Re: [OpenSIPS-Users] Opensips-cp 8.3.0 / Call to undefined function json_encode()
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 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: Creating default object from
> empty value in
> /var/www/html/opensips-cp/config/tools/system/drouting/local.inc.php on
> line 24
>
> [30-Nov-2020 14:13:54 UTC] PHP Stack trace:
>
> [30-Nov-2020 14:13:54 UTC] PHP 1. {main}()
> /var/www/html/opensips-cp/web/tools/system/drouting/apply_changes.php:0
>
> [30-Nov-2020 14:13:54 UTC] PHP 2. require()
> /var/www/html/opensips-cp/web/tools/system/drouting/apply_changes.php:25
>
> [30-Nov-2020 14:13:54 UTC] PHP Error: Call to undefined function
> json_encode() in /var/www/html/opensips-cp/web/common/mi_comm.php on line 31
>
> [30-Nov-2020 14:13:54 UTC] PHP Stack trace:
>
> [30-Nov-2020 14:13:54 UTC] PHP 1. {main}()
> /var/www/html/opensips-cp/web/tools/system/drouting/apply_changes.php:0
>
> [30-Nov-2020 14:13:54 UTC] PHP 2. mi_command($command = 'dr_reload',
> $params_array = NULL, $mi_url = 'json:127.0.0.1:8000/JSON', $errors =
> NULL)
> /var/www/html/opensips-cp/web/tools/system/drouting/apply_changes.php:43
>
> [30-Nov-2020 14:13:54 UTC] PHP 3. write2json($command = 'dr_reload',
> $params_array = NULL, $json_url = '127.0.0.1:8000/JSON', $errors = NULL)
> /var/www/html/opensips-cp/web/common/mi_comm.php:87
>
> [30-Nov-2020 14:13:54 UTC] PHP Fatal error: Uncaught Error: Call to
> undefined function json_encode() in
> /var/www/html/opensips-cp/web/common/mi_comm.php:31
>
> Stack trace:
>
> #0 /var/www/html/opensips-cp/web/common/mi_comm.php(87):
> write2json('dr_reload', NULL, '127.0.0.1:8000/...', NULL)
>
> #1
> /var/www/html/opensips-cp/web/tools/system/drouting/apply_changes.php(43):
> mi_command('dr_reload', NULL, 'json:127.0.0.1:...', NULL)
>
> #2 {main}
>
> thrown in /var/www/html/opensips-cp/web/common/mi_comm.php on line 31
>
>
> I followed installation document located at
> http://controlpanel.opensips.org/documentation.php.
>
>
> Here is my boxes.global.inc.php config:
>
> $boxes[$box_id]['mi']['conn']="json:127.0.0.1:8000/JSON";
>
> but i also tried with
>
> $boxes[$box_id]['mi']['conn']="json:127.0.0.1:8000/mi";
>
> with the same effect.
>
>
> Opensips (compiled from the sources) has got following modules up and
> running:
>
>
> HTTP
>
> loadmodule "httpd.so"
>
> modparam("httpd", "port", 8000)
>
>
> ###JSON
>
> loadmodule "json.so"
>
>
> ###MI_HTTP
>
> loadmodule "mi_http.so"
>
>
> Could you please point me where the problem might be located ?
>
>
> Thanks
>
> Maciej
>
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[OpenSIPS-Users] Opensips-cp 8.3.0 / Call to undefined function json_encode()
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: Creating default object from empty
value in
/var/www/html/opensips-cp/config/tools/system/drouting/local.inc.php on
line 24
[30-Nov-2020 14:13:54 UTC] PHP Stack trace:
[30-Nov-2020 14:13:54 UTC] PHP 1. {main}()
/var/www/html/opensips-cp/web/tools/system/drouting/apply_changes.php:0
[30-Nov-2020 14:13:54 UTC] PHP 2. require()
/var/www/html/opensips-cp/web/tools/system/drouting/apply_changes.php:25
[30-Nov-2020 14:13:54 UTC] PHP Error: Call to undefined function
json_encode() in /var/www/html/opensips-cp/web/common/mi_comm.php on line 31
[30-Nov-2020 14:13:54 UTC] PHP Stack trace:
[30-Nov-2020 14:13:54 UTC] PHP 1. {main}()
/var/www/html/opensips-cp/web/tools/system/drouting/apply_changes.php:0
[30-Nov-2020 14:13:54 UTC] PHP 2. mi_command($command = 'dr_reload',
$params_array = NULL, $mi_url = 'json:127.0.0.1:8000/JSON', $errors = NULL)
/var/www/html/opensips-cp/web/tools/system/drouting/apply_changes.php:43
[30-Nov-2020 14:13:54 UTC] PHP 3. write2json($command = 'dr_reload',
$params_array = NULL, $json_url = '127.0.0.1:8000/JSON', $errors = NULL)
/var/www/html/opensips-cp/web/common/mi_comm.php:87
[30-Nov-2020 14:13:54 UTC] PHP Fatal error: Uncaught Error: Call to
undefined function json_encode() in
/var/www/html/opensips-cp/web/common/mi_comm.php:31
Stack trace:
#0 /var/www/html/opensips-cp/web/common/mi_comm.php(87):
write2json('dr_reload', NULL, '127.0.0.1:8000/...', NULL)
#1
/var/www/html/opensips-cp/web/tools/system/drouting/apply_changes.php(43):
mi_command('dr_reload', NULL, 'json:127.0.0.1:...', NULL)
#2 {main}
thrown in /var/www/html/opensips-cp/web/common/mi_comm.php on line 31
I followed installation document located at
http://controlpanel.opensips.org/documentation.php.
