Hi,
$fn is from name and name maybe quoted in the header value.
Try to use $fU to have username part (actual number)
On Tue, Aug 30, 2022 at 9:00 AM Bogdan-Andrei Iancu
wrote:
> Again,
>
> your DP rule is performing NO change over the input. The whole input, as
> received, it provided as
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
wrote:
> Hello
>
> have issue with replace_body as it does not change SDP.
> My code looks like this:
>
> if
Hello Mark,
IMO, it is Asterisk side. Of course it depends on your setup but probably
you need Asterisk sip peer to opensips. Do not know for pjsip, for older
sip_chan it would be something like:
[opensips]
type=friend
deny=0.0.0.0/0.0.0.0
permit=OPENSIPS_IP/255.255.255.255
host=OPENSIPS_IP
You
s like I'm on the home straight.
> Thanks once again for all your help
>
> cheers,
>
> Mark
>
>
>
> On Fri, 21 Aug 2020 at 14:59, Stas Kobzar wrote:
>
>> Hello Mark,
>>
>> In my case I do have a path in the location record. Here is my example
>> from
Hello Mark,
In my case I do have a path in the location record. Here is my example from
"ul show" (I changed my real domain and IPs):
AOR:: 9...@example.com
Contact:: sip:suvp4v56@1p6pc0g6m3ml.invalid;transport=ws Q=
Expires:: 494
ani->ani;
> prefix->prefix; src_ip->src_ip; dst_ip->dst_ip; acctid->acctid;
> carrierid->carrierid; ruleid->ruleid;")
> modparam("acc", "db_url", "flatstore:/var/log/acc")
>
> Is there a proper placement of
> do
By the way, you have to copy "version" flat text storage file too
corresponding your opensips version
and correct path set in configuration file
like: text:///opt/opensips/etc/opensips/db
On Thu, Aug 6, 2020 at 1:37 PM Stas Kobzar wrote:
> Hello,
>
> You should create the fil
Hello,
You should create the file with headers. You can copy required storage file
from here:
https://github.com/OpenSIPS/opensips/tree/master/scripts/dbtext/opensips
And, of course, make sure you have good owner and permissions set to the
file.
On Thu, Aug 6, 2020 at 1:26 PM Vic Jolin wrote:
e?
> Also, perhaps 3.1 should handle it better if parameter is missing rather
> than giving a segfault (something for 3.2?)
>
> Once again I'm indebted to you for your help
>
> thanks very much & all the best
>
> Mark
>
>
>
>
>
> On Tue, 28 Jul 2020 at 15:
e?
> Also, perhaps 3.1 should handle it better if parameter is missing rather
> than giving a segfault (something for 3.2?)
>
> Once again I'm indebted to you for your help
>
> thanks very much & all the best
>
> Mark
>
>
>
>
>
> On Tue, 28 Jul 2020 at 15:
Hi Allen,
Did you try with two parameters: name, value?
*$avp(keys) = "registered";*
$avp(values) = "true";
xlog("Raised E_WFC_REGISTERED $avp(values)");
raise_event("E_WFC_REGISTERED", *$avp(keys)*, $avp(values));
I know they are said to be optional in the
PJSIP without any joy but it's not
> a priority for us at the moment so we'll have to investigate the cause later.
>
> Thanks for your help Stas, and thanks also to the others who took the time to
> reply.
>
> cheers,
>
> Mark
>
>
> On Tue, 14 Jul 2020 at 16:23,
lot of additional
> functionality linked to the Asterisk server so our main need for OpenSIPS is
> to reduce unnecessary load (e.g. re-REGISTER from mobile devices).
>
> On Tue, 14 Jul 2020 at 16:23, Stas Kobzar wrote:
>>
>> Hello Mark,
>>
>> I had a similar
Hello Mark,
I had a similar challenge. Using "path" module on both opensips helps
to overcome this problem.
https://opensips.org/docs/modules/3.2.x/path.html
In your mid-registerer you need to enable path support. See "save"
function params p0 and v.
in your webrtc opensips use path module and
do following:
if ($DLG_lifetime < 2) usleep("20");
This actually works fine for me.
Thank you for your help and ideas!
On Tue, Feb 7, 2017 at 8:45 AM, Stas Kobzar <stas.kob...@modulis.ca> wrote:
> Hi Bogdan,
>
> It seems like $DLG_status will not work in my case.
>
>
gt;
> It will delay slightly all reINVITES, but also will guarantee correct
> order of packets.
