you create the second leg
> with t_new_request? If os, you can use the ctx parameter from that function
> to make it visible in the new transaction.
>
>
>
> https://opensips.org/docs/modules/3.4.x/tm.html#func_t_new_request
>
>
>
> Ben Newlin
>
>
>
>
;
> Op ma 9 okt 2023 om 16:31 schreef Johan De Clercq :
>
>> write the info to a custom table and fetch it from there ?
>>
>>
>> Op ma 9 okt 2023 om 16:03 schreef Mickael Hubert :
>>
>>> Hi all,
>>> I need to send an INFO message with X-headers f
Hi all,
I need to send an INFO message with X-headers from other dialog.
I use t_new_request to send my request (apart from first dialog): OK
I use local_route to add fixed X-header: OK
But I want to copy some information from the first dialog to the new dialog
(created by t_new_request). How can
Hi all,
for an specific application, I need to get rtp ip and port for each side
(after 200OK)
can you tell me, if there is a way to catch rtp IP and port from sdp for
each side without regex ?
I can use this line $var(aline) = $(rb{sdp.line,c,0}); + regex, But maybe
there exists an easy way, ex:
//i3forum.org/one-consortium/
>
> -Brett
>
>
> On Wed, Sep 6, 2023 at 7:31 AM David Villasmil <
> david.villasmil.w...@gmail.com> wrote:
>
>> Is ST/SH being used other than the US? AFAIK it only applies to US
>> numbers, thus 10 digits, no?
>>
>> On Wed,
texte clarifié discuté ce matin
dès que je le reçois de la part de APNF.*
So we have to accept more than 15 digits in dest...
Le mer. 6 sept. 2023 à 16:07, Mickael Hubert a écrit :
> Answer of this french provider (in french sorry)
>
> *Pour le 1er point, la clause 2.5.3 des règles techn
think ?
Le mer. 6 sept. 2023 à 15:35, Mickael Hubert a écrit :
> Thanks a lot Daren,
> I have to contact this big french provider to explain its issue ;)
>
> Le mer. 6 sept. 2023 à 15:24, Daren FERREIRA a
> écrit :
>
>> We don’t have the same understand
iquement au niveau de l’en-tête SIP Request-URI.
>
> Cahier de tests:
>
> Appel (fixe ou mobile) depuis un ORT1 vers un ORT2 en transit SIP qui
> retransmet vers ORT3 avec présence de header Identity valide. ORT2 ajoute
> un préfixe de portabilité pour ORT3 dans R-URI mais pas
il: david.villasmil.w...@gmail.com
> phone: +34669448337
>
>
> On Wed, Sep 6, 2023 at 2:38 PM Mickael Hubert wrote:
>
>> We are deploying it in France.
>> In France on providers interconnections, we can see a format (made in
>> France maybe ;) )
>> prefix: +33
>>
com> a écrit :
> Is ST/SH being used other than the US? AFAIK it only applies to US
> numbers, thus 10 digits, no?
>
> On Wed, 6 Sep 2023 at 14:27, Mickael Hubert wrote:
>
>> yep I found...
>>
>> if (end - start < 2 || end - start > 15)
>> return -1;
>&
;
> Your number is to long
>
> E.164 is + [1-9] and {1-14} digits for total of 15 digits NOT starting
> with 0
> On 9/6/2023 7:16 AM, Mickael Hubert wrote:
>
> Hi all,
> I have an issue, when I verify a call with no E164 format (dest:
> +3310200123456789)
>
> *log
Hi all,
I have an issue, when I verify a call with no E164 format (dest:
+3310200123456789)
*logs:*
Sep 6 13:39:48 am-scr-001 /usr/local/sbin/opensips[622409]:
ERROR:stir_shaken:check_passport_phonenum: number is not in E.164 format:
3310200123456789
Sep 6 13:39:48 am-scr-001
Hi all,
Thanks Wadii for your help (in private ;) )
I developed a solution to check CRL in an external process (python script
scheduled by AWX).
