ent from Outlook for Android
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>
> Sent from Outlook for Android
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Hello,
Port spaces for UDP and TCP are independent.
-- Alex
> On Apr 4, 2024, at 6:45 AM, Sasmita Panda wrote:
>
> Hi All ,
>
> Is the below socket definition right ? In this case opensips will listen on
> both UDp and TCP protocol on the same 5507 port ?
>
>
ine is required?
>
>
> --
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> B.Prathibha
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Although I don't see a need to actually replace the diameter you might want
to consider that this is kind of bleeding edge stuff and the more choices
you have for interfacing the better.
The development and maturity of tools available is very fast and fluid and
in the future there might actually
Hi,
1) Neither Kamailio nor OpenSIPS send 200 OKs;
2) Neither Kamailio nor OPenSIPS send ACKs.
They merely relay these.
3) Contact URI alterations may be occurring along the chain, and are likely
causing your issue.
-- Alex
> On 9 Nov 2023, at 09:12, Social Boh wrote:
>
> Hello
藍藍藍
Original message From: johan Date: 11/16/22
11:03 AM (GMT-06:00) To: users@lists.opensips.org Subject: Re:
[OpenSIPS-Users] OpenSIPS Summit 2023 in USA - is Houston
isn't it : Houston, we have a problem :-)
On 16/11/2022 16:01, Bogdan-Andrei
Iancu
/drive.google.com/file/d/1bpwmXCB6qRxbDk8KZ6GuCg3-hwfABvwk/view?usp=sharing
вт, 23 авг. 2022 г. в 12:44, Bogdan-Andrei Iancu :
> Hi Alex,
>
> Have you tried something like this (for calls from Internet) :
>
> b2b_server_new("server1",$avp(b2b_hdrs), $avp(b2b_hdr_bodie
Hello,
setup mid registrar easy with this instruction.
https://www.opensips.org/Documentation/Tutorials-MidRegistrar
There is link to config
But It relays every Registration request every 60sec on my server. Please
let me know if you could set up like on picture in instruction.
чт, 18 авг.
Hello,
My opensips is behind NAT in cloud
internet - (1.1.1.1)Cone_NAT - port_5060__10.130.0.23(opensips)__port_5070
- main_registrar_sip_server
So I made 2 sockets
socket=udp:10.130.0.23:5070 # for LAN
socket=udp:10.130.0.23:5060 as 1.1.1.1:5060 #for Internet
I try to make call from lan to
> On Mar 15, 2022, at 10:13 AM, Saint Michael wrote:
>
> Every call that goes through Opensips should generate a record.
Is this just your opinion, or…?
--
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Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
Web: http://www.evaris
Via: SIP/2.0/UDP
10.193.169.56;received=10.193.169.56;rport=5060;branch=z9hG4bKycrgiemh
Max-Forwards: 69
To:
From: "Alex" ;tag=pydzo
Call-ID: vfytldehdpikbnt@ryzing
[Generated Call-ID: vfytldehdpikbnt@ryzing]
CSeq: 224 INVITE
Hi Ben,
That's very interesting. I'd read that the Contact address must be a public
address - but perhaps that's not true with double record-route.
First I'll try not rewriting the Contact and then look into TH or B2BUA.
Many thanks for your help!
Alex
From
902e;rport
Max-Forwards: 70
Route:
Route:
Contact:
To: "Alex" ;tag=lihnv
From: ;tag=f6ce187d
Call-ID: mpytwgabkcpnkef@ryzing
[Generated Call-ID: mpytwgabkcpnkef@ryzing]
CSeq: 2 BYE
User-Agent: 3CXPho
nds after the a leg hangs up.
>>
>> Does that make sense? If so, can someone point me in the right direction?
>>
>> Thank you,
>> Alex
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>
fix.
