nything of value, but instead creating architectural
>> challenges.
>> >
>> > I'd love to hear feedback or experiences from others. There's always
>> something to learn :)
>> > -Brett
>> >
>> > On Tue, Dec 20, 2022 at 11:43 AM Terra
that K8S
> is buying you anything of value, but instead creating architectural
> challenges.
> >
> > I'd love to hear feedback or experiences from others. There's always
> something to learn :)
> > -Brett
> >
> > On Tue, Dec 20, 2022 at 11:43 AM Terrance
Was it something I said?
Terrance
On Sun, Dec 18, 2022 at 12:50 PM Terrance Devor wrote:
> Hello Everyone,
>
> Wow! Blast from the past... I am a long time member of this list, been a
> while.
>
> Question, anyone successful in deploying RTPProxy to a dockerized
> environ
Hello Everyone,
Wow! Blast from the past... I am a long time member of this list, been a
while.
Question, anyone successful in deploying RTPProxy to a dockerized
environment? Preferably to a Kubernetes managed environment.
Please Help Team :)
Kind Regards,
Terrance
_
Please post your route and branch cfg
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Thank you guys for your response. Right now we use port forwarding or RTP
range and RTPProxy and it
works great I understand that it works with 1:1 nat however, will it work
with a server behind a cisco router
that forwards all the RTP media to the server?
Kind Regards,
Terrance
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Hello Sasmita,
Just replace with the version that you are looking for. It's like magic!
http://www.opensips.org/html/docs/modules/1.11.x/db_cachedb.html#id249655
Terrance
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Hello Everyone,
Our current NAT'ed environment is as follows
Internet <---> Cisco Green Box <-> OpenSIPS (192.168.2.2)
<-> RTPProxy
(192.168.2.5)
I think this is called far end nat? Anyhow, it works perfectly fine (ie,
flows m
Ooops sorry, I overlooked that you said that... If you don't mind, i'll
pull parts of the conf
from federated to get things up and going ;)
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I never felt right installing oversip... There is just not room for two SIP
proxies in my world... ;)
Thanks Eric!
On the client side, anything proven for webrtc? Not too concerned with
video but audio is what
I am looking at.
N
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Eric!
That is great news! I did not really want oversip... Is this available on
1.9 flavours or only in the 2.x
revisions?
I will now uninstall ruby and oversip from the server as I did not feel
right having it there
Terrance.
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Hello Everyone,
As the the server side is stable and far end nat is implemented, we are now
looking to include the webrtc stack on top. From what I can see there will
be:
* OpenSIPS+OverSIP on the server side
On the client side I am unclear on a few things. Is it best to go with
jssip + sipml5 o
Thanks guys!
Schneur we do cache a number of checks address included however, since it's
cached to begin with, I would question needing to "cache the cache"?
If it was reading from dialog or subscriber that would be a different.
T
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top and ps is all you need.
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Is the permission module available on OpenSIPS 1.9?
On Thu, Aug 13, 2015 at 1:07 PM, Trevor Steyn wrote:
> Hi
> The permissions module does this. The standard trunking script has support
> for limiting channels.
>
> Regards
> Trevor Steyn
> On 13 Aug 2015 6:14 PM, &q
Hello Everyone,
Was wondering what is the preferred and current means of managing
concurrent calls per
client IP address. I am thinking it has to do with the Dialog module
however there is no "number of channels" field there.
If we can take advantage of caching mechanism it would be great however
ngrep -d eth0 -qt -W byline portrange 5060
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Attach SIP signalling pls.
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Thanks Aron!
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Hello Everyone,
Is it possible to do regex in conditional statements (ie, if).
What I am trying to do is something like:
if($si=~"192\.168\.1\.*")
To get all internal IP address. Is this possible?
Terrance
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>> I have changed the values in onreply_route and see them change in the
message.
>> The usage of the Dialog module and topology_hiding() is what makes it
work.
I concur. I think the days of writing test headers (ie, append_hf("X-Test:
bl.\r\n", "Call-ID")),
and studying SIP signalling ha
Worked perfectly. It took me three weeks to finally get this to work
thank to you.
Kind Regards,
Terrance.
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You know what Trying it right now within my script Will post how
it goes.
T
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Hello Ben,
This is great! Are you sure the set_advertised_address(outbound ip)
actually changed the
ip within the on_reply? The reason I ask is because we could not get it to
work for us,
and according to Razvan's original post. "No, you cannot change the
advertised address
in the onreply_route"
Hello Ben!
I really appreciate this. I am assuming you are talking
about topology_hiding()? Furthermore,
what is the best place to have the command and signal internal devices with
internal ips and
external devices with external ips. We are trying to have the change
applied branch and reply
signal
Hello Razvan,
Thanks again for your input.
>> but only on requests, not on replies.
And that is the exact issue we are experiencing. If I understand correctly,
we are unable
to modify the Contact HDR and RR for relay signalling such as "ACK, Session
Progress
and 200 OKs"? Is this not capability
Hello Razvan,
Thank you so much for your response. In our case it's not a 1:1 nat so we
cannot listen on
the public IP. I just need to change the RR when communicating locally vs
externally. That
is why I was referring to set_advertised_address. Will the same scenario
still work? Or do
I have to
I hope it's ok to re-phrase my question in this same email. Where is the
best place to change
the ip address using set_advertised_address for 200OK being sent out by
OpenSIPS to the
UAS.
T.