My python script (download only in memory, not on disk)
*For CA certificates:*
- Download CA et intermediate certs
- Download PA cert (pa cert is used
juil. 2023 à 14:47, Mickael Hubert a écrit :
> Hi Razvan,
> Thanks a lot.
> I loaded the CRL for CA and certs and opensips start correctly ;)
>
> Have a good day !
>
> Le lun. 24 juil. 2023 à 16:07, Răzvan Crainea a
> écrit :
>
>> Hi, Mickael!
&
which seems reasonable to me: you need to provide
> the CRL of the entire path, not only of your intermediate cert. Did you
> try that?
>
> [1] https://stackoverflow.com/a/47398918
>
> Best regards,
>
> Răzvan Crainea
> OpenSIPS Core Developer
> http://www.opensips-solution
Hi all,
I'm working on stir and shaken, and I want to include all revoked
certificates.
I my list in DER format, I use this command to transform it to PEM format:
openssl crl -in man_crl.der -inform DER -outform PEM -out crl.pem
there is no erreur, I can read pem format (crl.pem):
-BEGIN X509
community. The PR
> will be reviewed and merged in the next days !
>
> Best regards,
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
> https://www.opensips-solutions.com
> https://www.siphub.com
>
> On 6/15/23 11:28 AM, Mickael Hubert wrote:
>
> H
Hi all,
I submited my first PR in the beautiful SIPssert tests project !
I pushed my work about authentication only for now (verification in the
future).
- Cert self generated (
https://blog.opensips.org/2022/10/31/how-to-generate-self-signed-stir-shaken-certificates/
)
- Private key stored in
ched_value(man_certificates_cache:certificate:https://
certs.example.org/public_am.pem);
stir_shaken_auth("$var(attest)","blabla","$var(cert)","$avp(privKey)",
"https://certs.example.org/public_am.pem","$var(orig)","$tU",
"$var(identity_hdr)&quo
olumn can by default only hold 128 characters. for an RSA
> private key in PEM format, it can go up to 800 chars.
>
> hope this helps
>
>
>
> *De :* Users *De la part de* Mickael
> Hubert
> *Envoyé :* lundi 29 mai 2023 14:55
> *À :* OpenSIPS users mailling list
> *Ob
Hi,
Can you tell me what is the best way to load our private key please ?
It would be great not to have it as clear text in opensips's configuration.
thanks in advance
Le lun. 21 nov. 2022 à 13:39, ryan embgrets a écrit :
> That was it.
>
> Working flawlessly.Thanks Vlad Patrascu
>
> Ryan
>
>
(public SBC) isn't forwarded too.
Is there a bug or misconfiguration for topology hiding module ?
thanks in advance
Le mar. 17 déc. 2019 à 11:09, Mickael Hubert a écrit :
> Sure !
> You can find it here (it's jinja2 template, but it readable ;) ) :
> https://github.com/Mickaelh51/op
(public SBC) isn't forwarded
too. But the BYE from UAC is forwarded correctly
Is there a bug or misconfiguration for topology hiding module ?
thanks in advance
Le mar. 17 déc. 2019 à 11:09, Mickael Hubert a écrit :
> Sure !
> You can find it here (it's jinja2 template, but it reada
, David Villasmil <
david.villasmil.w...@gmail.com> a écrit :
> Can we see your cfg ?
>
> On Mon, 16 Dec 2019 at 16:44, Mickael Hubert wrote:
>
>> Thanks for your help David,
>> I'm already in debug level:
>> log_level=4
>>
>> UAC is not in locati
thanks
Le lun. 16 déc. 2019 à 15:52, David Villasmil <
david.villasmil.w...@gmail.com> a écrit :
> please increase the debug level and paste the log. Also, check what is
> saved as the location for the user.