Sorry, me again - $socket_in(af) is working as expected in nightly 3.1
(thanks again!) but there now appears to be another issue whereby $json is
outputting $socket_in(port) as a string instead of numeric? It's numeric as
expected if I go back to 3.1.1
>
Thanks Liviu :)
Alex
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Hi Liviu,
Just found another similar related issue so thought I'd pop it on the same
thread.
$socket_in(af) returns INET or INET6 as expected but after passing through
$json it's numeric and 0 in both cases.
Redacted examples below. I'm on 3.1.1 from Debian/Ubuntu packages.
Thanks,
Alex
Feb
On Tue, 19 Jan 2021 at 17:17, Liviu Chircu wrote:
> For 3.1, I think we're still in the right time window to backport this
> fix without breaking any opensips.cfg files.
I'm using 3.1 so quite happy with that proposal :) Thanks for the fast
response and fix, much appreciated.
a redacted):
Jan 18 18:18:15 [33] request: INVITE from sip:XX@XX to sip:XX@XX
Jan 18 18:18:15 [33] Sending { "call_id": "942887463-1604195939-951239259",
"ts": "2021-01-18 18:18:15Z", "src_ip": "XX", "dst_ip&qu
Thank you for all the pointers, Donat
On Dec 2, 2020, 5:10 AM, at 5:10 AM, Donat Zenichev
wrote:
>Hello Alex.
>Firstly I would like to mention that t_on_failure() is only to be used
>when
>either of this is true:
>- receiving of a negative reply that completes the transaction
Hi,
I have been trying to find a solution for a particular scenario:
Opensips sends INVITE to a carrier
Carrier responds 180 Ringing (no SDP)
Carrier responds 200OK with SDP
Assuming that there are lines in SDP that are not desirable:
a) IS there any legal way to reject this call before
Bump on this one
On Sep 29, 2020, 3:15 PM, at 3:15 PM, Alex A
wrote:
>Hi Everyone,
>
>
>
>I am running into an issue getting the restart_persistency working on
>CentOS Linux release 7.7.1908
>
>Was hoping to get any pointers to further troubleshooting.
>
>
>
&
Hey,
been pulling my remaining 4 hairs out over the this one in past few days...
:)
Anyone can make any suggestions to tackle it?
Thank you in advance.
On Tue, 29 Sep 2020 15:15:36 -0400 Alex A
wrote
Hi Everyone,
I am running into an issue getting
Hi Everyone,
I am running into an issue getting the restart_persistency working on CentOS
Linux release 7.7.1908
Was hoping to get any pointers to further troubleshooting.
As soon as I enable these 2 lines:
restart_persistency_size = 512
modparam("drouting",
e Developer
http://www.opensips-solutions.com
On 9/13/20 3:18 AM, Alex A wrote:
> Thank.you
>
> On rabbitMQ event, it does not seem to be enabled though. All messages
> are being delivered as non persistent
>
> I was wondering if there are any tricks to change that
>
>
>
Thank.you
On rabbitMQ event, it does not seem to be enabled though. All messages are
being delivered as non persistent
I was wondering if there are any tricks to change that
Alex
On Sep 11, 2020, 3:23 AM, at 3:23 AM, "Răzvan Crainea"
wrote:
>Hi, Alex!
>
>
Hi,
Is it possible to enable persistence (rabbitmq delivery_mode:2) on RabbitMQ
CDR messages published by event_rabbitmq(3.1.x) ?
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control module in OpenSIPS supports the
loop-protect option, or whether specific module support for any command
is required in order to use it.
-- Alex
On 7/17/20 5:06 PM, Mario San Vicente wrote:
Thanks for your explanation Alex,
Actually i compiled the latest..git clone
https://github.com
This happens when an SDP body that has already been passed to RTPEngine, and
already adulterated by RTPEngine, is passed to it yet again.
Newer versions of RTPEngine have a loop protection feature to deal with it. It
involves injecting an unregistered a=rtpengine attribute into the SDP, to say
ffer from all the
problems of user-space in turn, so the economics can be really
different. Mileage of course greatly varies with the implementation details.