Here is my boxes.global.inc.php config:
$boxes[$box_id]['mi']['conn']="json:127.0.0.1:8000/JSON";
but i also tried with
$boxes[$box_id]['mi']['conn']="json:127.0.0.1:8000/mi";
with the same effect.
Opensips (compiled from the sources) has got following modules up and
running:
HTTP
loadmodule "httpd.so"
modparam("httpd", "port", 8000)
###JSON
loadmodule "json.so"
###MI_HTTP
loadmodule "mi_http.so"
Could you please point me where the problem might be located ?
Thanks
Maciej
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Re: [OpenSIPS-Users] CACHEDB_MEMCACHED Module - libmemcached undefined symbol issue
Hi,
Don't know there the problem was located. Port was only utilized by
memcached process.
Anyway i hope there are no issues within the memcached module code.
Thanks
Maciej.
2016-11-22 10:30 GMT+01:00 Bogdan-Andrei Iancu <bog...@opensips.org>:
> Hi Maciej,
>
> That is weired, but I'm glad 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, 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 <mbgathe...@gmail.com>:
>
>> Hi Bogdan,
>>
>> Thanks for the reply.
>>
>> It seems it is related to the key, it doesn't matter which query is it.
>> First query on the second key does not change anything.
>> I've just added additional key 49101112233 and it works (query was
>> fired), but 49331112233 does not.
>>
>> Thanks
>> Maciej.
>>
>>
>> 2016-11-21 12:59 GMT+01:00 Bogdan-Andrei Iancu <bog...@opensips.org>:
>>
>>> Hi Maciej,
>>>
>>> Thanks for the detailed report.
>>>
>>> Do you think the error is related to the key you are trying to fetch or
>>> is it related to the simply being the second query you perform ? What if
>>> you perform from the very beginning 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
>>> innodb_memcache.containers to implement memcached as a plugin.
>>>
>>> netstat -plnt | grep
>>>
>>> tcp0 0 127.0.0.1: 0.0.0.0:*
>>> LISTEN 18421/mysqld
>>>
>>> Everything looks fine i have full transparency, data provided by
>>> memcached CLI (telnet) are seen inside innodb table and vise versa.
>>> I am using the latest 2.2.2 git opensips rel. and memcached module
>>> loaded:
>>>
>>> loadmodule "cachedb_memcached.so"
>>>
>>> modparam("cachedb_memcached", "cachedb_url","memcached:default:
>>> //localhost:,127.0.0.1/")
>>> The script i am using is just the basic one, without any additional
>>> configuration.
>>> Inside the script there is following operation provided:
>>>
>>> cache_fetch("memcached:default","$tU",$avp(i:601));
>>> Innodb table contains following data:
>>>
>>> +-+-+--+--+--+
>>>
>>> | id | num | c3 | c4 | c5 |
>>>
>>> +-+-+--+--+--+
>>>
>>> | 49121112233 | 49121112233 |0 |3 |0 |
>>>
>>> | 49221112233 | 49221112233 |0 |1 |0 |
>>>
>>> | 49221112234 | 49221112234 |0 |2 |0 |
>>>
>>> +-+-+--+--+--+
>>> Now, i am sending INVITE with tU = 49121112233 and getting proper
>>> behavior which means:
>>> - no error inside the opensips.log, xlog following cache_fetch returns
>>> correct $avp(i:601) - mysqld.log shows
>>>
>>> <95 get 49121112233
>>>
>>> >95 sending key 49121112233
>>>
>>> >95 END
>>> but really strange is that calling tU = 49221112233 is causing quite
>>> opposite results:
>>> - following error is shown
>>>
>>> DBG:core:cachedb_fetch: from script [memcached] - with grp [default]
>>>
>>> ERROR:cachedb_memcached:wrap_memcached_get: Failed to get: SYSTEM ERROR
>>>
>>> - no mysqld debug is produced
>>>
>>> The last one example(tU = 49221112234)is failing with the same error.
>>>
>>> Memcached is loaded with all those data
>>>
>>> Connected to localhost.
>>>
>>> Escape character is '^]'.
>>>
>>> get 49221112233
>>>
>>> VALUE 49221112233 0 11
>>>
>>> 49221112233
>>>
>>> END
>>>
>>> get 49221112234
>>>
>>> VALUE 49221112234 0 11
>>>
>>> 49221112234
>>>
&
Re: [OpenSIPS-Users] CACHEDB_MEMCACHED Module - libmemcached undefined symbol issue
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 <mbgathe...@gmail.com>:
> Hi Bogdan,
>
> Thanks for the reply.
>
> It seems it is related to the key, it doesn't matter which query is it.
> First query on the second key does not change anything.
> I've just added additional key 49101112233 and it works (query was fired),
> but 49331112233 does not.
>
> Thanks
> Maciej.
>
>
> 2016-11-21 12:59 GMT+01:00 Bogdan-Andrei Iancu <bog...@opensips.org>:
>
>> Hi Maciej,
>>
>> Thanks for the detailed report.
>>
>> Do you think the error is related to the key you are trying to fetch or
>> is it related to the simply being the second query you perform ? What if
>> you perform from the very beginning 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
>> innodb_memcache.containers to implement memcached as a plugin.
