Dear All,
I have installed Opensips latest rel1.7 on centos 5.7 32bit, HP
microserver 1.3kHz 2-cores, 2GB of RAM.
That server has got two interfaces configured X1.X1.X1.X1 and X2.X2.X2.X2
Configuration uses force_send_socket and rewritehost commands to
direct all calls coming from Y.Y.Y.Y on inter
Hello,
First of all, i am sorry for long delay - i was unable to keep on
working on this.
Thank You for your replies.
Logan: there is no DB back end at all in my configuration.
It is not necessery for me and as you mentioned could cause delays.
Vlad: i was trying to specify only IP addresses and
Hello,
Here is an output from opensips.log file
=rtpmap:0 PCMU/8000 087637 IN IP4 127.0.0.16SIPpTag002:
WARNING:core:log_expiry: threshold exceeded : msg processing took too
long - 309013 us.Source : INVITE sip:21133@X1.X1.X1.X1:5060
SIP/2.0
=rtpmap:0 PCMU/8000 087637 IN IP4 127.0.0.16SIPp
xlog("L_INFO","[INFO] incoming INVITE $si");
rewritehost("Z.Z.Z.Z");
insert_hf("X-Access-IP: $si \r\n");
}
}
if (!t_relay()) {
sl_reply_error();
};
ex
Thanks to Vlad the issue is solved.
Syslog was not in async mode and that was were the problem was located.
(http://stackoverflow.com/questions/208098/can-syslog-performance-be-improved)
Thanks again,
Maciej.
___
Users mailing list
Users@lists.opensips
Hello,
What is the best way to replace or modify Via header of incoming INVITE?
I need to change private ip address with $si.
Oryginal header is Via: SIP/2.0/UDP 10.10.10.128:5060;branch=z9hG4bK-680826
Is it subst? What is your advice?
Regards,
Maciej
___
Thanks Dani for prompt feedback.
I will take a look on this.
Maciej.
2012/1/10 Dani Popa :
> none,
> I think you want and need to use topology_hiding() from dialog module.
>
> Dani
>
> On Tue, Jan 10, 2012 at 1:21 PM, Maciej Bylica wrote:
>>
>> Hello,
>>
&
Hello,
That is my first post here :)
I am playing around with NAT traversal and mediaproxy on opensips
1.6.3. (media-dispatcher 2.4.3, media-relay 2.4.3, python 2.6)
I've just encounterd a problem with my configuration that really worries me.
Here is my script:
# main request routing logic
route
Hello,
I am playing around with DRouting module plus my opensips 1.6.3-notls.
The wiki is quite nice written, but i have some doubts here.
My config is pretty simple.
if ($rU=~"^"){
...
...
route(1) }
if ($rU=~"^"){
...
...
route(2) }
...
route[1] {
if (!do_routing("1"))
Hi Bogdan,
>> 3) prefix is char(64), could I use * char there?
>>
> only numerical prefixes are accepted . If you want to define a rule to
> match all prefixes (wildcard), simpy use a an empty string prefix.
>
I meant, how to define a star char '*'?
Entry '*3 ' for dialed *3999 is not working.
Th
Bogdan,
> only digits are accepted. So you can:
> 1) remove the starting * before doing do_routing()
> 2) replace * with a digit (like 0)
>
This is exactly what i am doing now.
I need to find out some examples here to tune up my routeid.
Thanks Bogdan for clearing this up.
Maciej.
___
Hello
I have a question regarding $fU pseudo variable.
As mentioned at http://www.opensips.org/Resources/DocsCoreVar16 and on
the basis of opensips outputs:
ERROR:dialplan:dp_trans_fixup: the output PV is read-only!!
it clearly means that $fU is read-only.
Unfortunately it is quite big problem fo
>> Use replace_from :-)
>>
>> ciao
>> s
>>
>> Il 10/10/2010 19:19, Maciej Bylica ha scritto:
>>
>>> Hello
>>>
>>> I have a question regarding $fU pseudo variable.
>>> As mentioned at http://www.opensips.org/Resources/Do
Hello.