>
> 2017-02-06 22:46 GMT+03:00 Stas Kobzar <stas.kob...@modulis.ca>:
>
>> Hello Bogdan,
>>
>> In my case, ACK for previous INVITE has already been received by
>> Op
t; http://www.opensips.org/html/docs/modules/2.2.x/dialog.html#id297400
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 02/02/2017 10:31 PM, Stas Kobzar wrote:
>
> Hello List,
>
> My call flow has in
INVITEs (to
> force a later retransmission) :
> http://www.opensips.org/html/docs/modules/2.2.x/dialog.html#id297400
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 02/02/2017 10:31 PM, Stas Kobzar wrote:
>
&
string?
>
> For example, input string 123-45-67, or 12-345-67, but output string
> should be 1234567.
>
> Thank you.
>
>
> mailto:denis7...@mail.ru <denis7...@mail.ru>
>
> ___
> Users mailing list
> Users@lis
/pidf+xml presence content type.
It works as expected with OpenSIPS MI publish commands.
Case is closed.
Many thanks for comments and help.
Stas
On Fri, Jan 29, 2016 at 1:25 PM, Stas Kobzar <stas.kob...@modulis.ca> wrote:
> Hello Bogdan,
>
> Polycom actually did not want
; Hi Stas,
>
> Is this the actual draft :
>
> http://www.cs.columbia.edu/sip/drafts/impp/draft-rosenberg-impp-pidf-00.txt
>
> ?
>
> Best regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 23.01.2016 17:12, Sta
f-00.txt
>
> ?
>
> Best regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 23.01.2016 17:12, Stas Kobzar wrote:
>
> Hi Bogdan,
>
> I can confirm that Polycom still use xpidf with the latest firmware ver
>
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
--
Stas Kobzar
Developeur VoIP / VoIP Developer
Modulis.ca Inc.
# Bureau / Offi
Polycom and SIP SIMPLE.
>
> Ben Newlin
>
> From: <users-boun...@lists.opensips.org> on behalf of Stas Kobzar <
> stas.kob...@modulis.ca>
> Reply-To: OpenSIPS users mailling list <users@lists.opensips.org>
> Date: Monday, January 25, 2016 at 4:17 PM
>
sips-solutions.com
>
> On 13.01.2016 18:47, Stas Kobzar wrote:
>
> Hi Bogdan,
>
> I do not think the DOCTYPE is the problem here. What I see is that when I
> use MI to publish this application/xpidf doc, OpenSIPS does not want to
> parse the document, and if I understand correct,
e agent/server model). Am I wrong with this ?
>
> Best regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 12.01.2016 17:10, Stas Kobzar wrote:
>
> Hello Bogdan,
>
> Thank you for your response.
> The
to the crash, even if the XML is not properly formated, it
> should not crash - can you send me the actual MI command + content to try
> to reproduce the crash and have it fixed ?
>
> Best regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-s
Question:
Is there a way to make xpidf presence content type work with pua_mi module?
Thank you,
--
Stas Kobzar
Developeur VoIP / VoIP Developer
Modulis.ca Inc.
# Bureau / Office: 514-284-2020 x 246
Email: s <http://firstname.lastname>
Hi Bogdan,
Does it mean that the returned value is string now?
```avp_val.s.s = ""```
I think we might have to do many changes in our script.
What is the best way to check now if avp is set and not null?
Thank you,
Stas
---
Reply to this email directly or view it on GitHub:
packets
Thank you in advance,
--
Stas Kobzar
Developeur VoIP / VoIP Developer
Modulis.ca Inc.
# Bureau / Office: 514-284-2020 x 246
Email: s <http://firstname.lastname>tas.kob...@modulis.ca
https://www.modulis.com
<https://www.modulis.com/e
7c
main.c compiled on 20:32:30 Sep 26 2015 with gcc 4.4.7
core "bt full": https://www.dropbox.com/s/zmtkepwzanyfmq4/core.txt?dl=0
OpenSIPS config file:
https://www.dropbox.com/s/ut635s8s4d9axta/opensips.cfg.txt?dl=0
OpenSIPS log: https://www.dropbox.com/s/1ohh5f03sbvy3ej/opensips
You can view, comment on, or merge this pull request online at:
https://github.com/OpenSIPS/opensips/pull/631
-- Commit Summary --
* Fixing bug in debian init script.
-- File Changes --
M packaging/debian/opensips.init (2)
-- Patch Links --
io E. Goncalves
>
> 2015-09-02 11:26 GMT-03:00 Stas Kobzar <stas.kob...@modulis.ca>:
>
>> Hello,
>>
>> I hope this is a good list to post my question.
>>
>> I am interested in OpenSIPS Certified Professional.
>> Is there any upcoming certific
Hello,
I hope this is a good list to post my question.
I am interested in OpenSIPS Certified Professional.