>
> On Mon, 16 Dec 2019 at 14:48, Mickael Hubert wrote:
>
&
Le lun. 16 déc. 2019 à 14:27, Mickael Hubert a écrit :
> Hi David,
> Yes I use it
>
> if (nat_uac_test("3") && ($Ri == $var(publicip) || $Ri == $var(vpnip)))
> {
> xlog("L_INFO","$avp(startlog) -- Nated EP Detected\n");
> if (
}
if (fix_nated_contact())
{
xlog("L_INFO","$avp(startlog) -- Nated $rm's Contact Fixed !\n");
}
if (fix_nated_sdp("10"))
{
xlog("L_INFO","$avp(startlog) -- Nated SDP Fixed for $rm\n");
}
}
Le lun. 16 déc. 2019 à 13:50, David Villasmil <
david.v
:27 am-frontal1a-test /usr/local/sbin/opensips[26160]: retcode
= -6
I don't know why OpenSIPS tries to send the Re-invite to client private IP
instead client public port and IP (natted).
Do you have an idea please ?
thanks
Le jeu. 12 déc. 2019 à 11:09, Mickael Hubert a écrit :
> Hi all
ounder and Developer
> https://www.opensips-solutions.com
> OpenSIPS Bootcamp Pre-Registration
> https://opensips.org/training/OpenSIPS_Bootcamp/
>
> On 12/11/19 12:53 PM, Mickael Hubert wrote:
>
> Hi all,
> Someone can explain me the difference between force_send_socket(
Hi all,
I have an issue, opensips doesn't want forward Re-INVITE during UDP to TCP
mapping session.
Customer (NATTED) -- TCP --> (public interface listen tcp:8060) OpenSIPS
(private interface listen udp:5060) --> rest of infrastructure (udp:5060)
I can send a call from customer to OpenSIPS
Hi all,
Someone can explain me the difference between force_send_socket() and $fs
please ?
is it the same thing ?
thanks
___
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Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Hi all,
is there a way to force the SST expire by calls ?
For example, for "normal" call I want to have 1800s, but my proxy processes
MRCPv2 calls too. And for these calls, I want to change expire to 120s.
Thanks in advance
++
___
Users mailing list
The call must be establish between softphones in sip.
And opensips create siprec session with orkaudio.
Rtpproxy send rtp to orkaudio too.
Le jeu. 5 déc. 2019 19:47, VoIP Security via Users
a écrit :
> Thanks for that information Mickael. But for testing what I was doing
> is, I connected 2
Hi
I confirm that opensips + rtpproxy + drachtio server work like a charm !
I was be able to record the rtp flow on an Asterisk server.
++
Le jeu. 5 déc. 2019 19:11, VoIP Security via Users
a écrit :
> Thanks, I will try again using drachtio server.
>
>
>
>
here is a better way ...
Le mer. 27 nov. 2019 à 17:51, Ben Newlin a écrit :
> Have you tried using the substitution transformation?
>
>
>
> https://www.opensips.org/Documentation/Script-Tran-2-4#toc82
>
>
>
> Ben Newlin
>
>
>
> *From: *Users on behalf of M
Hi all,
I want to extract the channel ID from a line in SDP
(a=channel:d87c363c1b5b4f13@speechrecog)
I can extract this line, but I don't know how can I have only the ID
(d87c363c1b5b4f13)
Do you have a way for me please ?
thanks in advance
SDP example:
***
m=application 1544 TCP/MRCPv2
he caller and callee
> fields. Check Example 1.6 in the siprec documentation[1] to see how you
> can build this xml, that will appear in the participant's node.
>
> [1] https://opensips.org/html/docs/modules/3.0.x/siprec.html#idp5579200
>
> Best regards,
> Răzvan
>
> On 8/20/19 3
Hi all,
I use SIPREC module, that works like a charm with SRS drachtio server ;)
In my initial invite (sip only, no siprec), I have User-To-User header, and
I want to copy it into SIPREC xml part, like the participants.
Ex:
00FA081875333AA;encoding=hex
Do you have a way for me to add this
OK I'm stupid !