>>
>> netstat -plnt | grep
>>
>> tcp0 0 127.0.0.1: 0.0.0.0:*
>> LISTEN 18421/mysqld
>>
>>
>> Everything looks fine i have full transparency, data provided by
>> memcached CLI (telnet) are seen inside innodb table and vise versa.
>>
>> I am using the latest 2.2.2 git opensips rel. and memcached module loaded:
>>
>> loadmodule "cachedb_memcached.so"
>>
>> modparam("cachedb_memcached", "cachedb_url","memcached:default:
>> //localhost:,127.0.0.1/")
>>
>> The script i am using is just the basic one, without any additional
>> configuration.
>> Inside the script there is following operation provided:
>>
>> cache_fetch("memcached:default","$tU",$avp(i:601));
>>
>> Innodb table contains following data:
>>
>> +-+-+--+--+--+
>>
>> | id | num | c3 | c4 | c5 |
>>
>> +-+-+--+--+--+
>>
>> | 49121112233 | 49121112233 |0 |3 |0 |
>>
>> | 49221112233 | 49221112233 |0 |1 |0 |
>>
>> | 49221112234 | 49221112234 |0 |2 |0 |
>>
>> +-+-+--+--+--+
>>
>> Now, i am sending INVITE with tU = 49121112233 and getting proper
>> behavior which means:
>> - no error inside the opensips.log, xlog following cache_fetch returns
>> correct $avp(i:601)
>> - mysqld.log shows
>>
>> <95 get 49121112233
>>
>> >95 sending key 49121112233
>>
>> >95 END
>>
>> but really strange is that calling tU = 49221112233 is causing quite
>> opposite results:
>> - following error is shown
>>
>> DBG:core:cachedb_fetch: from script [memcached] - with grp [default]
>>
>> ERROR:cachedb_memcached:wrap_memcached_get: Failed to get: SYSTEM ERROR
>>
>> - no mysqld debug is produced
>>
>>
>> The last one example(tU = 49221112234)is failing with the same error.
>>
>>
>> Memcached is loaded with all those data
>>
>> Connected to localhost.
>>
>> Escape character is '^]'.
>>
>> get 49221112233
>>
>> VALUE 49221112233 0 11
>>
>> 49221112233
>>
>> END
>>
>> get 49221112234
>>
>> VALUE 49221112234 0 11
>>
>> 49221112234
>>
>> END
>>
>>
>> but because of some reasons memcached module is not utilized.
>> As aforementioned, opensips script does not have any $rU filtering setup,
>> so should query for any data it is asked for.
>> Maybe i am wrong with some of my assumptions or the way memcached is
>> configured, so kindly help me to understand where the problem is located.
>>
>> Thanks
>> Maciej.
>>
>>
>>
>>
>>
>>
>>
>> 2016-11-15 18:09 GMT+01:00 Bogdan-Andrei Iancu <bog...@opensips.org>:
>>
>>> 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, Maciej Bylica wrote:
>>>
>>> Hi Bogdan,
>>> Thanks for reply.
>>> Right, Opensips module was not the source
Re: [OpenSIPS-Users] CACHEDB_MEMCACHED Module - libmemcached undefined symbol issue
Hi Bogdan,
Thanks for the reply.
It seems it is related to the key, it doesn't matter which query is it.
First query on the second key does not change anything.
I've just added additional key 49101112233 and it works (query was fired),
but 49331112233 does not.
Thanks
Maciej.
2016-11-21 12:59 GMT+01:00 Bogdan-Andrei Iancu <bog...@opensips.org>:
> Hi Maciej,
>
> Thanks for the detailed report.
>
> Do you think the error is related to the key you are trying to fetch or is
> it related to the simply being the second query you perform ? What if you
> perform from the very beginning 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
> innodb_memcache.containers to implement memcached as a plugin.
>
> netstat -plnt | grep
>
> tcp0 0 127.0.0.1: 0.0.0.0:*
> LISTEN 18421/mysqld
>
>
> Everything looks fine i have full transparency, data provided by memcached
> CLI (telnet) are seen inside innodb table and vise versa.
>
> I am using the latest 2.2.2 git opensips rel. and memcached module loaded:
>
> loadmodule "cachedb_memcached.so"
>
> modparam("cachedb_memcached", "cachedb_url","memcached:default:
> //localhost:,127.0.0.1/")
>
> The script i am using is just the basic one, without any additional
> configuration.
> Inside the script there is following operation provided:
>
> cache_fetch("memcached:default","$tU",$avp(i:601));
>
> Innodb table contains following data:
>
> +-+-+--+--+--+
>
> | id | num | c3 | c4 | c5 |
>
> +-+-+--+--+--+
>
> | 49121112233 | 49121112233 |0 |3 |0 |
>
> | 49221112233 | 49221112233 |0 |1 |0 |
>
> | 49221112234 | 49221112234 |0 |2 |0 |
>
> +-+-+--+--+--+
>
> Now, i am sending INVITE with tU = 49121112233 and getting proper
> behavior which means:
> - no error inside the opensips.log, xlog following cache_fetch returns
> correct $avp(i:601)
> - mysqld.log shows
>
> <95 get 49121112233
>
> >95 sending key 49121112233
>
> >95 END
>
> but really strange is that calling tU = 49221112233 is causing quite
> opposite results:
> - following error is shown
>
> DBG:core:cachedb_fetch: from script [memcached] - with grp [default]
>
> ERROR:cachedb_memcached:wrap_memcached_get: Failed to get: SYSTEM ERROR
>
> - no mysqld debug is produced
>
>
> The last one example(tU = 49221112234)is failing with the same error.
