Hello,
I use Opensips 1.6 and 1.8.2 and when I dial I must wait 30 seconds
for the call
was successful!!
Can you help me please?
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*Hi,
I use this script to block an user declared in userblack list to calls an
number with prefix 01 but it's not works
$avp(fu) = $fU;
$avp(fd) = $fd;
$avp(rU) = $rU;
if ( !check_user_blacklist( "$avp($fu)", "$avp($fd)" ,"$avp(rU)" )
) {
sl_send_reply("403", "Forbi
Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
>
> On 10/26/2012 12:21 PM, Engineer voip wrote:
>
> Hi,
> I'm restarting my opensips server this morning and i get this error:
>
> ERROR:core:tcp_init: bind(7, 0xb725a7dc, 16) on 172.10.
Hi,
I'm restarting my opensips server this morning and i get this error:
ERROR:core:tcp_init: bind(7, 0xb725a7dc, 16) on 172.10.10.10:5060 : Address
already in use
- I did not make any modification
can you help me please?
--
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___
Users
*Hi,
Someone can help me please??*
Hi,
I'm trying to limit a calls of an user by using the script above
route[39]
{
if(!isflagset(31))
{
if( avp_db_load("$fu/username","a") &&
avp_check("$avp(channels)", "ge/0"))
{
# get c
Hi,
I'm trying to limit a calls of an user by using the script above
route[39]
{
if(!isflagset(31))
{
if( avp_db_load("$fu/username","a") &&
avp_check("$avp(channels)", "ge/0"))
{
# get current calls for user
Hello,
Can we integrate usr_preferences table in OCP to manage it ?
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Hello,
Can we integrate usr_preferences table in OCP to manage it ?
--
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Hi All,
i have resolved my problem of time by using$time(%Y%m%d%H%M).
thank you all
2012/10/19 Engineer Voip
> Hi,
> Iwant to convert 2012-10-19 09:00:00 to secondes
>
>
> Cordialement.
> Envoyé de mon iPhone
>
> Le 19 oct. 2012 à 09:01, "Gohar Ahme
p “2012-10-19 09:00:00”!
> Do you mean to extract only the “09:00:00” part !
>
> Regards,
> Gohar
>
> From: users-boun...@lists.opensips.org
> [mailto:users-boun...@lists.opensips.org] On Behalf Of Engineer Voip
> Sent: Friday, October 19, 2012 11:45 AM
> To: users@
Hi,
Can we convert this time 2012-10-19 09:00:00 to timestamp in opensips.cfg
without using an external script?
Cordialement.
Envoyé de mon iPhone
Le 18 oct. 2012 à 17:51, Engineer voip a écrit :
> Hi,
> I have resolved my probleme, now i stocked this time 2012-10-18 17:50
w how do it?
2012/10/18 Engineer voip
> Hi,
> I trying to do that with avpops module and usr_preferenses table.
> my script is:
>
> if ( avp_db_load("$ru/username","a") )
> {
> xlog("L_INFO"
route(13);
}
*but the result is:
*
* - OFF LINE FORWARD phone number: 12
- OFF LINE FORWARD from time: 14
- OFF LINE FORWARD TO time: *
*Why i get
P (avp_db_load function).
>>
>> // Binan
>>
>> --
>> *Från:* Engineer voip
>> *Till:* Binan AL Halabi ; OpenSIPS users
>> mailling list
>> *Skickat:* torsdag, 18 oktober 2012 10:21
>> *Ämne:* Re: [OpenSIPS-Users] Tran
Muhammad Shahzad, dynamic routing i use it but just to routing a calls to
a gateway but not to number phone !
2012/10/18 Muhammad Shahzad
> Use dynamic routing module.
>
> Thank you.
>
> On Oct 18, 2012 8:54 AM, "Engineer Voip" wrote:
> >
> > Hello all
ange you got it from
> database.
>
> http://www.opensips.org/html/docs/modules/1.7.x/avpops.html
>
> // Binan
>
> --
> *Från:* Engineer Voip
> *Till:* users@lists.opensips.org
> *Skickat:* torsdag, 18 oktober 2012 8:53
> *Ämne:* [OpenSIPS-Use
Hello all,
I want to transfert the call to user C when user A calls user B in interval of
time for example: 11h-14h
I can do that by asterisk but i prefer to do it by opensips
It's possible to do that by opensips?