-- Alex
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allocator), but nothing on the order of gigabytes upon gigabytes.
Assuming 4 KB per call and 200,000 concurrent calls, that's ~800 MB, and
that is a very generous assumption indeed.
-- Alex
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Tel: +1-706-510-6800 / +1-800-250-5920
a "tuning" or "optimisation" that yields a "performance
increase" is profoundly misleading.
-- Alex
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__
that
the before and after don't lie.
On Fri, Jun 12, 2020 at 4:02 PM Alex Balashov
mailto:abalas...@evaristesys.com>> wrote:
But increasing the depth of the queue by 78x (if I'm not mistaken,
212992 is the default--at least, it is on
oad don't cause
non-trivial packet or connection queueing on the OS side.
-- Alex
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User
can't
process them fast enough isn't a solution to the problem of not
processing them fast enough. You're infinitely better off just
processing them faster.
-- Alex
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x
systems, which I guess also have "absolutely terrible sysctl defaults")
is faker than a Ponzi scheme. In some other contexts, this would be
called morally bankrupt and intellectually fraudulent. I guess here we
call it "mad dialer CPS" or whatever.
-- Alex
On 6/12/2
. It's not that Ubuntu Server is
mistuned, it's that you're abusing it. :-) You can't put the milk back
in the cow, although it's quite a spectacle ...
-- Alex
On 6/12/20 6:02 PM, Calvin Ellison wrote:
I noticed a way-too-small receive buffer value in the OpenSIPS startup
messages and it turns
pensips.org/cgi-bin/mailman/listinfo/users
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ht
│ ├─pickup
│ ├─qmgr
│ └─trivial-rewrite
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3.0.2
On Apr 30, 2020, 9:12 AM, at 9:12 AM, Johan De Clercq wrote:
>on what version is this ?
>
>Op do 30 apr. 2020 om 15:09 schreef Alex A
>>:
>
>> Setting the "First Only" flag on the carrier seem to be done the
>trick for
>> me.
>> It ro
Setting the "First Only" flag on the carrier seem to be done the trick for me.
It round-robins, while failing over to another carrier directly.
Thank you for your help.
On Thu, 30 Apr 2020 07:12:04 -0400 Alex A
<mailto:ale...@gtanetworkconsulting.com> wrote
[1] - https://opensips.org/docs/modules/3.0.x/drouting.html#param_carrier_id_avp
[2] -
https://opensips.org/docs/modules/3.0.x/drouting.html#func_route_to_carrier
Ben Newlin
From: Users <mailto:users-boun...@lists.opensips.org> on behalf of Alex A
<mailto:ale...@gtanetw
Hi Bogdan,
Will "use only the first gateway from the carrier" allow for round-robin for
the regular calls (ie. does it choose the first gw randomly ) ?
Thank you.
Alex
On Thu, 30 Apr 2020 03:40:21 -0400 Bogdan-Andrei Iancu
<mailto:bog...@opensips.org> wrote
Hi A
Hi Everyone,
Is it possible to failover to next carrier (instead next gateway) while using
drouting?
I got the below to work; however currently, use_next_gw gets the next gateway
in the list, so
if gwlist= #0,#3
and one of the carriers has multiple gateway IPs, the retry happens
Thank you
I try it out via rabbitMQ event subscription
On Apr 23, 2020, 10:53 AM, at 10:53 AM, Bogdan-Andrei Iancu
wrote:
>Hi Alex,
>
>Typical approach in this case is to do the accounting via a very fast
>backend (like db_flatstore, into a text file) and import the files
>o
Hi,
We are looking to deploy accounting/homer integration on Opensips 3.0.2.
As the first step deployed acc module with pgsql backend.
The config seem to be pretty straight-forward - see attached.
It appears that as soon as volume hits about 30-35k in_use transactions - the
server
It will trigger a failure_route.
