date:20130612

[OpenSIPS-Users] IPv6 and Opensips Version 1.5

2013-06-12 Thread Jonathan Hunter

Hi Guys,Hope your well?I'm currently testing IP version 6 and I'm having some 
issues adding IPv6 addresses into the address and gw tables.Is there any 
limitation on the support for them, or any particular steps or syntax 
required?I need to send and receive calls via IPv6, and I have Opensips 
listening on IPv6, and an IPv6 client connected, its just when I try and define 
some addresses, after increasing the character length in the columns that 
opensips will not start.Any support appreciated, Im currently running version 
1.5.ThanksJon___
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Re: [OpenSIPS-Users] CDRTool - Callcontrol - Prepaid

2013-06-12 Thread Leonardo Uzcudun

thanks a lot Laszlo!!
I did it, i've configured wrong in the opensips the $avp(can_uri).
Now I've fixed it and CDRTool is answering properly.
I'll continue testing the CDRToll & CallControl.
Thanks again.

Leo.


 Da: Laszlo 
A: Leonardo Uzcudun  
Cc: OpenSIPS users mailling list  
Inviato: Mercoledì 12 Giugno 2013 21:17
Oggetto: Re: [OpenSIPS-Users] CDRTool - Callcontrol - Prepaid
 


Hmm, this sounds challenging :)
I'm working on a custom opensips user portal and admin interface at the moment, 
and it is a "nice to have" feature here too. 
cdrtool is also used here, so I'll play with this feature on my dev server. 
Let's see if there can be a solution, or not.

-Laszlo
2013.06.12. 12:27, "Leonardo Uzcudun"  ezt írta:

Thanks Laszlo, good idea! but it still not working.
> 
>Now i can see in the cdrtool web portal that the canonical-uri is  (check 
>the attached file, there is a comparison before and after changing the can-uri 
>to ) but when the sip user samsung@sip.mydomain places the call i still 
>seeing in the syslog:
> 
>Jun 12 11:56:50 sip cdrtool[1304]: MaxSessionTime Duration=7200 
>CallId=1517185829 From=sip:samsung@sip.mydomain Gateway=82.186.X.X 
>To=sip:ipad@sip.mydomain
>Jun 12 11:56:50 sip cdrtool[1304]: MaxSessionTime=unlimited Type=prepaid 
>CallId=1517185829 BillingParty=samsung@sip.mydomain DestId=None
>Jun 12 11:56:50 sip call-control[1985]: Call id 1517185829 of 
>samsung@sip.mydomain to sip:ipad@sip.mydomainsip is postpaid not limited
>
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
>One more thing, with change (can-uri ) when the cdrtool does the 
>normalization i see in the syslog:
> 
> 
> 
> 
>Jun 12 11:59:27 sip cdrtool[7012]: Error: cannot find destinations for 
>subscriber='samsung@sip.mydomain', domain ='', gateway='', reseller='0'
>Jun 12 11:59:27 sip cdrtool[7012]: Error: cannot find destination id for  
>of customer = '', total destinations = 0
> 
> 
> 
>Do i configuring something wrong in the cdrtool web portal -> Rating -> 
>Destinations?
> 
>Is is normal to have domain=''? shouldn't be sip.mydomain?
> 
>Thanks a lot for your help,
> 
>Leo.
> 
>
>
>Da: Laszlo 
>A: Leonardo Uzcudun ; OpenSIPS users mailling list 
> 
>Inviato: Martedì 11 Giugno 2013 18:26
>Oggetto: Re: [OpenSIPS-Users] CDRTool - Callcontrol - Prepaid
>
>
>
>what happens if you manually set $avp(can_uri) to a fixed value? like to , 
>and define this destination in cdrtool?
>2013.06.11. 18:08, "Leonardo Uzcudun"  ezt írta:
>
>Did it but still not working. I'm still receiving Maxsessiontime=unlimited and 
>DetsId=none :(
>>
>>
>>Is there any "default destination" (like a default gateway)? So no matched 
>>destination will use the "default destination" with an X price/rate/profile?
>>
>>
>>
>>Da: Adrian Georgescu 
>>A: OpenSIPS users mailling list  
>>Inviato: Martedì 11 Giugno 2013 17:18
>>Oggetto: Re: [OpenSIPS-Users] CDRTool - Callcontrol - Prepaid
>>
>>
>>
>>You need to add pricing info in the other billing tables too.
>>
>>
>>Adrian
>>
>>On Jun 11, 2013, at 3:24 PM, Leonardo Uzcudun wrote:
>>
>>Thanks again Adrian.
>>>You mean to insert in destinations (CDRtool web portal -> Rating -> 
>>>Destinations) the use...@example.com in the destination field?
>>>
>>>
>>>
>>>Da: Adrian Georgescu 
>>>A: OpenSIPS users mailling list  
>>>Inviato: Martedì 11 Giugno 2013 14:32
>>>Oggetto: Re: [OpenSIPS-Users] CDRTool - Callcontrol - Prepaid
>>>
>>>
>>>
>>>Never tried it myself for prepaid but you can setup SIP URIs as valid 
>>>destinations too and put a price on them.  
>>>
>>>
>>>Adrian
>>>
>>>
>>>
>>>On Jun 11, 2013, at 12:29 PM, Leonardo Uzcudun wrote:
>>>
>>>I guess this confirm why i can't use prepaid for calls between sip users..., 
>>>do you confirm it?
Thanks. 


Da: Adrian Georgescu 
A: OpenSIPS users mailling list  
Inviato: Martedì 11 Giugno 2013 11:57
Oggetto: Re: [OpenSIPS-Users] CDRTool - Callcontrol - Prepaid



The prepaid engine was designed for PSTN destinations using phone numbers 
only.  


