Hi
when call is being trasfered to another number MS Teams sends new Invite
with SDP as 'a=inactive'. How can I put ringback ton as MOH for this
sitation?
I tried with:
if(is_audio_on_hold()) {
xlog("L_INFO", "onHOLD");
rtpengine_play_med
6/18/2021 ob 1:04 PM napisal:
Hello,
Just do not use ^ and $ in the search pattern. It is probably trying
to match the whole SDP packet, not single line.
On Fri, Jun 18, 2021 at 5:09 AM Miha via Users
mailto:users@lists.opensips.org>> wrote:
Hello
have issue with replace_body
Hello
have issue with replace_body as it does not change SDP.
My code looks like this:
if (has_body("application/sdp")){
if(search_body("a=inactive")){
*replace_body("^a=inactive$", "a=sendonly");*
}
$var(rtpengine_flags) ="trust-address replace-origin
Hello
when call is not pickup on teams and just canceled I get 603 Declined
which opensips send it further to SBC (our main sbc). When we get back
ACK and from SBC, there is wrong RURI and opensips does not relay this
ACK to MS teams.
Is anything can be done on opensips side in this situatio
yes. Thank you.
is there any way to get also attended transfer working?
Johan De Clercq je 6/2/2021 ob 11:21 AM napisal:
remove Refer from your supported methods.
Do note that attended transfer will not work in this case.
wkr,
Op wo 2 jun. 2021 om 10:15 schreef Miha via Users
mailto:users
ok, it does new seq invite, so not is is working.
thank you for help.
miha
Miha via Users je 6/2/2021 ob 10:11 AM napisal:
Hello
i manage to fix this. I did not do t_relay() also seq Invites, after
this everything works ok.
Just on question, regarding transfers, i see that MS Teams send
s_gflag(0)) xlog("L_NOTICE", "...in-dialog $rm
request\n");
# ...do all the things...maybe more logging like the line above...
- Jeff
On Tue, Jun 1, 2021 at 4:57 AM Miha via Users
mailto:users@lists.opensips.org>> wrote:
Hello
I have an issue and I am u
Hello
I have an issue and I am unable to find out what is wrong. Incoming
calls are working but when doing outbound call after 200OK, which is
send to Teams I get back ACK and after that Teams do again initial. I
guess this is not ok.
I am doing this for outband calls:
xlog("L_INFO", "rtp
Thank you
I will check.
Br
miha
On 18 May 2021, 13:08 +0200, John Quick , wrote:
> The client I was working with used this:
> https://docs.microsoft.com/en-us/microsoftteams/direct-routing-sbc-multiple-tenants
>
> I touched on the topic in my article about MS Teams, under the heading
> "Termino
btw what is the trick if you have multiple trunks to sbc teams (inbound,
outbound)? multiple companies?
br
miha
John Quick je 5/18/2021 ob 11:15 AM napisal:
Miha
Altering the text in the Record-Route headers with subst() function is not
the correct approach.
I believe the problem is that you
hello John
i have found what was causing the issue. is was topology hiding when ACK
was received by opensips.
thank you for all your help and time :)
br
miha
John Quick je 5/18/2021 ob 11:15 AM napisal:
Miha
Altering the text in the Record-Route headers with subst() function is not
the co
e request uri, try again with subst()!
On Mon, 17 May 2021 at 08:58, Miha via Users <mailto:users@lists.opensips.org>> wrote:
Hello
i need a little help how to chnage RR in responses to UDP GW
(requestes goes via TLS to MS teams).
So in reply i have like this:
RECOR
Hello
i need a little help how to chnage RR in responses to UDP GW (requestes
goes via TLS to MS teams).
So in reply i have like this:
RECORD-ROUTE:
,.
But i should have like this: RECORD-ROUTE:
,.
I tried to do it like:
subst_uri('/mtsbc.test.com:5061;transport=tls/xxx.xxx.xxx.:5
hello
i tried to put this in address table:
"*.pstnhub.microsoft.com" but it does not work.
On Tue, 11 May 2021 09:13:37 +0200
Johan De Clercq wrote:
> the pstnhub's can change their ip address.
> Therefore you need to use the fqdn.
>
> Op ma 10 mei 2021 om 2
found an issue. It was missing ip in addresses. Is there
any easier way to put all servers from Ms to addresses,
maybe just domain with "*."?
thank you
On Mon, 10 May 2021 19:23:20 +0200
Miha via Users wrote:
> Hello
>
> it seems for me that this works now. I only do not
2021 às 04:41, Răzvan Crainea
>
> >> escreveu:
> >>
> >>> Hi, Miha!
> >>>
> >>> According to your logs, opensips is 100% sending the
> OPTIONS through
> >>> tls, but I am not sure it is using the right
> certificate.
> &g
Hello
I have used letsenrypt for generating certs for Opensips.
Regarding configuration i have fallowed your configuration steps on
OpenSips blog.
socket=udp:xxx.xxx.xxx.xxx:5060 # CUSTOMIZE ME
socket=tls:xxx.xxx.xxx.xxx:5061
### Proto TLS
loadmodule "proto_tls.so"
modparam("proto_tls",
Hi
I have falow opensips configuration from your blog regarding MsTeams.