>
>
> Memcached is loaded with all those data
>
> Connected to localhost.
>
> Escape character is '^]'.
>
> get 49221112233
>
> VALUE 49221112233 0 11
>
> 49221112233
>
> END
>
> get 49221112234
>
> VALUE 49221112234 0 11
>
> 49221112234
>
> END
>
>
> but because of some reasons memcached module is not utilized.
> As aforementioned, opensips script does not have any $rU filtering setup,
> so should query for any data it is asked for.
> Maybe i am wrong with some of my assumptions or the way memcached is
> configured, so kindly help me to understand where the problem is located.
>
> Thanks
> Maciej.
>
>
>
>
>
>
>
> 2016-11-15 18:09 GMT+01:00 Bogdan-Andrei Iancu <bog...@opensips.org>:
>
>> 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, Maciej Bylica wrote:
>>
>> Hi Bogdan,
>> Thanks for reply.
>> Right, Opensips module was not the source of the problem.
>> I've managed to solve the issue, memcache is working fine.
>> Thanks
>> Maciej.
>> 2016-11-10 12:56 GMT+01:00 Bogdan-Andrei Iancu <bog...@opensips.org>:
>>>
>>> Hi Maciej, As I see, you are manually compiling and installing 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 09.11.2016 18:41, Maciej Bylica wrote:
>>>
>>> Hello I am struggling with memcached installation with the latest git
>>> opensips 2.2.2 an
Re: [OpenSIPS-Users] CACHEDB_MEMCACHED Module - libmemcached undefined symbol issue
Hello
As i mentioned before memcached is already installed. I am using
innodb_memcache.containers to implement memcached as a plugin.
netstat -plnt | grep
tcp0 0 127.0.0.1: 0.0.0.0:*
LISTEN 18421/mysqld
Everything looks fine i have full transparency, data provided by memcached
CLI (telnet) are seen inside innodb table and vise versa.
I am using the latest 2.2.2 git opensips rel. and memcached module loaded:
loadmodule "cachedb_memcached.so"
modparam("cachedb_memcached", "cachedb_url","memcached:default:
//localhost:,127.0.0.1/")
The script i am using is just the basic one, without any additional
configuration.
Inside the script there is following operation provided:
cache_fetch("memcached:default","$tU",$avp(i:601));
Innodb table contains following data:
+-+-+--+--+--+
| id | num | c3 | c4 | c5 |
+-+-+--+--+--+
| 49121112233 | 49121112233 |0 |3 |0 |
| 49221112233 | 49221112233 |0 |1 |0 |
| 49221112234 | 49221112234 |0 |2 |0 |
+-+-+--+--+--+
Now, i am sending INVITE with tU = 49121112233 and getting proper behavior
which means:
- no error inside the opensips.log, xlog following cache_fetch returns
correct $avp(i:601)
- mysqld.log shows
<95 get 49121112233
>95 sending key 49121112233
>95 END
but really strange is that calling tU = 49221112233 is causing quite
opposite results:
- following error is shown
DBG:core:cachedb_fetch: from script [memcached] - with grp [default]
ERROR:cachedb_memcached:wrap_memcached_get: Failed to get: SYSTEM ERROR
- no mysqld debug is produced
The last one example(tU = 49221112234)is failing with the same error.
Memcached is loaded with all those data
Connected to localhost.
Escape character is '^]'.
get 49221112233
VALUE 49221112233 0 11
49221112233
END
get 49221112234
VALUE 49221112234 0 11
49221112234
END
but because of some reasons memcached module is not utilized.
As aforementioned, opensips script does not have any $rU filtering setup,
so should query for any data it is asked for.
Maybe i am wrong with some of my assumptions or the way memcached is
configured, so kindly help me to understand where the problem is located.
Thanks
Maciej.
2016-11-15 18:09 GMT+01:00 Bogdan-Andrei Iancu <bog...@opensips.org>:
> OK, thank you for the update Maciej,
>
> Best regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>
> On 15.11.2016 18:28, Maciej Bylica wrote:
>
> Hi Bogdan,
>
> Thanks for reply.
> Right, Opensips module was not the source of the problem.
> I've managed to solve the issue, memcache is working fine.
>
> Thanks
> Maciej.
>
> 2016-11-10 12:56 GMT+01:00 Bogdan-Andrei Iancu <bog...@opensips.org>:
>
>> Hi Maciej,
>>
>> As I see, you are manually compiling and installing the memcached stuff -
>> any special reason for doing that ? (versus using packages)
>>
>> As the problem seems to be in the lib, not in the OpenSIPS module.
>>
>> Regards,
>>
>> Bogdan-Andrei Iancu
>> OpenSIPS Founder and Developerhttp://www.opensips-solutions.com
>>
>> On 09.11.2016 18:41, Maciej Bylica wrote:
>>
>> Hello I am struggling with memcached installation with the latest git
>> opensips 2.2.2 and centos 6.8 Here are version releases i am using:
>> libmemcached-1.0.18 (./configure, make && make install) memcached-1.4.33
>> (./configure, make && make install) with
>> LD_LIBRARY_PATH=/usr/local/lib:$LD_LIBRARY_PATH
>> memcached -d -u nobody -m 1048 -p 127.0.0.1 does not produce any error
>> but what is really puzzling me during the opensips start is the error
>> below: DBG:core:load_module: loading module
>> /usr/local/lib64/opensips/modules/cachedb_memcached.so
>> ERROR:core:sr_load_module: could not open module
>> :
>> /usr/local/lib/libmemcached.so.11: undefined symbol: pthread_once Can
>> someone please guide me how to put memcached up and running ?