I am planning to provide opensips with a kind of mechanism to manage
customer services/features like call-forward/VM/follow-me and so on.
It should work in following way: If $rU is provided in subscriber
table then user enabled service name is obtained from some db table.
On the basis of th
Hi,
Have anyone tried to use usr_preferences, AVPops to determine the
service to be fetched by the script?
Then i am planning to use switch statement to add different prefixes
before the called number and t_relay to asterisk server to do the
rest.
Is this proper point of view?
Thx,
Maciej.
>
to the right function.
>
> Maciej.
>
> 2010/10/11 Bogdan-Andrei Iancu :
>> To be more precise:
>> http://www.opensips.org/html/docs/modules/1.6.x/uac.html#id228582
>>
>> Regards,
>> Bogdan
>>
>> Stefano Pisani wrote:
>>> Use replac
oute[10] to be used whenever i wish in my
> script.
>
> Thx,
> Maciej.
>
>
>> Bogdan, Stefano,
>>
>> Its working as is should :)
>> Thanks for pointing me to the right function.
>>
>> Maciej.
>>
>> 2010/10/11 Bogdan-Andrei Iancu :
>
script.
>>
>> Thx,
>> Maciej.
>>
>>
>>> Bogdan, Stefano,
>>>
>>> Its working as is should :)
>>> Thanks for pointing me to the right function.
>>>
>>> Maciej.
>>>
>>> 2010/10/11 Bogdan-Andrei Iancu
Hi Bogdan,
I've already installed Avpops, it works nice...
I fully agree, the scenario You've covered is in my wish list :)
Thanks for help,
Maciej
> Hi Maciej
>
> Maciej Bylica wrote:
>> Hi,
>>
>> Have anyone tried to use usr_preferences, AVPops to dete
Hello,
I am working on opensips 1.6.3 $Revision: 4448 together with
media-dispatcher 2.4.3, media-relay 2.4.3, python 2.5.2-3, freeradius
2.1.8, radiusclient-ng 0.5.6
Freeradius should gather radius messages directly from opensips and
dispatcher. Both are installed on the same server and use the s
> and radius client are not the sameIt is not an opensips issue, it is a
> matter of configuring the radius server and radius client library.
>
> Regards,
> Bogdan
>
> Maciej Bylica wrote:
>>
>> Hello,
>>
>> I am working on opensips 1.6.3 $Revision: 4448 t
" is not "opensips acting as dispatcher", what is
> this "dispatcher" ???
>
> Regards,
> Bogdan
>
> Maciej Bylica wrote:
>>
>> Hi Bogdan,
>>
>> >From my point of view it is not so clear, because opensips and
>> dispatcher use the
Nevertheless, thank You Bogdan.
Has Anybody more less similiar problem like me?
Thx,
Maciej.
2010/11/12 Bogdan-Andrei Iancu :
> I see...
>
> unfortunately I cannot help you with media-dispatchernever used it :-/
>
> Regards,
> Bogdan
>
> Maciej Bylica wrote:
>>
Dear ALL,
During clearing my misconfigurations I found following errors in log file:
ERROR:uri:check_username: No authorized credentials found (error in scripts)
ERROR:uri:check_username: Call {www,proxy}_authorize before calling
check_* functions!
After closer look it turnes out that it is gener
Guys
So the only wayout is to request my SIP operator to be complied with
the standards?
Thanks
Maciej
> Dear ALL,
>
> During clearing my misconfigurations I found following errors in log file:
> ERROR:uri:check_username: No authorized credentials found (error in scripts)
> ERROR:uri:check_user
Iñaki,
Thank You for clearing a few things up.
Yes You are absolutely right with all signalization aspect, but
transaction identifier inspection really makes me think and again i
see that there are a lot aspects i need to take care of.
Could you pls point me to some hints describing the proper way
Iñaki
> It's well explained in RFC 3261.
> An ACK for a [3456]XX response must have same branch and same CSeq
> number (but "ACK" method) as the INVITE of the transaction.
I meant some hints regarding script configuration, because as far as i
understand i should double check my .cfg
Okay i may p
if you want the auth ACK, if the ACK does not have an Authorize hdr
> from beginning (as RFC sais) you cannot do much about it.