Adrian


On 11 Jun 2013, at 11:49, Leonardo Uzcudun  wrote:

Hello Adrian:
>
>From the following site 
>(http://callcontrol.ag-projects.com/issues/1769#change-8449) i've found 
>that you say "All internals of CDRTool rely on the fact that there is a 
>normalized URI in the form of 00 + Country Code + Number."
>So, my question or doubt is:
>If there are 2 sip users within the same domain like use...@example.com 
>and use...@example.com, can i implement prepaid billing between them or 
>the prepaid will only work if they call to 00 + Country Code + Number sip 
>accounts?
>Thanks,
>
>Leo 
>Da: Adrian Georgescu 
>A: OpenSIPS users mailling list  
>Inviato: Lunedì 10 Giugno 2013 21:39
>Oggetto: Re: [OpenSIPS-Users] CDRTool - Callcontrol - Prepaid
>
>
>
>There is Rating.txt document that contain sample rates f

[OpenSIPS-Users] ERROR:core:start_timer_processes

2013-06-12 Thread Dragomir Haralambiev

Hello,

I try to start opensips 1_9 in Rasperry Pi/Pidora/ like
http://www.raspberryvoip.com/running-opensips/


Here is log:

May 17 22:19:47 raspi /usr/local/sbin/opensips[553]: NOTICE:core:main:
version: opensips 1.9.1-notls (armv6l/linux)
May 17 22:19:47 raspi /usr/local/sbin/opensips[553]:
NOTICE:signaling:mod_init: initializing module ...
May 17 22:19:47 raspi /usr/local/sbin/opensips[553]:
CRITICAL:tm:lock_set_init: semget (..., 251, 0700) failed: Invalid argument
May 17 22:19:47 raspi /usr/local/sbin/opensips[553]:
WARNING:dialog:mod_init: hash_size is not a power of 2 as it should be ->
rounding from 7 to 4
May 17 22:19:47 raspi /usr/local/sbin/opensips[564]:
ERROR:core:start_timer_processes: failed to send status code


How to correct this ERROR?

Best regards,
PlayMen
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Re: [OpenSIPS-Users] CDRTool - Callcontrol - Prepaid

2013-06-12 Thread Laszlo

Hmm, this sounds challenging :)
I'm working on a custom opensips user portal and admin interface at the
moment, and it is a "nice to have" feature here too.

cdrtool is also used here, so I'll play with this feature on my dev server.
Let's see if there can be a solution, or not.

-Laszlo
2013.06.12. 12:27, "Leonardo Uzcudun"  ezt írta:

> Thanks Laszlo, good idea! but it still not working.
>
> Now i can see in the cdrtool web portal that the canonical-uri is 
> (check the attached file, there is a comparison before and after changing
> the can-uri to ) but when the sip user samsung@sip.mydomain places
> the call i still seeing in the syslog:
>
> Jun 12 11:56:50 sip cdrtool[1304]: MaxSessionTime Duration=7200
> CallId=1517185829 From=sip:samsung@sip.mydomain Gateway=82.186.X.X
> To=sip:ipad@sip.mydomain
> Jun 12 11:56:50 sip cdrtool[1304]: *MaxSessionTime=unlimited*Type=prepaid 
> CallId=1517185829
> BillingParty=samsung@sip.mydomain *DestId=None
> *Jun 12 11:56:50 sip call-control[1985]: Call id 1517185829 of
> samsung@sip.mydomain to sip:ipad@*sip.mydomainsip* *is postpaid not
> limited*
>
>
>
>
>
>
>
>
>
>
>
> One more thing, with change (can-uri ) when the cdrtool does the
> normalization i see in the syslog:
>
>
>
>
> Jun 12 11:59:27 sip cdrtool[7012]: Error: cannot find destinations for
> subscriber='samsung@sip.mydomain', domain ='', gateway='', reseller='0'
> Jun 12 11:59:27 sip cdrtool[7012]: Error: cannot find destination id for
>  of customer = '', total destinations = 0
>
>
>
> Do i configuring something wrong in the cdrtool web portal -> Rating ->
> Destinations?
>
> Is is normal to have domain=''? shouldn't be sip.mydomain?
>
> Thanks a lot for your help,
>
> Leo.
>
>
>   *Da:* Laszlo 
> *A:* Leonardo Uzcudun ; OpenSIPS users mailling list <
> users@lists.opensips.org>
> *Inviato:* Martedì 11 Giugno 2013 18:26
> *Oggetto:* Re: [OpenSIPS-Users] CDRTool - Callcontrol - Prepaid
>
>  what happens if you manually set $avp(can_uri) to a fixed value? like to
> , and define this destination in cdrtool?
> 2013.06.11. 18:08, "Leonardo Uzcudun"  ezt írta:
>
>  Did it but still not working. I'm still receiving
> Maxsessiontime=unlimited and DetsId=none :(
>
> Is there any "default destination" (like a default gateway)? So no matched
> destination will use the "default destination" with an X price/rate/profile?
>
>   *Da:* Adrian Georgescu 
> *A:* OpenSIPS users mailling list 
> *Inviato:* Martedì 11 Giugno 2013 17:18
> *Oggetto:* Re: [OpenSIPS-Users] CDRTool - Callcontrol - Prepaid
>
>  You need to add pricing info in the other billing tables too.
>
> Adrian
>
>  On Jun 11, 2013, at 3:24 PM, Leonardo Uzcudun wrote:
>
>  Thanks again Adrian.
> You mean to insert in destinations (CDRtool web portal -> Rating ->
> Destinations) the use...@example.com in the destination field?
>
>   *Da:* Adrian Georgescu 
> *A:* OpenSIPS users mailling list 
> *Inviato:* Martedì 11 Giugno 2013 14:32
> *Oggetto:* Re: [OpenSIPS-Users] CDRTool - Callcontrol - Prepaid
>
>  Never tried it myself for prepaid but you can setup SIP URIs as valid
> destinations too and put a price on them.
>
> Adrian
>
>  On Jun 11, 2013, at 12:29 PM, Leonardo Uzcudun wrote:
>
>  I guess this confirm why i can't use prepaid for calls between sip
> users..., do you confirm it?
> Thanks.
>
>   *Da:* Adrian Georgescu 
> *A:* OpenSIPS users mailling list 
> *Inviato:* Martedì 11 Giugno 2013 11:57
> *Oggetto:* Re: [OpenSIPS-Users] CDRTool - Callcontrol - Prepaid
>
>  The prepaid engine was designed for PSTN destinations using phone
> numbers only.
>
> Adrian
>
>  On 11 Jun 2013, at 11:49, Leonardo Uzcudun  wrote:
>
>  Hello Adrian:
>
> From the following site (
> http://callcontrol.ag-projects.com/issues/1769#change-8449) i've found
> that you say "All internals of CDRTool rely on the fact that there is a
> normalized URI in the form of 00 + Country Code + Number."
> So, my question or doubt is:
> If there are 2 sip users within the same domain like use...@example.com
> and use...@example.com, can i implement prepaid billing between them or
> the prepaid will only work if they call to 00 + Country Code + Number sip
> accounts?
> Thanks,
>
> Leo
>  *Da:* Adrian Georgescu 
> *A:* OpenSIPS users mailling list 
> *Inviato:* Lunedì 10 Giugno 2013 21:39
> *Oggetto:* Re: [OpenSIPS-Users] CDRTool - Callcontrol - Prepaid
>
> There is Rating.txt document that contain sample rates files and examples
> of prepaid queries in the console that shows you how it works.
>
> Adrian
>
>
>  On Jun 10, 2013, at 9:30 PM, Leonardo Uzcudun wrote:
>
>   Hello All:
>
> After doing more tests i guess the problem is the missing DestId
> cdrtool log:
> Jun  8 02:51:12 sip cdrtool[13359]: MaxSessionTime=unlimited Type=prepaid
> CallId=1226626922 BillingParty=user01@mydomain DestId=None
>
> and considering the rating.txt doc point 3. Determination of the costs ->
> No rate will be returned if no 'destination id' is found.
>
> So, now my question is: If the Destin