>> Opensips is compiled with cachedb_memcached module.
>> Thanks in advance.
>> Maciej
>>
>> ___
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>> listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
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Re: [OpenSIPS-Users] CACHEDB_MEMCACHED Module - libmemcached undefined symbol issue
Hi Bogdan, Thanks for reply. Right, Opensips module was not the source of the problem. I've managed to solve the issue, memcache is working fine. Thanks Maciej. 2016-11-10 12:56 GMT+01:00 Bogdan-Andrei Iancu <bog...@opensips.org>: > Hi Maciej, > > As I see, you are manually compiling and installing 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 09.11.2016 18:41, Maciej Bylica wrote: > > 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:$LD_LIBRARY_PATH > > memcached -d -u nobody -m 1048 -p 127.0.0.1 > does not produce any error > > but what is really puzzling me during the opensips start is the error > below: > DBG:core:load_module: loading module /usr/local/lib64/opensips/ > modules/cachedb_memcached.so > ERROR:core:sr_load_module: could not open module > : > /usr/local/lib/libmemcached.so.11: undefined symbol: pthread_once > > Can someone please guide me how to put memcached up and running ? > Opensips is compiled with cachedb_memcached module. > > Thanks in advance. > Maciej > > > ___ > Users mailing > listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] CACHEDB_MEMCACHED Module - libmemcached undefined symbol issue
Hello I am struggling with memcached installation with the latest git opensips 2.2.2 and centos 6.8 Here are version releases i am using: libmemcached-1.0.18 (./configure, make && make install) memcached-1.4.33 (./configure, make && make install) with LD_LIBRARY_PATH=/usr/local/lib:$LD_LIBRARY_PATH memcached -d -u nobody -m 1048 -p 127.0.0.1 does not produce any error but what is really puzzling me during the opensips start is the error below: DBG:core:load_module: loading module /usr/local/lib64/opensips/modules/cachedb_memcached.so ERROR:core:sr_load_module: could not open module : /usr/local/lib/libmemcached.so.11: undefined symbol: pthread_once Can someone please guide me how to put memcached up and running ? Opensips is compiled with cachedb_memcached module. Thanks in advance. Maciej ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Class 5 & softphone supporting ZRTP
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 advanced Maciej. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Contact header modifications
Hi Răzvan, Thanks for clarifications. Maciej. 2014-09-29 10:23 GMT+02:00 Răzvan Crainea raz...@opensips.org: Hi, Maciej! The behavior you are describing is exactly how OpenSIPS should behave, so it's nothing wrong with your setup. The second affirmation is also right, if you want to 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: Hi, Guys could i ask you to share your experience here Thanks. 2014-09-25 23:00 GMT+02:00 Maciej Bylica mb...@gazeta.pl: 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 for instance) without Record-Route header. Opensips generates additional Record-Route header but doesn't modify Contact header at all and such request is sent to terminator. As an after-effect all subsequent requests properly match UAs (thanks to the rule hat RR overrides Contact header). First of all is this how Opensips behaves and there is nothing to worry about? What if i dont want to disclose Contact header information passing transparently to the other side. I assume that i may use B2B modules or topology-hiding within dialog module or setup Freeswitch for this purpose, am i right? Maybe I should play around with opensips script a little to modify that header? Thanks in advance, Maciej ___ Users mailing listUsers@lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Contact header modifications
Hi, Guys could i ask you to share your experience here Thanks. 2014-09-25 23:00 GMT+02:00 Maciej Bylica mb...@gazeta.pl: 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 for instance) without Record-Route header. Opensips generates additional Record-Route header but doesn't modify Contact header at all and such request is sent to terminator. As an after-effect all subsequent requests properly match UAs (thanks to the rule hat RR overrides Contact header). First of all is this how Opensips behaves and there is nothing to worry about? What if i dont want to disclose Contact header information passing transparently to the other side. I assume that i may use B2B modules or topology-hiding within dialog module or setup Freeswitch for this purpose, am i right? Maybe I should play around with opensips script a little to modify that header? Thanks in advance, Maciej ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Contact header modifications
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 for instance) without Record-Route header. Opensips generates additional Record-Route header but doesn't modify Contact header at all and such request is sent to terminator. As an after-effect all subsequent requests properly match UAs (thanks to the rule hat RR overrides Contact header). First of all is this how Opensips behaves and there is nothing to worry about? What if i dont want to disclose Contact header information passing transparently to the other side. I assume that i may use B2B modules or topology-hiding within dialog module or setup Freeswitch for this purpose, am i right? Maybe I should play around with opensips script a little to modify that header? Thanks in advance, Maciej ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Miliseconds precision for accounting module
Could somebody tell me a few words answering on my questions? Thanks. 2014-04-30 12:46 GMT+02:00 Maciej Bylica mb...@gazeta.pl: 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 Bylica mb...@gazeta.pl: 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, ...) Thanks Mac. 2014-04-28 3:36 GMT+02:00 Aamir aamir_...@yahoo.com: Is there a need ? Thanks Regards, Aamir Chougule Cell: 08097989101 Skype-ID: aamir_ryu --- Sent from my BlackBerry --- -Original Message- From: a...@ag-projects.com Sender: users-boun...@lists.opensips.org Date: Sun, 27 Apr 2014 19:18:04 To: OpenSIPS users mailling listusers@lists.opensips.org Reply-To: OpenSIPS users mailling list users@lists.opensips.org Subject: Re: [OpenSIPS-Users] Miliseconds precision for accounting module ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Roberto Spadim SPAEmpresarial Eng. Automação e Controle ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Miliseconds precision for accounting module
Hello,
ad1) I am just wondering why get_timestamp must be fired before has_totag
part of the script?