Cordialement.
Envoyé de mon iPhone
__
Hello All,
I have User A enregistred on my Opensips Server and i want to routing the
incomming calls for this User to an other phone number if some one call him
between 12h and 14h for example.
Have an idea to do that Please ??
--
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___
User
tection) function.
>> On Oct 8, 2012 11:17 PM, "Adrian Georgescu" wrote:
>>
>>> There is no standard for communicating signaling information from a
>>> non-IP network like GSM to a SIP based network. What you try to achieve is
>>> close to impossible
Hi,
Thank you all for the reply.
2012/10/9 Adam Raszynski
> I use the following code on all my production OpenSIPS servers.
> It's CPU friendly and avoids being spotted by bots searching for
> open-relay VoIP servers.
>
> route{
> # put it at the very beginning of route section
> if($u
ilar in OpenSIPs.
>>
>> I would be very curious to hear about other people's experiences using
>> the Pike module to block this type of traffic. For what it's worth, I've
>> seen attack traffic high enough in bandwidth to saturate a pretty beefy
>> internet c
Hello All,
I want to know if opensips is capable to detect the responder of a mobile
phone?
if yes, what can i do that!
** I explain what i want to do:*
I want to Opensips calls a phone number, but if the responder of this
phone is active, the opensips detect it and transfers the call to another
p
phone.
i hope this is clear
2012/10/8 Duane Larson
> Can you explain in more detail what you are trying to do?
>
>
> On Mon, Oct 8, 2012 at 8:32 AM, Engineer voip wrote:
>
>> Hello,
>> I do that and its good but if user A hangs up the call, the user B is
>> h
this can be done with
> OpenSIPS 1.8. See the append_branch() function at [1] .
>
> [1] http://www.opensips.org/Resources/DocsCoreFcn18#toc106
>
> Regards,
>
> Vlad Paiu
> OpenSIPS Developerhttp://www.opensips-solutions.com
>
>
> On 10/08/2012 12:00 AM, Engineer voip
opensips.cfg
> start put a condition $ua =~ "friendly-scanner". If matched return
> stateless some error.
> Other option is to use pike module.
> Another option is use fail2ban for opensips logs.
> More sophisticated options involve firewalls with IPS and IDS modules.
>
Hi All,
I receveid several packets of registration from a "friendly-scanner" on
my opensips server
how can i do to block that please??
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Hi All,
It is possible to call a standard (fix) phone and mobile phone at the same
time with OpenSIPS 1.8 ??
--
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___
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o what you are
> wanting to do.
>
>
> On Sun, Oct 7, 2012 at 3:14 PM, Engineer voip wrote:
>
>> Hi,
>> Yes, but what i do if i want to define a number of calls for each user?
>> For example: 5 calls of user A and 10 calls for user B.
>>
>> 2012/10/7 Duane L
t;
> http://www.opensips.org/Resources/DocsTutConcurrentCalls
>
>
>
> On Sun, Oct 7, 2012 at 1:45 PM, Engineer voip wrote:
>
>> Hello All,
>> I use opensips 1.8 and i want to limite a maximum calls for each user
>> enregistred on my opensips Server.
>> Someone
Hello All,
I use opensips 1.8 and i want to limite a maximum calls for each user
enregistred on my opensips Server.
Someone has an idea to do that please ?
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tml#id293148
>
> //Binan
>
>
> ------
> *Från:* Engineer voip
> *Till:* OpenSIPS users mailling list
> *Skickat:* fredag, 5 oktober 2012 12:46
> *Ämne:* Re: [OpenSIPS-Users] Failover routing
>
> Hi,
> thanks for the reply,
> can we do the failover in GW L
gwlists...
>
>
> Best Regards
>
>
> On 10/05/2012 11:56 AM, Nick Altmann wrote:
>
> Don't you forget use_next_gw() in failure_route?