Re: [OpenSIPS-Users] (no subject)

2013-06-12 Thread Bogdan-Andrei Iancu

Hi Jorge,

First of all, why do you want to use b2b support if you only need topo
hiding and to terminate a call ? For these you can simply use the dialog
module, which is a much lighter approach (with less probability for
complications).

Have you consider to use the dialog module, with topology_hiding()
script function and dlg_end_dlg MI function ?

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com


On 06/12/2013 07:35 PM, Jorge Henrique Pinho wrote:
>
> Hi, I am using opensips with b2b module with topology hiding.
>
> I am trying to terminate a call when i receive the 200OK response to a
> Re-Invite. To accomplish this i am using the 'terminate_call' function
> defined in b2b_logic module.
>
> The dialog is establish and an user sends an in dialog Invite to
> renegotiate codecs, but the 200OK reply is mal formed. In consequence
> I need to terminate the call at this stage.
>
> To accomplish this I am using the terminate_call function present in
> b2b_logic module. When I receive the 200OK bad formatted message, I
> invoke the terminate_call function. The behavior of b2b is this :
>
> A - Opensips (B2B)  B
>
> A   Re-Invite à  | Re-Invite à  B
>
> A | 200 OK*  ß 
>   B *Bad formatted 200OK
>
> ABYE ß
>   | B
>
> A |  BYEà B
>
> A |  ACK
>à B
>
> A |  BYE
> à B
>
> It seems that b2b is sending a first BYE to B that was not supposed
> to. Log is in the attachment.
>
> Can you help me fix this?
>
> Kind regards
>
> Jorge Pinho
>
>
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Re: [OpenSIPS-Users] AVP on local route

2013-06-12 Thread Bogdan-Andrei Iancu

Hello Khaled,

Yes, the ACC will generate the cdr record after the local route triggering.

But if you want to push some additional info to the CDR, I advice you to
use the extra_db accounting (see
http://www.opensips.org/html/docs/modules/1.9.x/acc.html#ACC-extra-id)
for the PDD value.

Best regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com


On 06/12/2013 06:26 PM, M.Khaled W Chehab wrote:
>
> Hello Bogdan,
>
>  
>
> I am using cdr flag and I think that what is causing this problem,
> since the  update is done before the acc module insert the record.
>
> And I don't want to use acc start/stop ,is there a way to set a delay
> timer  after acc_db_request("200 Dialog Timeout", "acc");
>
> Or let the acc module insert the invite when call is connected ( 200 ok )
>
>  
>
>  
>
> Regards
>
>  
>
>  
>
> *From:*Bogdan-Andrei Iancu [mailto:bog...@opensips.org]
> *Sent:* Wednesday, June 12, 2013 1:19 PM
> *To:* OpenSIPS users mailling list
> *Cc:* M.Khaled W Chehab
> *Subject:* Re: [OpenSIPS-Users] AVP on local route
>
>  
>
> Hello Khaled,
>
> What kind of accounting are you using in script ? Based on CDR flag ?
> Based on ACC flag (start/stop) ? or manual accounting (with functions) ?
>
> Maybe the record you are trying to update in local route does not
> exist in ACC table yet (depending on the accounting method you are using).
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developer
> http://www.opensips-solutions.com
>
>
> On 06/03/2013 02:27 PM, M.Khaled W Chehab wrote:
>
> Hi,
>
>  
>
> The   execution for the avp_db_query is not working fine  in local
> route after the  acc_db_request  ,
>
>  
>
> As I print the query using xlog(" and   insert t it to myql ,the query
> executed normally which means I have no problem in the mysql query  .
>
>  
>
> My question: how can update the record in the acc table  after
> acc_db_request("200 Dialog Timeout", "
>
>  
>
>  
>
> local_route {
>
> if (is_method("BYE")  && $DLG_dir=="UPSTREAM") {
>
>  xlog("L_INFO", "--Debug Customer
> ID:$avp(Cusid)/IP:$si-Reply:
> $T_reply_code--Source:$rd\nDialed:$fU\n:");
>
>  
>
> acc_db_request("200 Dialog Timeout", "acc");
>
>  
>
> avp_db_query("UPDATE `opensips`.`acc` SET `pdd`='$var(pdd)' WHERE
> `CALLID` ='$ci' and `method`='INVITE'" );
>
>  
>
>  
>
> BTW: I cant find the acc record for the call unless  the call  ended
> ,and in my case I set the  acc_db_request("200 Dialog Timeout  then
> call avp_db_query(   is  that should insert the call in the acc to
> update it later
>
>  
>
> Regards
>
> ,.
>
>  
>
>  
>  
> ___
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Re: [OpenSIPS-Users] AVP on local route

2013-06-12 Thread M.Khaled W Chehab

Hello Bogdan,

 

I am using cdr flag and I think that what is causing this problem, since the
update is done before the acc module insert the record.