I've found some threads on discussion group describing the same thing, but
without explaination.
ad2) i have set following rule:
if (is_method(INVITE) t_check_status(200) ) {
xlog(L_INFO,[INFO] Inside okay - $var(okay));
get_timestamp($avp(sec),$avp(usec));
}
then i think reINVITE/OK/ transaction will generate new timestamps, which
is wrong.
Is there any dialog variable that could be checked and then set inside the
{ } to last more than just one transaction.
Possible usage inside onreply_route...
if (is_method(INVITE) (t_check_status(200))
($var(okay)==NULL))
{
$var(okay)=1;
xlog(L_INFO,[INFO] Inside okay - $var(okay));
get_timestamp($avp(sec),$avp(usec));
}
Thanks
Mac.
2014-04-15 17:04 GMT+02:00 Maciej Bylica mb...@gazeta.pl:
Hello,
It works, but:
1) get_timestamp doesnt work inside has_totag section
if (has_totag()) {
if (loose_route()) {
if (is_method(BYE)) {
get_timestamp($avp(secbye),$avp(usecbye));
.
.
but works if called before that section
2) because i need to count duration, i should rather place it inside
onreply_route
if (t_check_status(200)) {
get_timestamp($avp(sec),$avp(usec));
}
but the question is how it will behave in case of reINVITE is triggered
from the originating side.
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 Maciej Bylica mb...@gazeta.pl:
Hi Vlad,
Thanks for reply.
I am using OpenSIPS (1.9.1-notls (x86_64/linux)) so get_timestamp is
available there.
Let me check this.
Regards,
Mac
2014-04-14 10:57 GMT+02:00 Vlad Paiu vladp...@opensips.org:
Hello,
Which OpenSIPS version are you using ?
You could use get_timestamp [1] from the Core to get the current second
and microsecond,
and set the two variables at INVITE time, and set them as db_extra [2] .
Then, at BYE time call again the get_timestamp function, store them in
some AVPs and set those AVPs in [3]. This way you should get both the
INVITE and BYE timestamps with microseconds precision in the CDR record.
[1]
http://www.opensips.org/Documentation/Script-CoreFunctions-1-10#toc18
[2] http://www.opensips.org/html/docs/modules/1.10.x/acc.html#id295028
[3] http://www.opensips.org/html/docs/modules/1.10.x/acc.html#id295056
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 call.
Couldn't i just use two extra db variables which will gather the $time
inside INVITE {} and BYE {}?
Thanks,
Mac
2014-04-12 6:39 GMT+02:00 Ryan Mitchell r...@tcl.net:
Hello Mac,
Each row in the acc table is for a transaction. To make a proper CDR
out of the data, you have to combine rows to find the start and end of the
call. That can be harder than it sounds, especially with forking
(parallel, or the more common case of serial forking when you are LCR
routing or simply sending calls to alt destinations after a timeout). I
wrote scripts that implement a simple dialog state machine to make sense of
all the distinct legs of a call, though there should be an easier way with
the auto-cdr / multi call-legs accounting feature of the acc module (anyone
comment on this please?).
The time field in the acc table will be the timestamp of the response
for the given transaction. If you assign an extra field for another
timestamp, it will depend on where you assign that var in your script. In
my 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 mb...@gazeta.plwrote:
Ryan,
One more question.
Currently i have some db extra attrs setup. My acc table looks like
following:
++--+--+-+-++
| Field | Type | Null | Key | Default | Extra
|
++--+--+-+-++
| id | int(10) unsigned | NO | PRI | NULL|
auto_increment |
| method | char(16) | NO | | |
|
| from_tag | char(64) | NO | | |
|
| to_tag | char(64) | NO | | |
|
| callid | char(64) | NO | MUL | |
|
| sip_code | char(3) | NO | | |
|
| sip_reason | char(32
Re: [OpenSIPS-Users] Call Generator
Hi Adrian,
Sry for delay. I must have overlooked your email.
I have done it with Freeswitch plus some extra Java scripts that manage the
way FS is triggering INVITEs.
Anyway, you've said that there are some tools
It is used for stress testing, billing tests.
Thanks.
2013-07-17 15:17 GMT+02:00 Adrian Georgescu a...@ag-projects.com:
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 mb...@gazeta.pl 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 + RTP calls. There must be many .wav or mp3 files
possible to be used
- heaving random call duration
- heaving a possibility to set Called and Called numbers random in
specified ranges (like 1122233[0-9]{4} for instance).
Have you tested any call generator that has aforementioned functionality
implemented?
I know that Opensips could be used for this purpose, but i am looking for
the ready-to-run product.
Thanks,
Mac.