I had deactivate my reply route for loose route part... So 200OK in
reinvite didn't work ;)
it's ok now.
sorry
Le mer. 24 avr. 2019 à 12:08, Mickael Hubert a écrit :
> Hi All,
> I have an issue when reinvite sdp is processed by opensips, it don't
> change IP in
Hi All,
I have an issue when reinvite sdp is processed by opensips, it don't change
IP in c header in 200ok SDP.
But it works well for the first INVITE.
Ex:
SIP
- IPBX (172.31.10.241) <--> (172.17.50.153) Opensips (10.1.15.153) <-->
(10.1.15.152) Asterisk server
RTP
- IPBX (172.31.10.241) <-->
nSIPS,
> you'll have to adjust your opensips.service file to run a different
> ExecStop command, something like `/usr/sbin/opensips -C -f
> /etc/opensips/opensips.cfg && /usr/bin/pkill --pidfile
> %t/opensips/opensips.pid`
>
> Hope this helps!
> Răzvan
>
> On
I would like to know, if there is a way to check opensips's config files
before stop service.
Ex: If I do a syntax error into cfg files (with opensips started), I want
to "ban" the daemon shutdown (service opensips stop)
I'm playing with opensips.service and opensips -C -f $config to try to
reach
numbers are stored in DB - which module/table are you
> talking about ?
>
> Regards,
>
> Bogdan-Andrei Iancu
>
> OpenSIPS Founder and Developer
> http://www.opensips-solutions.com
> OpenSIPS Bootcamp 2018
> http://opensips.org/training/OpenSIPS_Bootcamp_2018/
Hi
Thanks a lot for guys !
I'll try m4 to test.
But If prefer ansible ;) maybe change opensips startup script to work with
jinja template...
++
++
Le lun. 7 janv. 2019 12:43, Gerwin van de Steeg <
gerwin.van.de.st...@vadacom.com> a écrit :
> If you're ok with them being pulled from a secondary
Hi all,
I'm looking for a way to avoid all plain text password into configuration
files.
maybe store sensibles data into secret file and read variables into
opensips configuration file ?
Ex:
*secret file:*
MYSQL_USER: opensips
MYSQL_PWD: 4845123121
...
*Configuration file:*
from:
t;
> Răzvan Crainea
> OpenSIPS Core Developer
> http://www.opensips-solutions.com
>
> On 12/22/18 4:05 PM, Mickael Hubert wrote:
> > Hi all,
> > for my first day of vacations, I've created an ansible role for OpenSIPS
> ;)
> > I saw, there isn't any role fo
Hi all,
for my first day of vacations, I've created an ansible role for OpenSIPS ;)
I saw, there isn't any role for OpenSIPS on ansible galaxy website.
So, it's not perfect but it works like a charm for Debian 8 / 9.
https://galaxy.ansible.com/mickaelh51/ar_opensips_from_sources
Next step:
-
And it's not clean but works like a charm.
@Răzvan if you have an idea about the CANCEL issue with $dlg_val is not
populate, I'm interesting ;)
thanks
Le mar. 20 nov. 2018 à 13:58, Mickael Hubert a écrit :
> OK thanks, I will write something about that ;)
>
> I tested your solution, it wo
a feature request
> for this :):
>
> https://github.com/OpenSIPS/opensips/issues
>
> Best regards,
> Răzvan
>
> On 11/20/18 12:15 PM, Mickael Hubert wrote:
> > Hi Răzvan,
> > thanks a lot for your answer !
> >
> > Ok I will try this workaround
>
val(setid) = $avp(setid);
>
> # CANCEL or BYE, *after* loose_route() is called
> # so that the dialog is matched/found
> loose_route();
> $avp(setid) = $dlg_val(setid);
>
> Hope this helps.
>
> [1] https://opensips.org/html/docs/modules/2.4.x/dialog#pv_dlg_val
>
> Best
Hi all,
I have a rtpengine with opensips, when I use rtpengine without setid_avp
all works like a charm
modparam("rtpengine", "rtpengine_sock", "udp:10.13.0.129:12221")
*When I want use setid_avp:*
modparam("rtpengine", "setid_avp", "$avp(setid)")
modparam("rtpengine", "rtpengine_sock", "1 ==
Hi all,
I have a non technical question.
why phonenumbers into database are stocked whitout + sign (non E164 format)
?
is it for a prformance purposes ?