>
> --
> Nick
>
> 2012/10/5 Engineer voip
>
>> Hello All,
>> In my rules table below i have declared two rules
Hi,
No, i have do that.
thanks
2012/10/5 Nick Altmann
> Don't you forget use_next_gw() in failure_route?
>
> --
> Nick
>
> 2012/10/5 Engineer voip
>
>> Hello All,
>> In my rules table below i have declared two rules to use 06 prefix, it's
>&g
Hello All,
In my rules table below i have declared two rules to use 06 prefix, it's
work good but if the GW List #3 don't works my Server opensips
don't failover to the next rule witch use GW List #6,#7,#1.
Someone has an idea to resolve this problem?
Thanks
--
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___
Thank you for response
2012/10/3 Binan AL Halabi
>
> hi,
>
> Use probing (at SIP level) so opensips does not need to send it many times
> to realize that the gateway is down.
>
>
> //Binan
>
>
> --
> *From:* Engineer voip
Hello All,
I use drouting module to routing calls, it's work good but when i have a
Gateway HS the opensips sends the calls to this Gateway four times before
to send it to
another Gateway like that:
*INVITE sip:10800806@x.x.x.x;user=phone, with session description
INVITE sip:10800806x
Hello,
i want to use it to do "do_routing" and using rules declared in dr_rules
table.
2012/10/2 Nick Altmann
> What do you want to do with it?
>
> --
> Nick
>
>
> 2012/10/2 Engineer voip
>
>> Hello All,
>>
>> can we use dr_groups tabl
Hello All,
can we use dr_groups table without controling a domain and username fields
like:
username domain Group ID
.* .*0
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Hi,
I think the easiest way is to use a script with cron and send an email
following
the result of the command fifo opensipsctl dr_status GatewayID without
using the databases.
2012/9/14 alexandre Moutot
> Hi,
>
> Thank you for your answer. Yes i can actively check via fifo interface. To
> have
Hi,
Thank you all for your help, i will trying to do that
2012/9/15 Binan AL Halabi
> hi Engineer VOIP,
>
> You can use what is called "database replication"
> You have 4 database servers : one must be the master and 3 are slaves.
> opensips-cp update the master data
> configured on one server & give that database's major credentials like
> Hostname, username, password on the other 3 server's OpenSIPS configuration
> files. Please try it out, I am sure it'll work.
>
> Regards,
>
> Faisal Rehman
>
>
> ---
Hello All,
I have 4 opensips servers and i want modify all the servers when i modify
one of them.
can we use one opensips-cp web site to manage multiple opensips servers??
thank you
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htt
Hello All,
I am using opensips 1.8 and when i do the calls between two softphones on
local i have this error : *server error occurred (2/SL).*
my server and spftphones have a locals IP
**
if(method == "CANCEL" || method == "ACK")
{
xlog("L_INFO", "method <$rm> New
Hi Muhammad,
I already installed my opensips, what can i do to recompile my opensips to
have TLS option ?
2012/8/31 Muhammad Shahzad
> You need to enable it at compile time.
>
> Thank you.
>
>
> On Fri, Aug 31, 2012 at 1:27 PM, Engineer voip wrote:
>
>> Hello
Hello all,
where i get the opensips 1.8 tls versions please?
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Hi all,
can you help me please
-- Forwarded message --
From: Engineer voip
Date: 2012/8/28
Subject: cdr in opensips 1.8
To: OpenSIPS users mailling list , users-request <
users-requ...@lists.opensips.org>
Hello All,
I configured OpenSIPS 1.8 to have cdr and after
ader to see if there is a tag parameter.
>>
>> loose_route() examines the route headers to decide what's the destination
>> address for the request message.
>>
>> You probably need to do some reading on sip to understand tags,
>> record-route and route headers
Hello All,
Please someone can explain me this functions loose_toute() and hats_totag()
i read the opensips docs but i don't understand
thank you
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Hi Alli,
thank you for reply
2012/8/29 Ali Pey
> Hi Engineer voip,
>
> Yes, of course it is. Check out the probe_mode field in dr_gateways table.
>
> Regards,
> Ali Pey
>
>
> On Wed, Aug 29, 2012 at 3:20 PM, Engineer voip wrote:
>
>> Hallo Ali,
>>
probing and that would fix it.