And I don't want to use acc start/stop ,is there a way to set a delay timer
after acc_db_request("200 Dialog Timeout", "acc"); 

Or let the acc module insert the invite when call is connected ( 200 ok )

 

 

Regards

 

 

From: Bogdan-Andrei Iancu [mailto:bog...@opensips.org] 
Sent: Wednesday, June 12, 2013 1:19 PM
To: OpenSIPS users mailling list
Cc: M.Khaled W Chehab
Subject: Re: [OpenSIPS-Users] AVP on local route

 

Hello Khaled,

What kind of accounting are you using in script ? Based on CDR flag ? Based
on ACC flag (start/stop) ? or manual accounting (with functions) ?

Maybe the record you are trying to update in local route does not exist in
ACC table yet (depending on the accounting method you are using).

Regards,



Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com


On 06/03/2013 02:27 PM, M.Khaled W Chehab wrote: 

Hi,

 

The   execution for the avp_db_query is not working fine  in local route
after the  acc_db_request  ,

 

As I print the query using xlog(" and   insert t it to myql ,the query
executed normally which means I have no problem in the mysql query  .

 

My question: how can update the record in the acc table  after
acc_db_request("200 Dialog Timeout", "

 

 

local_route {

if (is_method("BYE")  && $DLG_dir=="UPSTREAM") {

 xlog("L_INFO", "--Debug Customer
ID:$avp(Cusid)/IP:$si-Reply:
$T_reply_code--Source:$rd\nDialed:$fU\n:");

 

acc_db_request("200 Dialog Timeout", "acc");

 

avp_db_query("UPDATE `opensips`.`acc` SET `pdd`='$var(pdd)' WHERE `CALLID`
='$ci' and `method`='INVITE'" );

 

 

BTW: I cant find the acc record for the call unless  the call  ended ,and in
my case I set the  acc_db_request("200 Dialog Timeout  then call
avp_db_query(   is  that should insert the call in the acc to update it
later 

 

Regards

,.

 

 
 
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Re: [OpenSIPS-Users] registration t_replicate()

2013-06-12 Thread Bogdan-Andrei Iancu

Doing just before the exit, after the save() stuff.

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com


On 06/12/2013 03:50 PM, Miha wrote:
> Hi Bogdan,
>
> where exactly t_replicate() should be used?
>
> my script:
>
>  if (is_method("REGISTER") && is_from_local())
> {
>
> # make pua_usrloc send PUBLISH for phones which do
> # not support presence
> # filter after User-Agent header
> if(!search("^User-Agent: X-Lite")) pua_set_publish();
>
> ## --NAT-- ##
> if (isflagset(5)) {
> setbflag(6);
> setbflag(7);
> };
> ## --/NAT-- ##
>
> # authenticate the REGISTER requests (uncomment to
> enable auth)
> if (!www_authorize("", "subscriber"))
> {
> www_challenge("", "0");
> exit;
> }
> *t_replicate("sip:xxx.xxx.xxx.xxx:5060");*
> if (!db_check_to())
> {
> sl_send_reply("403","Forbidden auth ID");
> exit;
> }
> if (!save("location"))
> sl_reply_error();
>
> exit;
> }
>
>
> I am getting this:
>
> Jun 12 14:45:39 localhost /usr/sbin/opensips[12659]:
> ERROR:pua:send_publish_int: New 'presence' PUBLISH and no body found -
> invalid request
> Jun 12 14:45:39 localhost /usr/sbin/opensips[12659]:
> ERROR:tm:_reply_light: failed to generate 200 reply when a final 401
> was sent out
> Jun 12 14:45:39 localhost /usr/sbin/opensips[12659]:
> ERROR:signaling:sig_send_reply_mod: failed to send reply with tm module
> Jun 12 14:45:39 localhost /usr/sbin/opensips[12659]:
> ERROR:registrar:send_reply: failed to send 200 OK
>
> thanks!
>
> miha
>
>
>
>
> Dne 6/12/2013 12:07 PM, pis(e Bogdan-Andrei Iancu:
>> Hello Miha,
>>
>> Yes, t_replicate() is designed to be used after to finished the local
>> processing of the REGISTER (like doing the save(location)).
>>
>> This mechanism is intended to be used between servers in the same
>> platform, so you can use IP-based authentication for the replicated
>> REGISTERs (no need for digest auth). And you should not accept
>> replicated registers from anywhere, but only from servers from your
>> platform.
>>
>> Regards,
>>
>> Bogdan-Andrei Iancu
>> OpenSIPS Founder and Developer
>> http://www.opensips-solutions.com
>>
>>
>> On 06/11/2013 02:49 PM, Miha wrote:
>>> HI,
>>>
>>> I need a little info how replication of registrations is done. I just
>>> use function t_replicate() in script where I am doing registration stuff?
>>>
>>> What about security must there be any credential or how other proxy
>>> deals with this? Can any proxy post this repication data to my server?
>>>
>>> Thanks!
>>> Miha
>>>
>>>
>>>
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>>>
>
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Re: [OpenSIPS-Users] registration t_replicate()

2013-06-12 Thread Miha


Hi Bogdan,

where exactly t_replicate() should be used?

my script:

 if (is_method("REGISTER") && is_from_local())
{

# make pua_usrloc send PUBLISH for phones which do
# not support presence
# filter after User-Agent header
if(!search("^User-Agent: X-Lite")) pua_set_publish();

## --NAT-- ##
if (isflagset(5)) {
setbflag(6);
setbflag(7);
};
## --/NAT-- ##

# authenticate the REGISTER requests (uncomment to 
enable auth)

if (!www_authorize("", "subscriber"))
{
www_challenge("", "0");
exit;
}
*t_replicate("sip:xxx.xxx.xxx.xxx:5060");*
if (!db_check_to())
{
sl_send_reply("403","Forbidden auth ID");
exit;
}
if (!save("location"))
sl_reply_error();

exit;
}


I am getting this:

Jun 12 14:45:39 localhost /usr/sbin/opensips[12659]: 
ERROR:pua:send_publish_int: New 'presence' PUBLISH and no body found - 
invalid request
Jun 12 14:45:39 localhost /usr/sbin/opensips[12659]: 
ERROR:tm:_reply_light: failed to generate 200 reply when a final 401 was 
sent out
Jun 12 14:45:39 localhost /usr/sbin/opensips[12659]: 
ERROR:signaling:sig_send_reply_mod: failed to send reply with tm module
Jun 12 14:45:39 localhost /usr/sbin/opensips[12659]: 
ERROR:registrar:send_reply: failed to send 200 OK


thanks!

miha




Dne 6/12/2013 12:07 PM, pis(e Bogdan-Andrei Iancu:

Hello Miha,

Yes, t_replicate() is designed to be used after to finished the local
processing of the REGISTER (like doing the save(location)).

