connections, which can be initiated by end-point
only (the registrar cannot). So if an endpoint registers with R1 server
and, sharing the contacts, the R2 server tries to deliver a call to
Hello,
I am trying to use the uac_registrant module (2.2) to register via TCP but
when it recieves the 401 it is not adding the Authorization header.
It does work with UDP so I get:
REGISTER -> 401 (with WWW-Authenticate) -> REGISTER (with Authorization) ->
200 OK
However when I tell it to use
Hi Søren,
The drop stats are incremented if one of the modules (like B2B)
indicates that the SIP message should be discarded (before getting into
the script). Maybe your messages look like B2B related, but the B2B
cannot find the matching sessions ? Could you provide more info on how
you get
Hi, Dragomir!
Most likely OpenSIPS B is too old and cannot parse WSS transport.
You could sort this out by doing topology hiding on OpenSIPS A.
Best regards,
Răzvan Crainea
OpenSIPS Solutions
www.opensips-solutions.com
On 04/20/2017 11:52 PM, Dragomir Haralambiev wrote:
Hello,
I make tes
Is there any documentation I could read to understand the process you just
described?
On April 20, 2017 11:15:54 PM Schneur Rosenberg
wrote:
In addition to iptables/fail2ban you should inspect the useragent that the
packets come from, most of them will come from sip vicious or friendly
sca
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