ot;101001",
> "Ref": 1,
> "Loged in": "YES",
> "State": "incall"
> },
> {
> "id": "101002",
> "Ref": 1,
> "Loged in
Hi Eran,
Have you tried rtpengine_manage() before playing the media ?
rtpengine_manage();
rtpengine_play_media("file=ringback_tone_file.wav");
exit;
Regards,
Jehanzaib
On Tue, May 7, 2024 at 2:10 AM Eran Leshem wrote:
> Hi, I would like to create a special extension
Hi Parathiba,
Could you capture the SIP packets? They'll provide insight into what's
going on.
Regards,
Jehanzaib
On Tue, May 7, 2024 at 12:40 AM Prathibha B
wrote:
> I'm able to hear the message given in the message queue but the call is
> not getting transferred to the online agent,
>
>
Hi,
You can use drouting for IP based authentication trunks.
For username/password based i think you can use digest_auth somethin like
digest_domain("MySIPTrunkRealm", "sip.trunk1.com", "username1", "password1")
digest_domain("MySIPTrunkRealm", "sip.trunk2.com", "username2", "password2")
in the
used to t_relay(2) but it is not
working in the 3.4.2 version anymore. i tried all the options but none of
them is working.
Regards,
Jehanzaib
On Tue, Nov 28, 2023 at 10:54 PM Jehanzaib Younis
wrote:
> I have the exact same issue. I am also not able to change the Via header.
>
>
I have the exact same issue. I am also not able to change the Via header.
Regards,
Jehanzaib
On Tue, Nov 28, 2023 at 10:48 PM Jehanzaib Younis
wrote:
> Hi Bogdan,
>
> The only difference is, I am receiving traffic on UDP and sending on TCP
> socket.
> sorry i forgot to
Hi Bogdan,
The only difference is, I am receiving traffic on UDP and sending on TCP
socket.
sorry i forgot to mention before the advertised_address is globally defined
(public ip)
Regards,
Jehanzaib
On Tue, Nov 28, 2023 at 10:26 PM Bogdan-Andrei Iancu
wrote:
> Hi,
>
> I just made a simple
Hi there,
Can you write an example of how you used rtpengine_offer() ?
Regards,
Jehanzaib
On Wed, Jun 7, 2023 at 3:07 AM wrote:
> Hello Community,
>
>
>
> We are trying to set up an Opensips with RTPEngine in a B2B configuration
> to handle media processing. There is no scenario for the B2B
Hi there,
Just t_relay(); to the Asterisk IP, it should work perfectly fine.
Regards,
Jehanzaib
On Fri, Nov 18, 2022 at 12:09 AM nutxase via Users
wrote:
> Yes so thats my idea but the problem i see is that asterisk sees the
> contact as the opensips details not the UAC device
>
>
> Sent
Hi Micheal,
You can use t_on_reply and then something like this:
if (t_check_status("200")) {
..your db query
}
Regards,
Jehanzaib
On Thu, Oct 27, 2022 at 9:22 AM Saint Michael wrote:
> Thank you for the clarification.
> But then how do you intercept the Connect?
> Where
rds,
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
> https://www.opensips-solutions.com
> OpenSIPS Summit 27-30 Sept 2022, Athens
> https://www.opensips.org/events/Summit-2022Athens/
>
> On 5/21/22 5:21 AM, Jehanzaib Younis wrote:
>
> Than
There’s bug tracking this issue:
> https://github.com/OpenSIPS/opensips/issues/2724
>
> For compiling tls_wolfssl, try from a clean clone.
>
> -ovidiu
>
> On Thu, May 19, 2022 at 08:08 Jehanzaib Younis
> wrote:
>
>> Thanks Ovidiu,
>> I just ch
estigated properly.
>
> Thanks,
> Ovidiu
>
> On Wed, May 18, 2022 at 09:02 Jehanzaib Younis
> wrote:
>
>> Thank you Bogdan,
>> That helped a lot. As you mentioned I need to start only with
>> server_domain or client_domain.