Is there any upcoming certification exam planned in nearest future?
Thank you!
--
Stas Kobzar
Developeur VoIP / VoIP Developer
___
Modulis.ca Inc.
# Bureau / Office
se print in gdb:
> _c->callid
> _ci->callid
> *_ci->callid
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 31.08.2015 18:32, Stas Kobzar wrote:
>
> Hello list,
>
> We had OpenSI
the problem is MI command ul_add. We use
datagram MI to add/remove permanent user locations via web interface.
Before we were using XML-rpc and we had similar problem. That's why we
moved to datagram.
Please, advice.
Thank you!
--
Stas Kobzar
Developeur VoIP / VoIP Developer
. But my email was bounced as I am not a member of the list.
Should I send it to OpenSIPS github issues?
--
Stas Kobzar
Developeur VoIP / VoIP Developer
___
Modulis.ca Inc.
# Bureau / Office: 514-284-2020 x 246
Email: s <http://firstname.lastname>tas.kob...@modulis.ca
Hello,
Another crash just happened.
Please, find attached gdb bt full.
Thank you,
On Mon, Aug 31, 2015 at 11:32 AM, Stas Kobzar <stas.kob...@modulis.ca>
wrote:
> Hello list,
>
> We had OpenSIPS crashed. Please, find attached crash full stack trace from
> core file attached.
Hello,
We had a couple another crashes which looks very much the same.
Please, see attached.
Thank you,
On Mon, Aug 31, 2015 at 12:00 PM, Stas Kobzar <stas.kob...@modulis.ca>
wrote:
> Hello,
> Another crash just happened.
> Please, find attached gdb bt full.
>
> Thank you,
@ovidiusas it works. thank you.
---
Reply to this email directly or view it on GitHub:
https://github.com/OpenSIPS/opensips/issues/537#issuecomment-111529146___
Devel mailing list
Devel@lists.opensips.org
@ovidiusas sorry, still have issues.
```
# opensips -V
version: opensips 1.11.5-notls (x86_64/linux)
flags: STATS: On, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, SHM_MEM,
SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN
Iancu
OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
On 04.06.2015 16:20, Stas Kobzar wrote:
Yes, I had the same idea first, but lookup function is not available in
local_route.
Thanks,
On Thu, Jun 4, 2015 at 9:15 AM, Newlin, Ben ben.new...@inin.com wrote:
Can you
@bogdan-iancu
Thank you, Bogdan.
Also I tried to put many \n after Event header and seems like they are all
trimmed.
The same thing happens to my body of message-summary NOTIFY.
---
Reply to this email directly or view it on GitHub:
@bogdan-iancu
This problem here is not about the double Via.
This issue is about a missing terminating character in the header Event after
value check-sync
Please, see the SIP packets in the description I got with ngrep.
The first onw has a problem and another one does not.
Could you
-solutions.com
On 04.06.2015 02:00, Newlin, Ben wrote:
It sounds like you may be sending the NOTIFY to yourself when you use
the domain name instead of the IP. Have you verified the address that the
domain resolves to? Is it the same as the OpenSIPS instance?
Ben Newlin
From: Stas Kobzar
.
Ben Newlin
From: Stas Kobzar
Reply-To: OpenSIPS users mailling list
Date: Wednesday, June 3, 2015 at 8:45 PM
To: OpenSIPS users mailling list
Subject: Re: [OpenSIPS-Users] Double via in MI generated NOTIFY
Yes, it resolves to itself. The domain is the one from OpenSIPS domains
. Have you verified the address that the
domain resolves to? Is it the same as the OpenSIPS instance?
Ben Newlin
From: Stas Kobzar
Reply-To: OpenSIPS users mailling list
Date: Wednesday, June 3, 2015 at 6:00 PM
To: Bogdan-Andrei Iancu
Cc: OpenSIPS users mailling list
Subject: Re
modify the RURI/DURI or
other parts of the requests?
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
On 01.06.2015 17:42, Stas Kobzar wrote:
Hello,
I am sending NOTIFY packet with event check-sync to reload phone
configuration.
I am doing
IP whenever NOTIFY is received and I still have this problem.
Is it a known issue?
Everything still works fine, but having a double Via in logs a bit
confusing.
Thank you,
--
Stas Kobzar
Developeur VoIP / VoIP Developer
___
Modulis.ca Inc.
# Bureau / Office: 514-284-2020 x
Hello,
I have a bug with mi_xmlrpc_ng module when using tm module MI function
t_uac_dlg.
I am using it to restart phones and manage MWI.
Python script example to reproduce problem (I have the same problem with ruby
xmlrpc client):
```
#!/usr/bin/env python
import xmlrpclib
opensips =
@bogdan-iancu Right! Thank you for explaining that in details. I have updated
functions.