This mechanism is intended to be used between servers in the same
platform, so you can use IP-based authentication for the replicated
REGISTERs (no need for digest auth). And you should not accept
replicated registers from anywhere, but only from servers from your
platform.

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com


On 06/11/2013 02:49 PM, Miha wrote:

HI,

I need a little info how replication of registrations is done. I just
use function t_replicate() in script where I am doing registration stuff?

What about security must there be any credential or how other proxy
deals with this? Can any proxy post this repication data to my server?

Thanks!
Miha



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Re: [OpenSIPS-Users] CDRTool - Callcontrol - Prepaid

2013-06-12 Thread Laszlo

what happens if you manually set $avp(can_uri) to a fixed value? like to
, and define this destination in cdrtool?
2013.06.11. 18:08, "Leonardo Uzcudun"  ezt írta:

> Did it but still not working. I'm still receiving Maxsessiontime=unlimited
> and DetsId=none :(
>
> Is there any "default destination" (like a default gateway)? So no matched
> destination will use the "default destination" with an X price/rate/profile?
>
>   --
>  *Da:* Adrian Georgescu 
> *A:* OpenSIPS users mailling list 
> *Inviato:* Martedì 11 Giugno 2013 17:18
> *Oggetto:* Re: [OpenSIPS-Users] CDRTool - Callcontrol - Prepaid
>
> You need to add pricing info in the other billing tables too.
>
> Adrian
>
> On Jun 11, 2013, at 3:24 PM, Leonardo Uzcudun wrote:
>
> Thanks again Adrian.
> You mean to insert in destinations (CDRtool web portal -> Rating ->
> Destinations) the use...@example.com in the destination field?
>
>   --
>  *Da:* Adrian Georgescu 
> *A:* OpenSIPS users mailling list 
> *Inviato:* Martedì 11 Giugno 2013 14:32
> *Oggetto:* Re: [OpenSIPS-Users] CDRTool - Callcontrol - Prepaid
>
> Never tried it myself for prepaid but you can setup SIP URIs as valid
> destinations too and put a price on them.
>
> Adrian
>
> On Jun 11, 2013, at 12:29 PM, Leonardo Uzcudun wrote:
>
> I guess this confirm why i can't use prepaid for calls between sip
> users..., do you confirm it?
> Thanks.
>
>   --
>  *Da:* Adrian Georgescu 
> *A:* OpenSIPS users mailling list 
> *Inviato:* Martedì 11 Giugno 2013 11:57
> *Oggetto:* Re: [OpenSIPS-Users] CDRTool - Callcontrol - Prepaid
>
> The prepaid engine was designed for PSTN destinations using phone numbers
> only.
>
> Adrian
>
> On 11 Jun 2013, at 11:49, Leonardo Uzcudun  wrote:
>
> Hello Adrian:
>
> From the following site (
> http://callcontrol.ag-projects.com/issues/1769#change-8449) i've found
> that you say "All internals of CDRTool rely on the fact that there is a
> normalized URI in the form of 00 + Country Code + Number."
> So, my question or doubt is:
> If there are 2 sip users within the same domain like use...@example.com
> and use...@example.com, can i implement prepaid billing between them or
> the prepaid will only work if they call to 00 + Country Code + Number sip
> accounts?
> Thanks,
>
> Leo
> --
> *Da:* Adrian Georgescu 
> *A:* OpenSIPS users mailling list 
> *Inviato:* Lunedì 10 Giugno 2013 21:39
> *Oggetto:* Re: [OpenSIPS-Users] CDRTool - Callcontrol - Prepaid
>
> There is Rating.txt document that contain sample rates files and examples
> of prepaid queries in the console that shows you how it works.
>
> Adrian
>
>
> On Jun 10, 2013, at 9:30 PM, Leonardo Uzcudun wrote:
>
> Hello All:
>
> After doing more tests i guess the problem is the missing DestId
> cdrtool log:
> Jun  8 02:51:12 sip cdrtool[13359]: MaxSessionTime=unlimited Type=prepaid
> CallId=1226626922 BillingParty=user01@mydomain DestId=None
>
> and considering the rating.txt doc point 3. Determination of the costs ->
> No rate will be returned if no 'destination id' is found.
>
> So, now my question is: If the Destination (cdrtool web portal -> Rating
> -> Destinations) must be a number (like 31 for the Netherlands), how will i
> match destination to mydomain?
>
> Thanks,
>
> Leo.
> --
> *Da:* leo 
> *A:* users@lists.opensips.org
> *Inviato:* Venerdì 7 Giugno 2013 9:38
> *Oggetto:* [OpenSIPS-Users] CDRTool - Callcontrol - Prepaid
>
> Hello:
>
> I need some help to understand the CDRTool - Callcontrol - Prepaid, all the
> efforts i could do reading, searching and testing to try to understand how
> it works were unsuccessful.
>
> My set-up would be very simple, I would like to give credit (in seconds) to
> the users and after the call is ended the call duration time should be
> subtracted to its balance.
>
> My first concern is how do i map a user in the opensips.subscriber
> (DB.table) to the prepaid billing?
> The second one would be, how would i configure the rating/quota/prepaid to
> have my set-up?
>
> Thanks for your help.
>
> Leo.
>
>
>
> --
> View this message in context:
> http://opensips-open-sip-server.1449251.n2.nabble.com/CDRTool-Callcontrol-Prepaid-tp7586750.html
> Sent from the OpenSIPS - Users mailing list archive at 
> Nabble.com
> .
>
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>
>
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>
>
>
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>
>
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Re: [OpenSIPS-Users] CDRTool - Callcontrol - Prepaid

2013-06-12 Thread Leonardo Uzcudun

Thanks Laszlo, good idea! but it still not working.
 