Version that I am using is:
(opensips-cli): mi version
{
"Server": "OpenSIPS (3.1.1 (x86_64/linux))"
}
ay 7 13:15:06 mtsbc opensips[1966]: WARNING:core:utimer_ticker: utimer
task already scheduled 100 ms ago (now
://opensips:opensipsrw@localhost
database_schema_path: /tmp/opensips-3.1/scripts
Miha via Users je 5/3/2021 ob 12:00 PM napisal:
Pasting also log:
database create
DEBUG: running command 'create' '[]'
DEBUG: db_name: 'opensips'
Password for admin MySQL user (root):
DEBUG: read passw
t;)
cursor.execute(statement, parameters)
INFO: created user 'opensips'
INFO: set password 'ow' for 'opensips' (MySQL)
INFO: granted access to user 'opensips' on DB 'opensips'
INFO: flushed privileges
DEBUG: connecting to mysql://opensips:opens
Hello
I have config for opensips-cli like this in /etc/opensips-cli.cfg
database_modules: ALL
log_level: WARNING
prompt_name: opensips-cli
prompt_intro: Welcome to OpenSIPS at SECUREVOIP
prompt_emptyline_repeat_cmd: False
history_file: ~/.opensips-cli.history
history_file_size: 1000
output_type
Hello
due to debugging i would like to combine cap from opensips and also cap
from rtpproxy (they are on different servers) so that I can check if RTP
is missing for certain call.
Can you help me with solving this issue :)
thank you
miha
___
Users
Hi
What is best tool for load testing that can generate also RTP?
Tnx
miha
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Hello
we are running still on 2.1. Due to some other things I would like first
to migrate to version 2.4. I went over documentation for version
migration from 2.1 to 2.2 and from 2.2 to 2.3 and from 2.3. to 2.4.
What I would like to know is what exactly is wrong in my config in where
i shoul
not been terrific,
> > due to some design choices made early in our work.
> Hovewer I believe it
> > should be much better in 2.0 and 2.1 vs. 1.x series.
> Some of it is
> > inherently due to VM scheduling jitter, some is because
> we are unwilling to
> > pu
t though,
but it also depends on hypervisor version and even particular CPU
generation.
-Max
On Tue., May 5, 2020, 6:10 a.m. Miha via Users,
mailto:users@lists.opensips.org>> wrote:
Hello
we have virtualized opensips and rtpproxy running on the same
server which is virtua
miha
> > >
> > > On Tue, 5 May 2020 10:27:22 -0300
> > > Daniel Zanutti wrote:
> > > > Hi Miha
> > > >
> > > > Could you explaining how does it break? We use it in
> > > > virtual machines and
> > > > our sa
iha
>
> Could you explaining how does it break? We use it in
> virtual machines and
> our safe limit is around 500 simultaneous calls, on
> dedicated single core
> VPS. Does CPU usage reach 100%?
>
>
>
> On Tue, May 5, 2020 at 10:11 AM Miha via Users
>
> wrote:
&
Hello
we have virtualized opensips and rtpproxy running on the same server
which is virtualized in vmware infrastructure. Servers are not old, also
traffic is not so big (cca 50 simultaneous calls). when there is a peak
cca 80 simultaneous calls RTP starts to break.
is there any special sett
the domain supports SRVyou know the drill .
But in both cases the SIP domain in SIP messages will be 'sip.test.com'
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
https://www.opensips-solutions.com
On 4/14/20 11:57 AM, Miha via Users wrote:
Hello
we have dns srv
Hello
we have dns srv record for failover. In dns srv we have two record.
So, one version of our devices does not support dns srv records. Is it
possible to register device directly to one A record which is wirtten in
DNS SRV record and then use ALIAS in opensips to right domain?
DNS SRV.
si
Liviu
thank you very much for your quick answer! I will try then to stick as
it is as it is the right way. If there will be no other choise that
maybe i try this.
thank you again!
miha
Liviu Chircu je 1/28/2020 ob 1:52 PM napisal:
On 28.01.2020 14:43, Miha via Users wrote:
Costumer is
Hi
first call flow.
1. Invite with FROM 12345@1.2.3.4
2. 200 ok with FROM 1.2.3.4@1.2.3.5
3. ACK, FROM is like in initial invite 12345@1.2.3.4
Costumer is saying that he expects from like it was send in 200ok (not
in inital invite, tag and CALLERID stays always the same) and we should
confirm
Hello
I get two connection infomrmation in SDP (doubled), which are the same.
How to remove one?
ps.: i am using rtpproxy.
thank you.
Miha
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.0, .8)
Must I do something else? This is done on on_replay route. I guess I
must change to 183 session in progress?
thank you for help!
Miha
On 8/19/2019 8:57 PM, Miha via Users wrote:
Hi, Răzvan!
thank you, so i was thinking right :)
br
miha
On Mon, 19 Aug 2019 17:28:07 +0300
Răzv
/github.com/sippy/rtpproxy/tree/master/makeann
>
> Best regards,
> Răzvan
>
> On 8/19/19 4:07 PM, Miha via Users wrote:
> > Hello guys
> >
> > first time doing this, normally I use freeswitch... Se
> in combination with rtpproxy how to enable ringback tone.
>
Hello guys
first time doing this, normally I use freeswitch... Se in combination
with rtpproxy how to enable ringback tone. I need to call
rtpproxy_stream2() i add it as file? Or there is some other option for
this if I would like that is played by UAS?
thank you for help!
miha
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