I'm looking for the best way to store E164 phonenumbers for a personnal dev.
thanks in advance
++
Micka
I've modified my init.d file with
/usr/local/sbin/opensips -P /var/run/opensips/opensips.pid *-m 256 -M 8* -u
opensips -g opensips -f /usr/local/etc/opensips/opensips.cfg
wait and see ;)
2014-09-16 15:12 GMT+02:00 Mickael Hubert m.hub...@hexanet.fr:
Hi all,
after upgrade Opensips, I've
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
MAX_URI_SIZE 1024, BUF_SIZE 65535
poll method support: poll, epoll_lt, epoll_et, sigio_rt, select.
git revision: c529bb1
main.c compiled on 16:58:09 Sep 11 2014 with gcc 4.7
2014-09-10 16:16 GMT+02:00 Mickael Hubert m.hub
Hi Bogdan,
in addition, we use these options: -m 64 -M 4 and in opensips.cfg
children=20
private memory: 4 Mo
Share memory: 64Mo
We use a server with 16Go of ram
it's good conf ?
Thanks a lot for your help.
2014-09-10 11:44 GMT+02:00 Kevin Mathy k.ma...@hexanet.fr:
Hi Bogdan,
Here's
On 10.09.2014 12:58, Mickael Hubert wrote:
Hi Bogdan,
in addition, we use these options: -m 64 -M 4 and in opensips.cfg
children=20
private memory: 4 Mo
Share memory: 64Mo
We use a server with 16Go of ram
it's good conf ?
Thanks a lot for your help.
2014-09-10 11:44 GMT+02:00 Kevin
Hi list,
I want rewrite user part in all uris.
Ex:
INVITE -- sip:*+33*X@ -- OpenSIPS -- sip:*0*X@
-- Client X
I already use this command in textops module:
*subst_user('/\+33/0/g');*
It's work fine, but it's just r-ruri is changed, no From header or To
header uri.
to use uac_replace_from/to()
functions from the uac module.
Best regards,
Razvan Crainea
OpenSIPS Core Developer
http://www.opensips-solutions.com
On 04/17/2014 12:59 PM, Mickael Hubert wrote:
Hi list,
I want rewrite user part in all uris.
Ex:
INVITE -- sip:*+33*X
the
sipcature module when the HEP messages is received by opensips.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 09.12.2013 16:05, Mickael Hubert wrote:
Hi list,
I have an issue with sipcapture module in capture server.
When client send
Hi list,
I have an issue with sipcapture module in capture server.
When client send HEP message to capture server, I have this message in
Opensips.log:
*Dec 9 14:48:49 sipcapture1 /usr/local/sbin/opensips[16871]:
INFO:core:parse_first_line: method not followed by SPDec 9 14:48:49
sipcapture1
the trace_dialog() between a set_debug(6) and set_debug(), to
see what is going on inside the function. Maybe the debug logs will give
you an idea.
Best regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
On 05.12.2013 09:47, Mickael Hubert wrote:
Hi
Hi list,
I have an issue with my siptrace module.
see below my extract of opensips.cfg.
I want duplicate dialogs's messages to the other server (10.84.8.201), no
DB.
But, it doesn't work... no sip message in my wireshark trace.
Have you an idea ?
Thanks in advance
*#
Hi list,
Can you confirm that this command (with 0), registration (toto = yes) is
never delete ?
*opensipsctl fifo cache_store local toto yes 0*
Thanks in advance
--
Cordialement
HUBERT Mickaël
Ingénieur VOIP - Hexanet
--
___
Users mailing list
Hi jef,
I advice you to use the perl module OpenSIPS, and create a perl
function. This is what I do to get the fields in resolv NAPTR
++
Le 24/07/2013 14:50, Jeff Pyle a écrit :
Hello,
Is there a way to extract an IP address list from SRV and/or NAPTR
records from within the script?