Now i can see in the cdrtool web portal that the canonical-uri is  (check 
the attached file, there is a comparison before and after changing the can-uri 
to ) but when the sip user samsung@sip.mydomain places the call i still 
seeing in the syslog:
 
Jun 12 11:56:50 sip cdrtool[1304]: MaxSessionTime Duration=7200 
CallId=1517185829 From=sip:samsung@sip.mydomain Gateway=82.186.X.X 
To=sip:ipad@sip.mydomain
Jun 12 11:56:50 sip cdrtool[1304]: MaxSessionTime=unlimited Type=prepaid 
CallId=1517185829 BillingParty=samsung@sip.mydomain DestId=None
Jun 12 11:56:50 sip call-control[1985]: Call id 1517185829 of 
samsung@sip.mydomain to sip:ipad@sip.mydomainsip is postpaid not limited

 
 
 
 
 
 
 
 
 
 
 
One more thing, with change (can-uri ) when the cdrtool does the 
normalization i see in the syslog:
 
 
 
 
Jun 12 11:59:27 sip cdrtool[7012]: Error: cannot find destinations for 
subscriber='samsung@sip.mydomain', domain ='', gateway='', reseller='0'
Jun 12 11:59:27 sip cdrtool[7012]: Error: cannot find destination id for  
of customer = '', total destinations = 0
 
 
 
Do i configuring something wrong in the cdrtool web portal -> Rating -> 
Destinations?
 
Is is normal to have domain=''? shouldn't be sip.mydomain?
 
Thanks a lot for your help,
 
Leo.


Da: Laszlo 
A: Leonardo Uzcudun ; OpenSIPS users mailling list 
 
Inviato: Martedì 11 Giugno 2013 18:26
Oggetto: Re: [OpenSIPS-Users] CDRTool - Callcontrol - Prepaid



what happens if you manually set $avp(can_uri) to a fixed value? like to , 
and define this destination in cdrtool?
2013.06.11. 18:08, "Leonardo Uzcudun"  ezt írta:

Did it but still not working. I'm still receiving Maxsessiontime=unlimited and 
DetsId=none :(
>
>
>Is there any "default destination" (like a default gateway)? So no matched 
>destination will use the "default destination" with an X price/rate/profile?
>
>
>
>Da: Adrian Georgescu 
>A: OpenSIPS users mailling list  
>Inviato: Martedì 11 Giugno 2013 17:18
>Oggetto: Re: [OpenSIPS-Users] CDRTool - Callcontrol - Prepaid
>
>
>
>You need to add pricing info in the other billing tables too.
>
>
>Adrian
>
>On Jun 11, 2013, at 3:24 PM, Leonardo Uzcudun wrote:
>
>Thanks again Adrian.
>>You mean to insert in destinations (CDRtool web portal -> Rating -> 
>>Destinations) the use...@example.com in the destination field?
>>
>>
>>
>>Da: Adrian Georgescu 
>>A: OpenSIPS users mailling list  
>>Inviato: Martedì 11 Giugno 2013 14:32
>>Oggetto: Re: [OpenSIPS-Users] CDRTool - Callcontrol - Prepaid
>>
>>
>>
>>Never tried it myself for prepaid but you can setup SIP URIs as valid 
>>destinations too and put a price on them.  
>>
>>
>>Adrian
>>
>>
>>
>>On Jun 11, 2013, at 12:29 PM, Leonardo Uzcudun wrote:
>>
>>I guess this confirm why i can't use prepaid for calls between sip users..., 
>>do you confirm it?
>>>Thanks. 
>>>
>>>
>>>Da: Adrian Georgescu 
>>>A: OpenSIPS users mailling list  
>>>Inviato: Martedì 11 Giugno 2013 11:57
>>>Oggetto: Re: [OpenSIPS-Users] CDRTool - Callcontrol - Prepaid
>>>
>>>
>>>
>>>The prepaid engine was designed for PSTN destinations using phone numbers 
>>>only.  
>>>
>>>
>>>Adrian
>>>
>>>
>>>On 11 Jun 2013, at 11:49, Leonardo Uzcudun  wrote:
>>>
>>>Hello Adrian:

From the following site 
(http://callcontrol.ag-projects.com/issues/1769#change-8449) i've found 
that you say "All internals of CDRTool rely on the fact that there is a 
normalized URI in the form of 00 + Country Code + Number."
So, my question or doubt is:
If there are 2 sip users within the same domain like use...@example.com and 
use...@example.com, can i implement prepaid billing between them or the 
prepaid will only work if they call to 00 + Country Code + Number sip 
accounts?
Thanks,

Leo 
Da: Adrian Georgescu 
A: OpenSIPS users mailling list  
Inviato: Lunedì 10 Giugno 2013 21:39
Oggetto: Re: [OpenSIPS-Users] CDRTool - Callcontrol - Prepaid



There is Rating.txt document that contain sample rates files and examples 
of prepaid queries in the console that shows you how it works.  


Adrian




On Jun 10, 2013, at 9:30 PM, Leonardo Uzcudun wrote:

Hello All:
>
>
>
>After doing more tests i guess the problem is the missing DestId
>cdrtool log:Jun  8 02:51:12 sip cdrtool[13359]: MaxSessionTime=unlimited 
>Type=prepaid CallId=1226626922 BillingParty=user01@mydomain DestId=None
>
>and considering the rating.txt doc point 3. Determination of the costs -> 
>No rate will be returned if no 'destination id' is found. 
>
>
>So, now my question is: If the Destination (cdrtool web portal -> Rating 
>-> Destinations) must be a number (like 31 for the Netherlands), how will 
>i match destination to mydomain?
>
>
>Thanks,
>
>Leo. 
>Da: leo 
>A: users@lists.opensips.org 
>Inviato: Venerdì 7 Giu

Re: [OpenSIPS-Users] AVP on local route

2013-06-12 Thread Bogdan-Andrei Iancu

Hello Khaled,

What kind of accounting are you using in script ? Based on CDR flag ?
Based on ACC flag (start/stop) ? or manual accounting (with functions) ?

Maybe the record you are trying to update in local route does not exist
in ACC table yet (depending on the accounting method you are using).