---
Reply to this email directly or view it on GitHub:
https://github.com/OpenSIPS/opensips/pull/517#issuecomment-103946920___
Devel mailing list
With improved memory management patch from bogdan-iancu.
You can view, comment on, or merge this pull request online at:
https://github.com/OpenSIPS/opensips/pull/517
-- Commit Summary --
* httpd: Make sure httpd can recognize Content-Type headers with params.
* Make sure httpd can
Hi Bogdan,
Thank you.
I have created a new pull request with the patch you have sent:
https://github.com/OpenSIPS/opensips/pull/517
Please, delete this one.
If it is ok, should I also do a cherry-pick for that commit to branches master
and 2.1?
Regards,
Stas
---
Reply to this email directly
I did the tests on my OpenSIPS server.
---
Reply to this email directly or view it on GitHub:
https://github.com/OpenSIPS/opensips/pull/506#issuecomment-103646584___
Devel mailing list
Devel@lists.opensips.org
@bogdan-iancu
Thanks for your comment, Bogdan.
Can you give me a hint how can I fix this issues? Or may be there is an example
in existing code?
Thank you,
Stas
---
Reply to this email directly or view it on GitHub:
Cherry-pick from master branch to 1.11
You can view, comment on, or merge this pull request online at:
https://github.com/OpenSIPS/opensips/pull/506
-- Commit Summary --
* httpd: Make sure httpd can recognize Content-Type headers with params.
-- File Changes --
M
Content-Type headers like text/xml; charset=utf-8 was not recognized. Some
libs (like built-in ruby xmlrpc client) force these additional params.
You can view, comment on, or merge this pull request online at:
https://github.com/OpenSIPS/opensips/pull/411
-- Commit Summary --
* httpd: Make
It is done. Please close this pull request and check #411
https://github.com/OpenSIPS/opensips/pull/411
Thank you
---
Reply to this email directly or view it on GitHub:
https://github.com/OpenSIPS/opensips/pull/408#issuecomment-74120588___
Devel
that for you via the trap
command. Of course, you can do the same with a small shell script doing
gdb+bt for all opensips processes.
Regards,
Bogdan-Andrei Iancu
OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
On 25.07.2014 21:30, Stas Kobzar wrote:
Hello List,
We have
uac_auth.so
uac.so
uri.so
usrloc.so
Does anyone had any similar problem? Any idea where to search?
Thank you,
--
Stas Kobzar
VoIP Developer
514 284 2020
www.modulis.ca
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin
You can merge this Pull Request by running:
git pull https://github.com/staskobzar/opensips master
Or you can view, comment on it, or merge it online at:
https://github.com/OpenSIPS/opensips/pull/150
-- Commit Summary --
* opensipsctl is calling ds_relaod for dr instead of dr_reload
Instead of reloading dynamic routes, script is trying to reload dispatcher
gateways. Example command to reproduce problem:
# opensipsctl dr addgw 1 10 192.168.1.11 0 0 main_gw
Returns: 500 command 'ds_reload' not available
---
Reply to this email directly or view it on GitHub:
Thank you, Ovidiu!
On Mon, Dec 2, 2013 at 2:04 PM, Ovidiu Sas o...@voipembedded.com wrote:
If you use the default top hiding scenario, it should work.
Regards,
Ovidiu Sas
On Mon, Dec 2, 2013 at 2:00 PM, Stas Kobzar stas.kob...@modulis.ca
wrote:
Hello list,
I am trying
for any help or hint!
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___
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Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
.
Is it possible to use chan_sip methods in other modules? If yes, could you,
please, give me a hint where to look?
Thank you,
--
Stas Kobzar
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk
Thank you!
On Sun, Dec 1, 2013 at 9:02 PM, Joshua Colp jc...@digium.com wrote:
Stas Kobzar wrote:
Hello list,
I am trying to develop my own Asterisk module.
I need to create and send PUBLISH SIP message with special headers
and/or message body.
I found in that in include folder
Hello List,
Is there a way to use shared variables in modules' configuration?
For example:
modparam(cfgutils, shvset, DBURI=s:mysql//user:pass@host/accdb)
modparam(acc, db_url, $shv(DBURI))
does not work for me:
ERROR:core:db_check_api: module db_$shv(DBURI) does not export db_use_table
Hi,
If you experience this problem with IE but everything is ok in Fierfox,
then try to check if server has correct time/date set.
Make sure that you have correct Default timezone set in php.ini.
Regards,
Stas
Dear List,
Recently I downloaded the new 4.2.2 version. I worked all well, but
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