>
> Regards,
> Ali Pey
>
>
> On Wed, Aug 29, 2012 at 2:59 PM, Engineer voip wrote:
>
>> Hello All,
>>
>> When i enabling a GW (drouting) and after a few minutes the Gateway
>> becomes inactives
>>
>> wh
Hello All,
When i enabling a GW (drouting) and after a few minutes the Gateway becomes
inactives
what can i do to have my Gateway active all the time??
Thank you
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dr_carriers.
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
>
> On 08/29/2012 01:35 PM, Engineer voip wrote:
>
> Hello All,
>
> if i disable the GW1 in Gateways table but i use it in dr_carriers
> (dr_g
Hello All,
if i disable the GW1 in Gateways table but i use it in dr_carriers
(dr_gw_list) table
*and i want to know if the GW1 still receives calls from dr_carriers table.
* ??
thank you
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Hello All,
In dr_rules tables i don't understand what's the difference between
choising a GW declared in dr_gateways and GW declared in dr_carriers
someone can explain me please?
thank you
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Hello All,
I configured OpenSIPS 1.8 to have cdr and after making some calls between
two softphones I geted the cdr
but now I don't get it without any modification, and in cdrs table i have
cdr_id = 0 !!
someone have an idea please?
thank you
___
U
of db
> To: users@lists.opensips.org
> Date: Tuesday, August 28, 2012, 2:03 AM
>
>
> Hello,
>
> Then it means that you do not have the drouting.so module loaded.
>
> Regards,
>
> Vlad Paiu
> OpenSIPS Developerhttp://www.opensips-solutions.com
>
>
> On 0
rds,
>
> Vlad Paiu
> OpenSIPS Developer
> http://www.opensips-solutions.**com <http://www.opensips-solutions.com>
>
>
>
> On 08/28/2012 09:47 AM, Engineer Voip wrote:
>
>> Hello,
>> It is necerary to restart opensips 1.8 when we modifying a data base like
thank you nick
2012/8/28 Nick Altmann
> "gw1=70,gw2=30" in gwlist column of dr_carriers table.
>
> --
> Nick
>
> 2012/8/28 Engineer Voip
>
>> Hello,
>> It is possible to send for example 70% of calls to gw1 and 30% of calls
>> to a gw2 by using
Hello,
It is possible to send for example 70% of calls to gw1 and 30% of calls to a
gw2 by using drouting module in opensips 1.8
Thank you all
Cordialement.
Envoyé de mon iPhone
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Hello,
It is necerary to restart opensips 1.8 when we modifying a data base like
drouting tables?
I know that is necerary in opensips 1.6
Thank you
Cordialement.
Envoyé de mon iPhone
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Hello,
I have 2 GW and i want to send 50% of calls on each one by usine the
loadbalancer module but i don't have any idea to do that.
can you help me please !
thank you
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teway.
>
> Did you mean "prefix" of dr_rules table? These prefixes are used to match
> based on the request uri, but not based on from/to header.
>
>
>
> Best Regards
>
> Max M.
>
>
> On 08/27/2012 11:28 AM, Engineer voip wrote:
>
> Hello A
to
> the specific gateway.
>
> Regards,
>
> Vlad Paiu
> OpenSIPS Developerhttp://www.opensips-solutions.com
>
>
> On 08/27/2012 12:28 PM, Engineer voip wrote:
>
> Hello All,
>
> someone can tell me if the PRI_PREFIX fields of the dr_rules table is
&
Hello All,
someone can tell me if the PRI_PREFIX fields of the dr_rules table is used
to matching the "To" o "From" fields.
Thank you.
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On Sat, Aug 25, 2012 at 1:21 AM, Engineer voip wrote:
>
>> Hello,
>>
>> In opensips 1.6 we use LCR and GW tables,
>>
>> Can you tell me what is the equivalent tables in Opensips 1.8.?