—
Sent from mobile, with due apologies for brevity and errors.
> On Mar 15, 2020, at 4:20 AM, johan wrote:
>
> How can I catch in the script that fr-timer has expired ?
>
> I need to be able to see this expiry as I would like to failover on this.
>
>
> BR,
On Mon, Aug 12, 2019 at 10:52:09AM +0200, Giovanni Maruzzelli wrote:
> https://xkcd.com/378/
<3
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Dear All, I am working what appears to be a simple function for opensips 2.2.3,
however cannot seem to get it working.. Essentially, extract the groupID from
permissions module and add a prefix to R-URI on the egress side.
https://www.opensips.org/Documentation/Script-CoreFunctions-2-2#toc26
much older
version of mediaproxy that don't have this fault but I've got to update because
of a bug with the relay component on that build.
Does anyone have any suggestions.
Thanks, Alex
Alex Tatham
Technical Director
T:01233 220 943
E:alex.tat...@dmcplc.co.uk
W:www.dmctechnologies.co.uk
DMC Tech
RT* socket option so
>that opensips could bind to the same port after restart?
-- Alex
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ips")
>
>
>
> which fails
>
>
>
> Any ideas on a workaround on this?
>
>
>
> BR / Olle
>
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result.
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>failure_route[initial_request] {
># How can we arrive here right upon the receipt of the 302, not in
>onreply_route?
>}
>
>> On Sep 5, 2017, at 4:54 PM, Alex Balashov <abalas...@evaristesys.com>
>wrote:
>>
>> Yes, failure_route is the answer to all your objectives h
Yes, failure_route is the answer to all your objectives here. You can
intercept the 302, extract what you want from it, create a new branch
and fork the call elsewhere.
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gt; WARNING:core:do_action: error in expression at
> /etc/opensips/opensips.cfg:602
>
> does anyone have any idea what is causing this error or if this flag is
> even being evaluated ?
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, presume that the IP and port
endpoints on both ends stay the same.
So, if you suddenly start sending media from another place and expecting
to receive it there likewise, that will not be considered to be part of
the same phone call.
-- Alex
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ave, yeah.
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My understanding is that this is a rather simple module without sophisticated
state componentry, and that it logs things immediately as received, in the same
iteration of message processing.
-- Alex
--
Principal, Evariste Systems LLC (www.evaristesys.com)
Sent from my Google Nexus
r on 200 OK of BYE?
Are you referring to the ACC module, or some other method of accounting?
:-)
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>
> OpenSIPS Bootcamp 2017, Houston, US
> http://opensips.org/training/OpenSIPS_Bootcamp_2017.html
>
> On 07/04/2017 06:22 PM, Alex Megalokonomos wrote:
>
> As you may have noticed in my last reply, I reached that far as well but
> got stuck later on o
> Hi Alex,
>
> Thank you for the offlist provided data. Shortly, the ACK received by
> OpenSIPS from OmniPCX is broken as it is missing all the Route headers.
> According to the pcap, it looks like:
>
> ACK sip:udoioiia@10.0.1.106:49246;transport=ws SIP/2.0
> Record-Ro
/docs/modules/2.3.x/uac_registrant.html
>
> Let me know if you get stuck in this first step.
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developer
> http://www.opensips-solutions.com
>
> OpenSIPS Bootcamp 2017, Houston, US
> http://opensips.org/t
but was
unsuccesful.
In your second scenario, I am not interested in WS->WS calls so that auth
part is not an issue.
So I guess I need the uac_registrar, authorize by IP and usrloc parts.
Any relevant documentation to get me started since I'm still not clear on
what I need to change?
Best regards,
A
in order to convert the UDP-only sip
extensions to ws+ webRTC capable ones.
I have used this tutorial
http://www.opensips.org/Documentation/Tutorials-WebSocket-2-1 to get what I
assume is half the work (for RTP proxying) but I havent figured out the
rest yet.