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Hello Razvan,
Thank you for your response. The problem I am experiencing is that I cannot
catch the 200OK that
OpenSIPS sends out except for within the onreply_route. I put debug
messages all over the script,
and the only place the 200OK that opensips would fire was in the onreply.
Can you please
Hello Everyone,
We are using Opensips 1.9.x and seeing that set_advertised_address is not
changing the recoird route within the on_reply route. What we are trying to
do
is change the record route for internal vs. external signalling. Is it not
possible
to change the advertised address within the o
Who?
T
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Can you please provide a sip trace. Who is sending the 500? Your media
server?
T
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Hello Satish,
I'm not a developer, but could you post your is_method("REGISTER") for the
authentication block? Also post one of the entries in the Subscriber table.
Are
you using clear text password or encrypted?
Please look at:
http://www.techsupportalert.com/content/how-ask-question-when-you-wa
Or Metaswich for that matter...
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Vlad, I think we could all benefit from the snippets if you know
what I mean ;).
T
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Bogdan/OpenSIPS team,
What is the take on this. Can other users approach such offers or
does the OpenSIPS team take presidency? Either way it's ok, and
thought it was a good idea to ask before approaching John.
Terrance
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Hello Everyone,
I sent this email earlier however did not seem to reach the list. Sorry for
the
redundancy.
We are running opensips in a NAT environment. When trying to change the
record_route using set_advertised_address in the on_reply route there is no
change. I am testing using some test head
1) What is the best way?
- Finite number of ways to cook a potato
2) Is there a way?
If there is a will... :)
Terrance.
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Hello Danilo,
Thank you for your response. We use nat_uac_test() in the config file for
originating clients, unfortunately this is focused towards the destination
(ie, the IP address of which the next signal is going to be sent to)
T
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Good news. Can install it into my OpenSIPS 1.8 server and click play?
Or are there modifications needed in opensips.cfg
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Hello Vardhan,
Seems very interesting at first look. Are you one of the developers of the
project?
Also, can it be run in a NAT environment as a bridge (ie,
WAN<--->192.168.2.5<-->WAN)
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Hello Everyone,
When trying to change the record_route using set_advertised_address in the
on_reply route there is no change. I am testing using some test headers and
know that the location is getting triggered however, cannot change the RR
using
set_advertised_address. Is this possible?
Kind Reg
The OpenSIPS project owners offer some excellent consulting solutions
that will save you a lot of time.
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Hello Răzvan,
Thank you for your response. I think I am making a mess of this. Maybe
if I explain what I am trying to accomplish. Please consider the following:
U 2015/03/10 15:31:47.591498 25.21.74.12:5060 -> 192.168.2.5:5060
INVITE sip:9187321212@74.75.31.22 SIP/2.0.
Contact: .
U 2015/03/10 15
Hello Mahan,
My suggestion would be to do a lot more research:
i)
http://www.opensips.org/Documentation/Tutorials-OpenSIPSAsteriskIntegration
ii)
http://www.voip-sip.org/wp-content/uploads/2011/08/Building-Telephony-Systems-with-OpenSIPS-1.6.pdf
Take a few weeks and get to know opensips and build
On Tue, Mar 10, 2015 at 6:06 AM, mahan77 wrote:
> Hello Terrance,
>
> Thank you for your time to replay back.
> It was basic dispatcher Config file and posted in the first place.
>
> This is my sip trace.
>
> interface: eth0 (192.168.1.0/255.255.255.0)
> filter: (ip or ip6) and ( port 5060 )
>
>
Hello,
Just visited your site and say that you can only download the .iso.
Do you make the full source available?
Terrance
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I tried to test for $dd everywhere, including right after loose_route
etc.., and I get null's everywhere.
I have a hair raising scenario where I need to know where the next message
(ie, destination) will
be in the branch and relay routes. This also includes re-invites, session
progresses, failures
Hello Sathees,
It is the appropriate place. I myself would need some more information such
as SIP trace, and maybe
more of the config?
Terrance
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Hello Răzvan,
Thank you for your response. Testing against the $dd and local ip would
work perfectly however, I have attempted to place some
test messages `xlog("L_INFO","Destiantion IP1: $dd\n");` in the
main,branch,relay and $dd is always null. I am not sure how this
is possible? Do I need to ad
Good news,
What is rtpengine support. Will the proxy manage RTP directly? Or will
we still have to use RTP/Media Proxy
Terrance
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Hello Everyone,
Our opensips is in a nat'ed environment , and we need a way to test if the
destination ip (ie, where the next message is going to be sent) is public or
private. Is there a variable, or even better a function that can test for
this.
Kind Regards,
Terrance
_
Hello Everyone,
Our environment is in an EC2 instance, and would like to fix the Contact
when
signalling internally (ie, private IP), and externally (ie, public IP). I
have added
the following code:
In branch:
if(!isflagset(5)) {
remove_hf("Contact:");
append_hf
ALG is an evil three letter abbreviation
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OpenSIPS+RTPProxy config will do that right out of the box more or less...
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SIP Listen Port - Whatever is defined in cfg (ie, 5060,5080 etc...)
RTPProxy Signalling Port if managing media - Whatever is defined in cfg
(ie, 7700)
RTP if managing media - Usually ranges from 8000-65535
Thank you come again :)
Nick from Toronto
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Hello Everyone,
We are currently using OpenSIPS 1.8.9 and would like to add the ability to:
- Maintain different route grades via prefixes (ie, standard, premium,
direct). The prefixes could be something like (1001, 1002, 1003) for
example. The different grades will have different /min pricing. R
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