>
> Regards,
> Bogdan
>
> Maciej Bylica wrote:
>>
>> Iñaki
>>
>>
>>
>>>
>>> It's well explained in R
Hello,
It is quite old post, but i have just encoutered quite similiar problem.
I have the latest revision installed $Revision: 4448 in my server.
Opensips is starting itself properly:
# ps -ef | grep opensips
root 20982 6115 0 02:01 pts/100:00:00 gdb /usr/local/sbin/opensips
root 2
etstat -ulnp" to see where opensips is listening and
> if there is any pending data to be read.
>
> Regards,
> Bogdan
>
> Maciej Bylica wrote:
>>
>> Hello,
>>
>> It is quite old post, but i have just encoutered quite similiar problem.
>> I have the
Hi.
I am running Opensips 1.6.3 and trying to do topology hiding.
This is my scenario:Operator_1 -- > my Opensips --> Operator_2
The goal is not to convey any information of Operator_2 to Operator_1
like Contact, User-Agent headers and so on and to do rtp proxying.
For rtp proxying i'v
Hi,
> I am running Opensips 1.6.3 and trying to do topology hiding.
> This is my scenario: Operator_1 -- > my Opensips --> Operator_2
> The goal is not to convey any information of Operator_2 to Operator_1
> like Contact, User-Agent headers and so on and to do rtp proxying.
> For rtp pr
t; On Sun, Feb 6, 2011 at 9:23 AM, Maciej Bylica wrote:
>> Hi,
>>
>>> I am running Opensips 1.6.3 and trying to do topology hiding.
>>> This is my scenario: Operator_1 -- > my Opensips --> Operator_2
>>> The goal is not to convey any informat
Hello,
Does anyone knows how to change the server header content the proxy
presents itself.
For answering the call i have:
SIP/2.0 100 Giving a try
Via: SIP/2.0/UDP
11.22.33.44:5060;branch=z9hG4bK6e7776ee5332c5d8a782609dda3550a4;rport=5060
From: ;tag=164494841543104e5b53d38
ocs; there is a Server header you can set;
>>
>> I am not in front of it;
>>
>>
>>
>> On 2/7/11 7:42 PM, Maciej Bylica wrote:
>>>
>>> Hello,
>>>
>>> Does anyone knows how to change the server header content the proxy
>>> p
Dear OS Fans,
I've just managed to configure new acc with dialog cdr generation
feature with Mysql.
It looks fine and realy help to do accouting for some of us.
In my scenario there is a need to use Radius.
As stated in acc module description, there is a need to use cdr_flag
and setflag in initial
Hi,
Do you have any experience in this?
Thx,
Maciej.
> I've just managed to configure new acc with dialog cdr generation
> feature with Mysql.
> It looks fine and realy help to do accouting for some of us.
> In my scenario there is a need to use Radius.
> As stated in acc module description, the
omes with the OpenSIPS sources. Are you using that provided
> dictionary ?
>
> Regards,
> Vlad
>
>
> On 04/17/2011 11:38 PM, Maciej Bylica wrote:
>>
>> Dear OS Fans,
>>
>> I've just managed to configure new acc with dialog cdr generation
>> f
Hello,
I am in need to build a box which should have some functions of SBC.
To be more precisely server will be heaving two interfaces, the first
one i could say on access side, the last one on core/private side. I
want to implement a kind of call limitation mechanism (by using pike
module as a tr
80\." || $si=~"^10\.10\.10\." {
> force_send_socket(udp:108.109.180.12:5060);
> }
>
>
>
> t_relay();
>
>
>
>
> Best Regards
>
> Max M.
>
> Am 14.10.2011 11:26, schrieb Maciej Bylica:
>>
>> Hello,
>>
>> I am in need to buil
Hi All,
I am looking for a class 5 platform (basic VAS) and softphone (IOS,
Android) both supporting ZRTP protocol to achieve the highest voice
security.
C.5 and UA should be delivered from the same supplier (like sipwise for
instance)
Could anybody recommend me any solution here?