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com


On 06/03/2013 02:27 PM, M.Khaled W Chehab wrote:
>
> Hi,
>
>  
>
> The   execution for the avp_db_query is not working fine  in local
> route after the  acc_db_request  ,
>
>  
>
> As I print the query using xlog(" and   insert t it to myql ,the query
> executed normally which means I have no problem in the mysql query  .
>
>  
>
> My question: how can update the record in the acc table  after
> acc_db_request("200 Dialog Timeout", "
>
>  
>
>  
>
> local_route {
>
> if (is_method("BYE")  && $DLG_dir=="UPSTREAM") {
>
>  xlog("L_INFO", "--Debug Customer
> ID:$avp(Cusid)/IP:$si-Reply:
> $T_reply_code--Source:$rd\nDialed:$fU\n:");
>
>  
>
> acc_db_request("200 Dialog Timeout", "acc");
>
>  
>
> avp_db_query("UPDATE `opensips`.`acc` SET `pdd`='$var(pdd)' WHERE
> `CALLID` ='$ci' and `method`='INVITE'" );
>
>  
>
>  
>
> BTW: I cant find the acc record for the call unless  the call  ended
> ,and in my case I set the  acc_db_request("200 Dialog Timeout  then
> call avp_db_query(   is  that should insert the call in the acc to
> update it later
>
>  
>
> Regards
>
> ,.
>
>  
>
>
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Re: [OpenSIPS-Users] Redis support on Wheezy

2013-06-12 Thread Bogdan-Andrei Iancu

Hi Pete,

Thanks for pointing this out, we will update the docs.

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com


On 05/31/2013 12:45 PM, Pete Kelly wrote:
> Hi
>
> The docs say that a manual compile of hiredis is required for this,
> however on Debian Wheezy the package libhiredis-dev is now available,
> and opensips with redis support compiles successfully once this is
> installed.
>
> Pete
>
>
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Re: [OpenSIPS-Users] SIP caller is not notified when the INVITE has Request Timeout(408) from opensips

2013-06-12 Thread Bogdan-Andrei Iancu

Hello Miguel,

In the bogus case, the problem is that the replies are received long
after the transaction was deleted by opensips (as it was completed via
the local timeout).

If the transaction would still be in memory when the late 180 is
received, the CANCEL will be fired (like in the working case).

As configured right now, the UAS, as not receiving an ACK for the 200 OK
, must drop the call - this is what the RFC3261 says.

Another option for you would be to prelong how long OpenSIPS keeps
transactions in mem after completion (see wait timer - wt_timer in TM
module -
http://www.opensips.org/html/docs/modules/1.9.x/tm.html#id250377) . Be
careful as using higher values here may result in more sh mem used (as
transaction will accumulate in memory).

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com


On 06/10/2013 05:20 PM, Miguel París Díaz wrote:
> Hi, I am using opensips-1.9.1-tls and some times I have a problem when there 
> is a Request Timeout (408) for the INVITE request in the proxy side.
>
> The problem is caused when the 180 Ringing provisional response arrives after 
> some seconds from the SIP caller.
>
> If the Ringing arrives early we have a *successful case*:
>
>
> UAC(caller) PROXY(opensips) UAS(callee)
> INVITE -->
> < 100 Giving a try
>  INVITE--->
> < 408 request timeout
> ACK ->
> <- 180 Ringing
>  CANCEL -->
> <- 200 OK (CANCEL)
> <- 487 Request Terminated
>
>
> But if the Ringing is late, we have a *error case*, and the SIP caller is not 
> notified, so it retry the 200 OK responses of the invite indefinitely.
>
>
> UAC(caller) PROXY(opensips) UAS(callee)
> INVITE -->
> < 100 Giving a try
>  INVITE--->
> < 408 request timeout
> ACK ->
>    some secons 
> <- 180 Ringing
> <--- 200 OK (invite)
> <--- 200 OK (invite)
> <--- 200 OK (invite)
> <--- 200 OK (invite)
>
> The problem in this case is that cannot found a matching transaction,
> as we can see in the next log:
>
> Jun  7 11:14:10 DBG:core:tcp_read_req: content-length= 0
> Jun  7 11:14:10 DBG:core:parse_msg: SIP Reply  (status):
> Jun  7 11:14:10 DBG:core:parse_msg:  version: 
> Jun  7 11:14:10 DBG:core:parse_msg:  status:  <180>
> Jun  7 11:14:10 DBG:core:parse_msg:  reason:  
> Jun  7 11:14:10 DBG:core:parse_headers: flags=2
> Jun  7 11:14:10 DBG:core:get_hdr_field: cseq : <2> 
> Jun  7 11:14:10 DBG:core:parse_to_param: tag=spekujkorj
> Jun  7 11:14:10 DBG:core:parse_to: end of header reached, state=29
> Jun  7 11:14:10 DBG:core:parse_to: display={},
> ruri={sip:no...@hnj.kurento.com }
> Jun  7 11:14:10 DBG:core:get_hdr_field:  [44];
> uri=[sip:no...@hnj.kurento.com ]
> Jun  7 11:14:10 DBG:core:get_hdr_field: to body
> [mailto:sip%3ano...@hnj.kurento.com>>]
> Jun  7 11:14:10 DBG:core:parse_via_param: found param type 232,
>  = ; state=6
> Jun  7 11:14:10 DBG:core:parse_via_param: found param type 236,  =
> <03>; state=6
> Jun  7 11:14:10 DBG:core:parse_via_param: found param type 235,
>  = <5080>; state=9
> Jun  7 11:14:10 DBG:core:parse_via: next_via
> Jun  7 11:14:10 DBG:core:parse_via_param: found param type 234,
>  = <176.83.57.96>; state=6
> Jun  7 11:14:10 DBG:core:parse_via_param: found param type 232,
>  = ; state=6
> Jun  7 11:14:10 DBG:core:parse_via_param: found param type 235,
>  = <30531>; state=16
> Jun  7 11:14:10 DBG:core:parse_via: end of header reached, state=5
> Jun  7 11:14:10 DBG:core:parse_headers: via found, flags=2
> Jun  7 11:14:10 DBG:core:parse_headers: this is the first via
> Jun  7 11:14:10 DBG:core:receive_msg: After parse_msg...
> Jun  7 11:14:10 DBG:core:forward_reply: found module tm, passing reply
> to it
> Jun  7 11:14:10 DBG:tm:t_check: start=0x
> Jun  7 11:14:10 DBG:core:parse_headers: flags=22
> Jun  7 11:14:10 DBG:core:parse_headers: flags=8
> Jun  7 11:14:10 DBG:tm:t_reply_matching: hash 50704 label 1366865684
> branch 0
> Jun  7 11:14:10 DBG:tm:t_reply_matching: no matching transaction exists
> Jun  7 11:14:10 DBG:tm:t_reply_matching: failure to match a transaction
> Jun  7 11:14:10 DBG:tm:t_check: end=(nil)
> Jun  7 11:14:10 DBG:core:destroy_avp_list: destroying list (nil)
> Jun  7 11:14:10 DBG:core:receive_msg: cleaning up
>
>
> To force this behavior I have set the next values in opensips.cfg
> modparam("tm", "fr_timer", 2) # Timeout if a provisonal response for
> INVITE does not arrive before 2 seconds
> modparam("tm", "fr_inv_timer", 60)
>
> -- 
> http://twitter.com/mparisdiaz
>
>
> ___
> Users mailin