>>
>>
>> Thanks
>>
>> _
quot;$fu","$avp(trace)")) {
> $avp(s:traceuser)=$fu;
> setflag(22);
> sip_trace();
> xlog("L_INFO","User $fu being traced");
> }
>
> The $avp(s:trace) will be loaded from usr_references table
> so you have to add the user :
&
url","mysql://opensips:opensipsrw@localhost/opensips")
Also i changed this ligne "mysql -h $HOSTNAME -u $USER -p$PASS -e
"call opensips_cdrs_1.6(); " $DATABASE"
in generate_cdrs.mysql to "mysql -h $HOSTNAME -u $USER -p$PASS -e
"call opensips_cdrs(); " $DA
Hello,
In opensips 1.6 we use LCR and GW tables,
Can you tell me what is the equivalent tables in Opensips 1.8.?
Thanks
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Hello All,
I want to get siptrace from opensips 1.8 using OCP but i haven't no
idea to use SIPTRACE
module.
someone have an idea to do that, please?
Best Regards.
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2, Ali Pey wrote:
>
> From: Ali Pey
> Subject: Re: [OpenSIPS-Users] cdr and Dialog on opensips-cp
> To: "Engineer voip"
> Cc: "OpenSIPS users mailling list"
> Date: Friday, August 24, 2012, 8:37 AM
>
> Did you get an error for the dlg_list command? D
nds:
>
> opensipsctl fifo arg
> opensipsctl fifo dlg_list
>
> and post the results here. This should give you a pretty good clue.
>
> Regards,
> Ali Pey
>
> On Fri, Aug 24, 2012 at 8:47 AM, Engineer voip wrote:
>
>> Hello,
>>
>> ** how can i verify t
ndrei Iancu :
> Hi,
>
> Are you 100% sure that the MI connector you put in CP
> ("/tmp/opensips_fifo") points to the server you actually want to
> interrogate for dialogs ?
>
> As I see the MI communication works (as you get the "command dlg_list is
> not availabl
user']="admin";
$boxes[$box_id]['monit']['pass']="admin";
$boxes[$box_id]['monit']['has_ssl']=0;
// description (appears in mi , monit )
$boxes[$box_id]['desc']="SIP server";
$boxes[$box_id]['assoc_id']=
HI,
So we must specify the username and domain for any group???
thanks
2012/8/23, Binan AL Halabi :
>
> Hi,
>
> Table dr_groups is used to associate routes to specific users.
>
> //Binan
>
> --- On Thu, 8/23/12, Engineer voip wrote:
>
> From: Engineer voip
> Su
Hi,
someone can help me please !!
2012/8/23, Engineer voip :
> Hello,
> I did all this procedures but don't work
>
> my dialog configuration is:
>
> ### dialog module
> loadmodule "dialog.so"
> modparam("dialog", "dlg_match_mode", 1
Hello All,
Why we can't adding two groups with the same "username" and "domain"
like ".*" in dr_groups table?
For example one groups is for the inbound calls and the other is for
the outbound calls
Thanks for your help.
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ate")
modparam("dialog", "start_time_column", "start_time")
Thank you for your help.
Regards
2012/8/23, Binan AL Halabi :
> Hi ,
> 1- for Dialog : you have to load & configure the
> module "dialog" in OpenSIPS script.
>
> 2- for CD
hello,
when i click "dialog" i get the message "
ERROR:mi_fifo:mi_fifo_server: command dlg_list is not available "
thank you for help
2012/8/23, Binan AL Halabi :
> Hi,
> what you get when you click "dialog" under "system" list on opensips-cp ?
>
Hello All,
I'm using opensips-cp with opensips 1.8, and i can do the call between two
users but i can't get the CDRs and Dialog informations by using
opensips-cp.
someone can help me, please?
thanks
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htt
Hello,
In 1.8 Version of Opensips you don't need to use "i:xx", use juste "xx"
without "i:"
Regards
2012/8/23 zhi sun
> Hi everyone,
>
> i am trying to setup opensips with freeswitch, and got the following
> error messages, when running ./opensips start
> [root@mydev init.d]# ./op
Hello,
how we enable TLS on opensips 1.8?
thanks
2012/8/23 Muhammad Shahzad
> You need to enable TLS support at compile type. It is not enabled by
> default.
>
> Thank you.
>
>
> On Thu, Aug 23, 2012 at 4:32 AM, vivid333 wrote:
>
>> why show notls? I need tls function. can anyone help me ? t
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