Best regards,
Alex
On Thu, Jun 29
Hello,
We have the following scenario: our office call center is an Alcatel
OmniPCX Office setup.
This handles most of our needs and also provides 4 SIP extensions.
These are provided by what appears to be a Kamailio SIP server v 3.2.2 (no
webrtc or websockets support)
What we would like to do
;
>Please see attached trace.
>
>volga629
-- Alex
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at 11:39:14PM -0400, Satish Patel wrote:
> after google found bunch of post where people suggesting use
> fix_nated_sdp() is that right approach ?
>
> On Mon, Apr 24, 2017 at 11:25 PM, Alex Balashov
> <abalas...@evaristesys.com> wrote:
> > Yes, but RTP can come fr
isn't public then
> media never work.
>
> c=IN IP4 192.168.1.8.
>
> It should be
>
> c=IN IP4
>
> On Mon, Apr 24, 2017 at 11:04 PM, Alex Balashov
> <abalas...@evaristesys.com> wrote:
> > Satish,
> >
> > When you say "origin public address&q
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__
hnologies Ltd.
>
>
>
>*M*: +972535265553 l *Skype*: ziv_gabel l *E*: z...@communitake.com
>
>*T*: +972.4.696.8908 l *F*: +972.4.959.1654 l www.communitake.com
-- Alex
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Sent from my Google Nexus.
__
ile ago
>
>https://lists.cs.columbia.edu/pipermail/sip-implementors/2001-March/000601.html
>
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If MediaProxy is a SIP endpoint, that would be news to me.
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at the UA layer where failure to adhere to them does not
adversely affect the proxy's ability to relay the message.
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On 10/21/2016 06:36 PM, Newlin, Ben wrote:
Not only that, but provisional responses (except 100 Trying) are
required to have a To tag [1]. So you would likely run into issues with
UAs if you start returning messages without them.
That is an astute point.
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result on subsequent messages.
Do you guys see any problem on removing the to-tag of all 1XX messages?
Thanks
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that doesn't gather message form.
In contrast, onreply_route is for actual reply messages.
-- Alex
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On 06/28/2016 11:06 AM, Owais Ahmad wrote:
Thats not the case Alex. But I am expecting a large number of UDP
messages arriving on the same port as my udp listening socket.
Just want to be sure there is no wasteful work done parsing such packets.
You can be reasonably sure that OpenSIPS
stinfo/users
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On 05/23/2016 12:17 PM, Gupta, Rahul wrote:
Hi Alex, thanks for the quick reply. I don't see if
msg_apply_changes() is available in opensips. When I use it, I get
the bad config file error. I did add loadmodule textops.so
Oh, sorry! I thought this function was available in OpenSIPS too
uot;);
You can run msg_apply_changes() after calling fix_nated_contact(),
assuming it doesn't have any effects harmful to your cause:
http://kamailio.org/docs/modules/4.4.x/modules/textopsx.html#textopsx.f.msg_apply_changes
-- Alex
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1447
server would then
send this data to OpenSIPS over UDP.
This sounds substantially similar to a VPN, except without the benefit
of encryption.
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On 05/02/2016 05:08 PM, Dragomir Haralambiev wrote:
I have registration activity with Radius asin
So, why do you expect fragmentation from time to time as the OpenSIPS
memory manager allocates and frees SHM blocks?
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On 05/02/2016 04:44 PM, Alex Balashov wrote:
On 05/02/2016 04:22 PM, Dragomir Haralambiev wrote:
Opensips has not routed any calls.
Has it done anything else, including passive registration activity or
any periodic database-bound synchronisation tasks?
Also, what about passively deflecting
On 05/02/2016 04:22 PM, Dragomir Haralambiev wrote:
Opensips has not routed any calls.
Has it done anything else, including passive registration activity or
any periodic database-bound synchronisation tasks?