>> Now I changed my config a b
ch_ip_address", "[formsteams_server]")
> modparam("tls_mgm", "match_sip_domain", "[formsteams_server]")
> modparam("tls_mgm", "certificate", "[formsteams_server].)
>
>
>
> modparam("tls_mgm&q
Hi Bela,
I think you can use (opensips-cli): database add avp command
Regards,
Jehanzaib
On Wed, May 18, 2022 at 6:20 PM Bela H wrote:
> Hello,
>
> I want to set up a call forwarding and followed the instructions from the
> "Building telephony systems with OpenSIPS". However, this is a
;, "[dom1.formsteams.com]1")
modparam("tls_mgm", "client_sip_domain_avp", "tls_sip_dom")
When i enable the MS-Teams direct route domain i get the below error:
no certificate for tls domain ' dom1.formsteams.com ' defined
Regards,
Jehanzaib
On Wed, May 18, 202
Hi,
I am having trouble to send/receive OPTIONS to ms teams.
Using the dispatcher module. The socket is defined as tls:*mysbcip*:5061
Looks like when my opensips (3.2.x) tries to send OPTIONS. it is giving me
the following error
ERROR:proto_tls:proto_tls_conn_init: no TLS client domain found
d we receive an ACK back and then
> immediately another INVITE and so it goes on until the call is terminated.
>
> I'm less interested in working out what is broken on the client end and
> more interested in protecting our network.
>
> Thanks,
>
> Ray
> On 14/05/22 2:59 pm, Je
Hi Ray,
I assume the Max-Forwards remains same but the CSeq is increasing ya?
Regards,
Jehan
On Sat, May 14, 2022 at 2:42 PM Ray Jackson wrote:
> Hi all,
>
> I have a misbehaving SIP client which is repeatedly sending SIP ReINVITE
> messages over and over again (in the hundreds) and want to
gards,
>>
>> Bogdan-Andrei Iancu
>>
>> OpenSIPS Founder and Developer
>> https://www.opensips-solutions.com
>> OpenSIPS eBootcamp
>> https://www.opensips.org/Training/Bootcamp
>>
>> On 2/23/22 9:17 AM, Jehanzaib Younis wrote:
>>
&g
not defined
Does anyone know where to define this TLS domain? i have already set
Regards,
Jehanzaib
On Wed, Feb 23, 2022 at 8:17 PM Jehanzaib Younis
wrote:
> Thank you for your response.
> yes that is what i did but i am getting a couple of errors as shown below:
>
> Feb 23 02:14:1
ds,
Jehanzaib
On Wed, Feb 23, 2022 at 7:00 PM Ovidiu Sas wrote:
> What exactly is the issue?
> You can load both libraries and switch between them via the
> tls_library param of tls_mgm module:
> loadmodule "tls_openssl.so"
> loadmodule "tls_wolfssl.so"
> loadm
Hi,
Does anyone have a working example with tls_wolfssl ? I was able to work
with tls_openssl but i am try to test with wolfssl library and having a
hard time at the moment ;)
Thank you.
___
Users mailing list
Users@lists.opensips.org
Hi Mike,
I used to save all the blocked international prefixes in the local cache
and then you can check in the INVITE section.
If it matches, just drop the packet.
Regards,
Jehan
On Wed, Dec 15, 2021 at 11:44 AM Mike O'Connor wrote:
> Hi All
>
> I'm working with a company running an old
Looks like the domain name you are using is not matching with the
certificate name.
I have also noticed you are using tlsv1, better to generate the certificate
with tlsv1.2 or tlsv1.3
Regards,
Jehanzaib
On Tue, Nov 23, 2021 at 1:58 AM Devang Dhandhalya <
devang.dhandha...@ecosmob.com> wrote:
to-2-4-0#toc4
>
> Regards,
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
> https://www.opensips-solutions.com
> OpenSIPS eBootcamp 2021
> https://opensips.org/training/OpenSIPS_eBootcamp_2021/
>
> On 11/19/21 6:04 AM, Jehanzaib Younis wrote:
>
> Hi
Hi experts,
I have just upgraded my old opensips to 3.2 version. Getting quite a few
syntax errors. Running out of time, so thought maybe someone can quickly
help out .