Re: [OpenSIPS-Users] opensips control panel

2013-06-12 Thread Bogdan-Andrei Iancu

Hello Nandini,

The mi_xmlrpc module is not be default compiled - use "make menuconfig"
(configure&compile options ->configure exclude modules) to enable the
compiling of that module (and recompile and re-install opensips of course)

Regards,

Bogdan-Andrei Iancu
OpenSIPS Founder and Developer
http://www.opensips-solutions.com


On 06/12/2013 08:28 AM, Nandini madhu wrote:
> Thanks for your reply.
>
>
> iam using the latest version of opensips i.e 1.9.1-tls.i could not
> find the mi_xmlrpc.so file.
> in usr/local/lib/opensips/modules.
>
> acc.so cachedb_local.so  dialog.so 
> event_route.somediaproxy.so peering.so   
> rtpproxy.sosst.so uri.so
> alias_db.so   cachedb_sql.sodispatcher.so 
> event_xmlrpc.so   mi_datagram.sopermissions.so   
> seas.sostatistics.so  userblacklist.so
> auth_aaa.so   call_control.so   diversion.so  
> exec.so   mi_fifo.sopike.so  
> signaling.so   stun.sousrloc.so
> auth_db.socfgutils.so   dns_cache.so  
> gflags.so msilo.so  presence_callinfo.so 
> sipcapture.so  textops.so
> auth_diameter.so  closeddial.so domainpolicy.so   
> group.so  nathelper.so  presence_xcapdiff.so 
> sipmsgops.so   tm.so
> auth.so   db_flatstore.so   domain.so 
> imc.sonat_traversal.so  qos.so   
> siptrace.souac_auth.so
> avpops.so db_mysql.so   drouting.so   
> load_balancer.so  options.soratelimit.so 
> sl.so  uac_redirect.so
> b2b_entities.so   db_text.soenum.so   
> mangler.sopath.so   registrar.so 
> sms.so uac_registrant.so
> benchmark.so  db_virtual.so event_datagram.so 
> maxfwd.so pdt.sorr.so
> speeddial.so   uac.so
>
>
> by giving netstat -tlnp | grep opensips
> i could not find any output.
>
> please help me
>
>
>
>
>
>
>
>
> On Tue, Jun 11, 2013 at 8:57 PM, Bogdan-Andrei Iancu
> mailto:bog...@opensips.org>> wrote:
>
> Hi,
>
> Are you using the mi_xmlrpc module in your opensips cfg ? That's
> the module responsible for listening on 8000 port - btw check if
> the port in opensips  (mi_xmlrpc module) matches the one in CP. Do:
> netstat -tlnp | grep opensips
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developer
> http://www.opensips-solutions.com
>
>
> On 06/11/2013 10:53 AM, Nandini madhu wrote:
>> Dear All,
>>
>> In the control panel I get these error messages:Cannot connect to
>> OpenSIPS Server via Management Interface (192.168.2.40/8000
>> )
>>
>> my opensips server is running ,
>> i installed opensips server and opensips control panel on same pc.
>>
>> please help me...
>>
>>
>> On Mon, Jun 10, 2013 at 5:33 PM, Bogdan-Andrei Iancu
>> mailto:bog...@opensips.org>> wrote:
>>
>> That line has to be in config/boxes.global.inc.php
>>
>> Regards,
>>
>> Bogdan-Andrei Iancu
>> OpenSIPS Founder and Developer
>> http://www.opensips-solutions.com
>>
>>
>> On 06/10/2013 02:37 PM, Nandini madhu wrote:
>>> Thanks for your response...
>>>
>>> in load balancer in local.inc.php
>>>
>>> i could not find this line..
>>>
>>> hange $box[1]['mi']['conn']="/tmp/opensips_fifo"
>>>
>>> but i added the line in local.inc.php
>>>
>>> and i restarted the server,but the error is same.
>>>
>>> please help me.
>>>
>>> thank you
>>>
>>>
>>>
>>>
>>>
>>> On Mon, Jun 10, 2013 at 3:23 PM, Miha >> > wrote:
>>>
>>> hi,
>>>
>>> in: opensips-cp/config/tools/system/dispatcher/local.inc.php
>>>
>>> change $box[1]['mi']['conn']="/tmp/opensips_proxy_fifo";
>>> to $box[1]['mi']['conn']="/tmp/opensips_fifo";
>>>
>>>
>>> br,
>>> miha
>>>
>>> Dne 6/10/2013 11:46 AM, piše Nandini madhu:
 Dear all.
 Greetings.

 I have resolved the issue.i can log in to web page.
 In *System / Dispatcher / Read-Write

 *iam getting error ;-
 *Array ( [0] => Cannot connect to OpenSIPS Server via
 Management Interface (/tmp/opensips_proxy_fifo) )

 *
 how to solve this problem.


 thank you*
 *


 On Mon, Jun 10, 2013 at 2:56 PM, Miha >>> > wrote:

 Dne 6/8/2013 1:26 PM, piše Nandini madhu:
> Dear All,
>
> iam installed opensips server.
> As the steps given in installation guide.
>  
>

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