--
Alex Balashov | Principal | Evariste Systems LLC
1447 Peachtree Street NE
cumstances of the
timeout in a failure_route.
-- Alex
--
Alex Balashov | Principal | Evariste Systems LLC
1447 Peachtree Street NE, Suite 700
Atlanta, GA 30309
United States
Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct)
Web: http://www.evaristesys.com/, http://www.csrp
NOT be included in an application-level
referral that might leave the scope).
--
Alex Balashov | Principal | Evariste Systems LLC
1447 Peachtree Street NE, Suite 700
Atlanta, GA 30309
United States
Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct)
Web: http://www.evaristesys.com
On 03/31/2016 02:49 PM, Travis Manson-Drake wrote:
I would be more than happy to send it to you privately if that's ok?
Of course!
--
Alex Balashov | Principal | Evariste Systems LLC
1447 Peachtree Street NE, Suite 700
Atlanta, GA 30309
United States
Tel: +1-800-250-5920 (toll-free) / +1
hardware as your SIP proxy, correct?
--
Alex Balashov | Principal | Evariste Systems LLC
1447 Peachtree Street NE, Suite 700
Atlanta, GA 30309
United States
Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com
otline: 520.545.0333
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--
Alex Balashov | Principal | Evariste Systems LLC
1447 Peachtree Street NE, Suite 700
Atlanta, GA 30309
United States
Tel:
On 03/25/2016 03:48 PM, Dragomir Haralambiev wrote:
unknown command , missing loadmodule?
That sounds like a typo in the config.
--
Alex Balashov | Principal | Evariste Systems LLC
1447 Peachtree Street NE, Suite 700
Atlanta, GA 30309
United States
Tel: +1-800-250-5920 (toll-free) / +1-678
On 03/21/2016 04:36 PM, Rodrigo Pimenta Carvalho wrote:
According to the documentation "...It can be an IP address, hostname or
network interface id".
So, can I do the following configuration?
listen=tcp:wlan0:5060
Why can't you just do exactly that?
--
Alex Balashov |
Travis,
rewriteuri() is a legacy core function that does not support PVs.
Have you considered ...?
# Option 1.
$rU = $fU;
$rd = $avp(variable);
# Option 2.
$ru = "sip:" + $fU + "@" + $avp(variable);
-- Alex
--
Alex Balashov | Principal | Evariste System
What is the exact text of the error?
--
Alex Balashov | Principal | Evariste Systems LLC
303 Perimeter Center North, Suite 300
Atlanta, GA 30346
United States
Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
Sent from my
Have you attempted to increase the shared memory allocation to OpenSIPS (-m CLI
option)?
--
Alex Balashov | Principal | Evariste Systems LLC
303 Perimeter Center North, Suite 300
Atlanta, GA 30346
United States
Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct)
Web: http
Ali,
Is there any danger that you are calling rtpproxy_offer() twice, or
using rtpproxy_offer() in combination with fix_nated_sdp()[1]?
-- Alex
[1] http://www.opensips.org/html/docs/modules/2.1.x/nathelper.html#id293899
--
Alex Balashov | Principal | Evariste Systems LLC
303 Perimeter
Hello Aqs,
Why not strip the Route header instead of denying the request? That is
to say:
if(is_present_hf("Route"))
remove_hf("Route");
-- Alex
--
Alex Balashov | Principal | Evariste Systems LLC
303 Perimeter Center North, Suite 300
Atlanta, GA 30346
United Sta
Process:: ID=9 PID=743941 Type=SIP receiver udp:x.x.x.x:5060
Process:: ID=10 PID=743942 Type=SIP receiver udp:x.x.x.x:5060
Process:: ID=11 PID=743943 Type=SIP receiver udp:x.x.x.x:5060
-- Alex
--
Alex Balashov | Principal | Evariste Systems LLC
303 Perimeter Center North, Suite 300
Atlanta, GA
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