Thanks in adva
Hello
I am struggling with memcached installation with the latest git opensips
2.2.2 and centos 6.8
Here are version releases i am using:
libmemcached-1.0.18 (./configure, make && make install)
memcached-1.4.33 (./configure, make && make install)
with LD_LIBRARY_PATH=/usr/local/lib:$LD_LIBRARY_PAT
the memcached stuff -
> any special reason for doing that ? (versus using packages)
>
> As the problem seems to be in the lib, not in the OpenSIPS module.
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 09.1
he problem is located.
Thanks
Maciej.
2016-11-15 18:09 GMT+01:00 Bogdan-Andrei Iancu :
> OK, thank you for the update Maciej,
>
> Best regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 15.11.2016 18:28, Macie
ing a a query on the second key ?
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 18.11.2016 19:53, Maciej Bylica wrote:
>
> Hello
>
> As i mentioned before memcached is already installed. I am using
Ok, i figured it out, that the problem relies in port number definition.
I am getting no issues with 11211.
Thanks
Maciej
2016-11-21 22:20 GMT+01:00 Maciej Bylica :
> Hi Bogdan,
>
> Thanks for the reply.
>
> It seems it is related to the key, it doesn't matter which query
ad you solved it. I mean it is weired (with the
> wrong port) why it worked for some and did not for other keys :-/
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 21.11.2016 23:44, Maciej Bylica wrote:
>
> Ok,
Hello,
I am looking for call generator that is capable of:
- generating and in the same time pick up the call (the call will traverse
infrastructure under testing and get back to generator)
- generating SIP + RTP calls. There must be many .wav or mp3 files possible
to be used
- heaving random call
On Jun 18, 2013, at 7:03 PM, Maciej Bylica wrote:
>
> Hello,
>
> I am looking for call generator that is capable of:
> - generating and in the same time pick up the call (the call will traverse
> infrastructure under testing and get back to generator)
> - generating SIP + RT
Hello,
I have a problem to verify and change headers in OK message that Opensips
is receiving within the dialog by using insert_hf and search functions.
The problem is not with these functions but to catch OK that is a part of
the sip dialog.
Any changes are applied to INVITE unfortunately.
Is th
r your desired replies and do whatever you want to do with
> them. It should always work, that's what reply_route is designed to do...!
>
> Thank you.
>
>
> On Sat, Jul 13, 2013 at 9:33 PM, Maciej Bylica wrote:
>
>> Hello,
>>
>> I have a problem to verify an
Hello,
I have the same problem on 1.9 rel.
++-+--+---++
| id | username | domain | groupid | description |
++-+--+---++
| 4 | .* | 10.10.10.5| 0 | TEST
|
++-
Hello,
Thanks for reply.
Yeah i did it by asking db for..
avp_db_query("SELECT groupid FROM dr_groups WHERE domain =
'$fd'","$avp(i:600)");
and then using exactly the same avp for do_routing.
It works, but i am still wondering how to match domain different way (
do_routing())
Thanks.
2014-02-25
example.
>
> С уважением,
> Александр Мустафин
> mustafin.aleksa...@gmail.com
>
>
>
> 26 февр. 2014 г., в 20:45, Maciej Bylica написал(а):
>
> Hello,
>
> Thanks for reply.
> Yeah i did it by asking db for..
> avp_db_query("SELECT groupid FROM dr_groups WHERE
Hello
I just want to know how to achieve miliseconds precision for accounting
module.
This is quite important while trying to sum up total traffic duration with
the accuracy of hundred of ms.
As i see there is no rounding feature implemented as well, but heaving ms
precision it could be done dire
database level with a trigger or
> auto-update column.
>
>
>
> On Thu, Apr 10, 2014 at 10:01 AM, Maciej Bylica wrote:
>
>> Hello
>>
>> I just want to know how to achieve miliseconds precision for accounting
>> module.
>> This is quite important
your acc table, use the $time var
> with a format such as "%s.%N" or similar.
>
> Or, as you suggested, do it on the database level with a trigger or
> auto-update column.