I
use:
remove_hf(Diversion), to rewrite the second diversion header. Opensips
delete the first diversion header.
How can I rewrite the good diversion header, here the second ?
thanks
Le 02/05/2013 19:21, Saúl Ibarra Corretgé a écrit :
On May 2, 2013, at 4:20 PM, Mickael HUBERT wrote:
Thank you
some details:
remove_hf delete all Diversion header, how can I delete or rewrite good
header ?
Le 03/05/2013 10:48, Mickael HUBERT a écrit :
Hi,
I have coded with while for treat all diversion headers.
Ex:
/while ($var(i) $(hdrcnt(Diversion)))//
//{//
//if(!$(hdr(Diversion)[$var(i
http://www.opensips-solutions.com
On 05/03/2013 10:58 AM, Mickael HUBERT wrote:
some details:
remove_hf delete all Diversion header, how can I delete or rewrite good
header ?
Le 03/05/2013 10:48, Mickael HUBERT a écrit :
Hi,
I have coded with while for treat all diversion headers.
Ex:
/while
Hi list,
I would like parse Diversion header and to have in variable counter
parameter.
I have alredy $dip to privacy parameter, etc ... , but I don't have
counter para.
how can I do have the parameter please ?
thanks in advance
Mike
___
Users
Thank you very much, it's perfect ;)
but, if I have many Diversion header in the same INVITE ? This command
works ?
Le 02/05/2013 15:06, Saúl Ibarra Corretgé a écrit :
On May 2, 2013, at 1:51 PM, Mickael HUBERT wrote:
Hi list,
I would like parse Diversion header and to have in variable
Hi Marco,
see bellow my configuration, it's works ;)
/debug=3//
//log_stderror=no//
//log_facility=LOG_LOCAL0//
//
//fork=yes//
//children=4//
//
///* uncomment the next line to disable TCP (default on) *///
//disable_tcp=yes//
//
//port=5060//
//
///* uncomment and configure the following
,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 03/07/2013 06:25 PM, Mickael HUBERT wrote:
I list,
I want extract User to P-Asserted-Identity URI
Hi list,
I have a issue when my backend mysql (to acc's table) is slow (it's
mysql server shared between differents services).
Endeed Opensips slowed too to SIP processing or if mysql crash, Opensips
crash too.
My Acc's table is the only realtime table.
Is there any method to improve this ?
the db_flatstore engine for acc (as SQL backend) to write Acc data
on files on disk. You can later import this files into DB without
affecting or slowing down opensips.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 03/08/2013 02:39 PM, Mickael HUBERT
I list,
I want extract User to P-Asserted-Identity URI, but I cannot find a
variable.
To user's PPI is $pU, but to PAI ?
Thanks in advance
Mickael
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On 03/05/2013 11:25 AM, Mickael HUBERT wrote:
Hi list,
I have a issue in my drouting module.
In fact I have many prefixes and overlap is not OK to me.
_Example:_
Prefix: 3669 (In France is premium rate number, 4 digits is talking
clock service)
Prefix: 36 (is prefix hungary)
In my
correct, I would say.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com
On 03/06/2013 12:09 PM, Mickael HUBERT wrote:
Hi,
It's not OK for me, look:
If I have:
+33 366985475 (landline number in France)
and
+33 3669 (premium rate number in France
Hi list,
I have a issue in my drouting module.
In fact I have many prefixes and overlap is not OK to me.
_Example:_
Prefix: 3669 (In France is premium rate number, 4 digits is talking
clock service)
Prefix: 36 (is prefix hungary)
In my dr_rule table is:
/INSERT INTO `dr_rules`
Hi Nick,
call drouting command if(!do_routing(...)) with another priority ?
how ?
thanks
Le 05/03/2013 11:12, Nick Altmann a écrit :
By using another priority.
--
Nick
2013/3/5 Mickael HUBERT mick...@winlux.fr mailto:mick...@winlux.fr
Hi list,
I have a issue in my drouting module
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