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Re: [OpenSIPS-Users] Miliseconds precision for accounting module
Hello,
It works, but:
1) get_timestamp doesnt work inside has_totag section
if (has_totag()) {
if (loose_route()) {
if (is_method(BYE)) {
get_timestamp($avp(secbye),$avp(usecbye));
.
.
but works if called before that section
2) because i need to count duration, i should rather place it inside
onreply_route
if (t_check_status(200)) {
get_timestamp($avp(sec),$avp(usec));
}
but the question is how it will behave in case of reINVITE is triggered
from the originating side.
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 Maciej Bylica mb...@gazeta.pl:
Hi Vlad,
Thanks for reply.
I am using OpenSIPS (1.9.1-notls (x86_64/linux)) so get_timestamp is
available there.
Let me check this.
Regards,
Mac
2014-04-14 10:57 GMT+02:00 Vlad Paiu vladp...@opensips.org:
Hello,
Which OpenSIPS version are you using ?
You could use get_timestamp [1] from the Core to get the current second
and microsecond,
and set the two variables at INVITE time, and set them as db_extra [2] .
Then, at BYE time call again the get_timestamp function, store them in
some AVPs and set those AVPs in [3]. This way you should get both the
INVITE and BYE timestamps with microseconds precision in the CDR record.
[1] http://www.opensips.org/Documentation/Script-CoreFunctions-1-10#toc18
[2] http://www.opensips.org/html/docs/modules/1.10.x/acc.html#id295028
[3] http://www.opensips.org/html/docs/modules/1.10.x/acc.html#id295056
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 call.
Couldn't i just use two extra db variables which will gather the $time
inside INVITE {} and BYE {}?
Thanks,
Mac
2014-04-12 6:39 GMT+02:00 Ryan Mitchell r...@tcl.net:
Hello Mac,
Each row in the acc table is for a transaction. To make a proper CDR
out of the data, you have to combine rows to find the start and end of the
call. That can be harder than it sounds, especially with forking
(parallel, or the more common case of serial forking when you are LCR
routing or simply sending calls to alt destinations after a timeout). I
wrote scripts that implement a simple dialog state machine to make sense of
all the distinct legs of a call, though there should be an easier way with
the auto-cdr / multi call-legs accounting feature of the acc module (anyone
comment on this please?).
The time field in the acc table will be the timestamp of the response
for the given transaction. If you assign an extra field for another
timestamp, it will depend on where you assign that var in your script. In
my case I assign it in the main routing section so the timestamp indicates
the start of the transaction.
best regards,
Ryan
On Fri, Apr 11, 2014 at 10:06 AM, Maciej Bylica mb...@gazeta.pl wrote:
Ryan,
One more question.
Currently i have some db extra attrs setup. My acc table looks like
following:
++--+--+-+-++
| Field | Type | Null | Key | Default | Extra
|
++--+--+-+-++
| id | int(10) unsigned | NO | PRI | NULL| auto_increment
|
| method | char(16) | NO | | |
|
| from_tag | char(64) | NO | | |
|
| to_tag | char(64) | NO | | |
|
| callid | char(64) | NO | MUL | |
|
| sip_code | char(3) | NO | | |
|
| sip_reason | char(32) | NO | | |
|
| time | datetime | NO | | NULL|
|
| duration | int(11) unsigned | NO | | 0 |
|
| setuptime | int(11) unsigned | NO | | 0 |
|
| SourceAddr | char(30) | NO | | NULL|
|
| DestAddr | char(30) | NO | | NULL|
|
| Anum | char(30) | NO | | NULL|
|
| Bnum_rU| char(30) | NO | | NULL|
|
| Bnum_tU| char(30) | NO | | NULL|
|
| created| datetime | YES | | NULL|
|
++--+--+-+-++
modparam(acc, db_extra, SourceAddr=$si; DestAddr=$rd; Anum=$fU;
Bnum_rU=$rU; Bnum_tU=$tU)
Now using additional data like $time will give me the exact moment
the call is ended, nothing more, am i right?
To have detailed call duration i need to know exact answer and
disconnect timestamps.
Btw: i am using OpenSIPS (1.9.1-notls
Re: [OpenSIPS-Users] Miliseconds precision for accounting module
Hi Vlad,
Thanks for reply.
I am using OpenSIPS (1.9.1-notls (x86_64/linux)) so get_timestamp is
available there.
Let me check this.
Regards,
Mac
2014-04-14 10:57 GMT+02:00 Vlad Paiu vladp...@opensips.org:
Hello,
Which OpenSIPS version are you using ?
You could use get_timestamp [1] from the Core to get the current second
and microsecond,
and set the two variables at INVITE time, and set them as db_extra [2] .
Then, at BYE time call again the get_timestamp function, store them in
some AVPs and set those AVPs in [3]. This way you should get both the
INVITE and BYE timestamps with microseconds precision in the CDR record.
[1] http://www.opensips.org/Documentation/Script-CoreFunctions-1-10#toc18
[2] http://www.opensips.org/html/docs/modules/1.10.x/acc.html#id295028
[3] http://www.opensips.org/html/docs/modules/1.10.x/acc.html#id295056
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 call.
Couldn't i just use two extra db variables which will gather the $time
inside INVITE {} and BYE {}?