>
>
>
> On Thu, Apr 10, 2014 at 10:01 AM, Maciej Bylica wrote:
>
>> Hello
>>
&g
case I
> assign it in the main routing section so the timestamp indicates the start
> of the transaction.
>
> best regards,
> Ryan
>
>
>
> On Fri, Apr 11, 2014 at 10:06 AM, Maciej Bylica wrote:
>
>> Ryan,
>>
>> One
>
> Best Regards,
>
> Vlad Paiu
> OpenSIPS Developerhttp://www.opensips-solutions.com
>
> On 12.04.2014 23:44, Maciej Bylica wrote:
>
> Hello Ryan,
>
> I am using dialog accounting, so each row is fully qualified cdr record,
> not only single transaction of a
de.
I think $avp(sec),$avp(usec) will be overwritten.
So maybe wise idea will be to set some flag in first 200 message and make
another statement like if ((t_check_status("200")) && !(isflagset(XX)))
What do you think about p1 and p2?
Thanks
Mac
2014-04-14 12:56 GMT+02:00 Maci
7 GMT+02:00 Adrian Georgescu :
> There are such tools but it depends for what purpose. Do you want to test
> heavy load or just call flows?
>
> Adrian
>
> On Jun 18, 2013, at 7:03 PM, Maciej Bylica wrote:
>
> Hello,
>
> I am looking for call generator that is capable
xlog("L_INFO","[INFO] Inside okay -> $var(okay)");
get_timestamp($avp(sec),$avp(usec));
}
Thanks
Mac.
2014-04-15 17:04 GMT+02:00 Maciej Bylica :
> Hello,
>
> It works, but:
> 1) get_timestamp doesnt work inside has_totag
Frankly such precision is not needed.
As i saw call duration is rounded mathematically, but sometimes in telco
world (my case) 0.1sec call should be counted as 1sec call.
Thats why i wanted to have milisec precision to be able to round durations
by myself...(1.01 = 2secs, 1.49 = 2secs, 1.99=2secs,
Hi
Right, i need ceiling function = to get smallest integral value not less
than argument.
Thanks
that's not "round", that's "ceiling"
> ceil(0.0001,0)= 1
> round(0.0001,0)= 0
>
>
> 2014-04-29 19:22 GMT-03:00 Maciej Bylic
Could somebody tell me a few words answering on my questions?
Thanks.
2014-04-30 12:46 GMT+02:00 Maciej Bylica :
> Hi
>
> Right, i need ceiling function = to get smallest integral value not less
> than argument.
>
> Thanks
>
>
> that's not "round", tha
Hello,
I just want to setup Opensips as SIP Proxy node.
Release 1.11.2-notls and DRouting module is already in place.
I just want to ask you what do you think about Contact header modification
in such case.
Some of my incoming INVITEs have only Contact header (describing
originator, like IPPABX fo
Hi,
Guys could i ask you to share your experience here
Thanks.
2014-09-25 23:00 GMT+02:00 Maciej Bylica :
> Hello,
>
> I just want to setup Opensips as SIP Proxy node.
> Release 1.11.2-notls and DRouting module is already in place.
> I just want to ask you what do you thin
o change the Contact
> header, you have to use topology-hiding, either the one provided by the
> dialog module, or the B2B module.
>
> Best regards,
>
> Răzvan Crainea
> OpenSIPS Solutionswww.opensips-solutions.com
>
> On 09/29/2014 01:33 AM, Maciej Bylica wrote:
Hello
I am struggling with OpenSIPS-CP 8.3.0 (.zip source) configuration on
Centos 8.2
Opensips 3.1 uses port 8000 to interop with opensips-cp, but there are no
tcpdump packets on that port.
It turned out that i am getting following errors on php level:
[30-Nov-2020 14:13:54 UTC] PHP Warning:
Hello
Could somebody please point me to where I should look for the clue ?
Thanks
Maciej
pon., 30 lis 2020 o 15:51 Maciej Bylica napisał(a):
> Hello
>
>
> I am struggling with OpenSIPS-CP 8.3.0 (.zip source) configuration on
> Centos 8.2
>
> Opensips 3.1 uses port
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