The error comes at if(status=~"[12][0-9][0-9]")
May be someone can write us the correct syntax for status code checking?
Thank
Hi experts,
I am running opensips 2.4.x just checkin if there is a way to replace the
SIP header name?
I can see the remove_hf in the sipmsgops module but can't find a way to
replace the sip header name.
For example i have a header
H-ABC:
i want to replace H-ABC to H-NEWABC
Regards,
Jehanzaib
the solution regardless of carrier.
>
> Mark.
>
>
> On Wed, 9 Jun 2021 at 02:48, Jehanzaib Younis
> wrote:
>
>> Thank you for sending me the link but I am not trying to integrate with
>> MS teams.
>> The Dispatcher sends the OPTIONS without Contact to my carr
Ovidiu Sas wrote:
> It seems that you are trying to integrate with MS.
> Check this blog, you have your answer there:
> https://blog.opensips.org/2019/09/16/opensips-as-ms-teams-sbc/
>
> -ovidiu
>
>
> On Tue, Jun 8, 2021 at 20:52 Jehanzaib Younis
> wrote:
>
>> Hi f
Hi folks,
I have opensips 2.4.x.
I am using the dispatcher module. I can not see the Contact header in the
OPTIONS to the destination, is there any way we can add the contact header?
Thank you
Regards,
Jehanzaib
___
Users mailing list
Hi Alex,
You can save the data from your database to memory and fetch it during all
control. I will not recommend to use db directly. There might be any other
better way someone from the community can tell us ;)
Regards,
Jehanzaib
On Thu, Jan 21, 2021 at 4:01 PM Alexander Perkins <
Ohh ya. I have just disabled nat_traversal and removed nat_keepalive();
Need to add the bflag properly which actually fixed my issue.
Thank your help!
On Wed, Mar 11, 2020 at 6:59 PM Liviu Chircu wrote:
> On 09.03.2020 02:24, Jehanzaib Younis wrote:
> > if (is_method(
setbflag(3);# Mark as NATed
setbflag(NAT);
}
On Mon, Mar 9, 2020 at 12:56 PM David Villasmil <
david.villasmil.w...@gmail.com> wrote:
> Have you tried setting the bflag right before save()’ing during the
> REGISTER?
>
> On
flag", "NAT") & modparam("nathelper",
"sipping_bflag", 7)
On Mon, Mar 9, 2020 at 12:56 PM David Villasmil <
david.villasmil.w...@gmail.com> wrote:
> Have you tried setting the bflag right before save()’ing during the
> REGISTER?
>
> On Su
r 2020 at 23:22, Jehanzaib Younis
> wrote:
>
>> but i have strange issue.
>> nathelper keep on sending the OPTION to old IP:PORT. As soon as the new
>> REGISTER comes in, it should only send the option to the latest one.
>> For example, I see OPTION going to xx.xx
setting)
The new register comes in and the xx.xxx.xx.xx:25004 letsay and now
opensips sends to 25001 as well as 25004. Obviously my cpe only replies to
the latest one.
Anyone have faced this issue ?
Thank you
On Fri, Mar 6, 2020 at 2:54 PM Jehanzaib Younis
wrote:
> Thank you for your suggesti
> User-agent:: Phone Ver2.2
> Received:: sip:[RECEIVED IP]:5060 /*This IP will be used as
> request domain on relay the packet*/
> State:: CS_SYNC
> Flags:: 0
> Cflags:: SIPPING_RTO SIPPING_ENABLE NAT_FLAG
&g
> # Replace the contact IP with the received address from the network
> fix_nated_contact();
> }
>
> If you look at the registrations via CLI or database (if in use) you'll
> see that OpenSIPs is tracking both a contact and received value to deal
> with this - that will be where
Hi gang,
I have quite strange issue. When in some cases when my customer's have
10.202.128.xx range they are behind NAT. when i see opensipsctl ul show
number@sip
i see contact header as Contact:: sip:num...@10.202.128.xx:5060;transport=udp
Q=
which means when i try to relay the invite goes to
40 matches
Mail list logo