Thanks,
Mac
2014-04-12 6:39 GMT+02:00 Ryan Mitchell r...@tcl.net:
Hello Mac,
Each row in the acc table is for a transaction. To make a proper CDR
out of the data, you have to combine rows to find the start and end of the
call. That can be harder than it sounds, especially with forking
(parallel, or the more common case of serial forking when you are LCR
routing or simply sending calls to alt destinations after a timeout). I
wrote scripts that implement a simple dialog state machine to make sense of
all the distinct legs of a call, though there should be an easier way with
the auto-cdr / multi call-legs accounting feature of the acc module (anyone
comment on this please?).
The time field in the acc table will be the timestamp of the response
for the given transaction. If you assign an extra field for another
timestamp, it will depend on where you assign that var in your script. In
my case I assign it in the main routing section so the timestamp indicates
the start of the transaction.
best regards,
Ryan
On Fri, Apr 11, 2014 at 10:06 AM, Maciej Bylica mb...@gazeta.pl wrote:
Ryan,
One more question.
Currently i have some db extra attrs setup. My acc table looks like
following:
++--+--+-+-++
| Field | Type | Null | Key | Default | Extra |
++--+--+-+-++
| id | int(10) unsigned | NO | PRI | NULL| auto_increment |
| method | char(16) | NO | | ||
| from_tag | char(64) | NO | | ||
| to_tag | char(64) | NO | | ||
| callid | char(64) | NO | MUL | ||
| sip_code | char(3) | NO | | ||
| sip_reason | char(32) | NO | | ||
| time | datetime | NO | | NULL||
| duration | int(11) unsigned | NO | | 0 ||
| setuptime | int(11) unsigned | NO | | 0 ||
| SourceAddr | char(30) | NO | | NULL||
| DestAddr | char(30) | NO | | NULL||
| Anum | char(30) | NO | | NULL||
| Bnum_rU| char(30) | NO | | NULL||
| Bnum_tU| char(30) | NO | | NULL||
| created| datetime | YES | | NULL||
++--+--+-+-++
modparam(acc, db_extra, SourceAddr=$si; DestAddr=$rd; Anum=$fU;
Bnum_rU=$rU; Bnum_tU=$tU)
Now using additional data like $time will give me the exact moment the
call is ended, nothing more, am i right?
To have detailed call duration i need to know exact answer and
disconnect timestamps.
Btw: i am using OpenSIPS (1.9.1-notls (x86_64/linux))
Thanks,
Mac
2014-04-10 22:03 GMT+02:00 Ryan Mitchell r...@tcl.net:
Using db_extra to stuff custom data into 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 mb...@gazeta.plwrote:
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
Re: [OpenSIPS-Users] Miliseconds precision for accounting module
Hello Ryan,
I am using dialog accounting, so each row is fully qualified cdr record,
not only single transaction of a call.
Couldn't i just use two extra db variables which will gather the $time
inside INVITE {} and BYE {}?
Thanks,
Mac
2014-04-12 6:39 GMT+02:00 Ryan Mitchell r...@tcl.net:
Hello Mac,
Each row in the acc table is for a transaction. To make a proper CDR out
of the data, you have to combine rows to find the start and end of the
call. That can be harder than it sounds, especially with forking
(parallel, or the more common case of serial forking when you are LCR
routing or simply sending calls to alt destinations after a timeout). I
wrote scripts that implement a simple dialog state machine to make sense of
all the distinct legs of a call, though there should be an easier way with
the auto-cdr / multi call-legs accounting feature of the acc module (anyone
comment on this please?).
The time field in the acc table will be the timestamp of the response for
the given transaction. If you assign an extra field for another timestamp,
it will depend on where you assign that var in your script. In my case I
assign it in the main routing section so the timestamp indicates the start
of the transaction.
best regards,
Ryan
On Fri, Apr 11, 2014 at 10:06 AM, Maciej Bylica mb...@gazeta.pl wrote:
Ryan,
One more question.
Currently i have some db extra attrs setup. My acc table looks like
following:
++--+--+-+-++
| Field | Type | Null | Key | Default | Extra |
++--+--+-+-++
| id | int(10) unsigned | NO | PRI | NULL| auto_increment |
| method | char(16) | NO | | ||
| from_tag | char(64) | NO | | ||
| to_tag | char(64) | NO | | ||
| callid | char(64) | NO | MUL | ||
| sip_code | char(3) | NO | | ||
| sip_reason | char(32) | NO | | ||
| time | datetime | NO | | NULL||
| duration | int(11) unsigned | NO | | 0 ||
| setuptime | int(11) unsigned | NO | | 0 ||
| SourceAddr | char(30) | NO | | NULL||
| DestAddr | char(30) | NO | | NULL||
| Anum | char(30) | NO | | NULL||
| Bnum_rU| char(30) | NO | | NULL||
| Bnum_tU| char(30) | NO | | NULL||
| created| datetime | YES | | NULL||
++--+--+-+-++
modparam(acc, db_extra, SourceAddr=$si; DestAddr=$rd; Anum=$fU;
Bnum_rU=$rU; Bnum_tU=$tU)
Now using additional data like $time will give me the exact moment the
call is ended, nothing more, am i right?
To have detailed call duration i need to know exact answer and disconnect
timestamps.
Btw: i am using OpenSIPS (1.9.1-notls (x86_64/linux))
Thanks,
Mac
2014-04-10 22:03 GMT+02:00 Ryan Mitchell r...@tcl.net:
Using db_extra to stuff custom data into 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 mb...@gazeta.pl wrote:
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 directly on DB level.
Could somebody give me a hand.
Thanks in advanced
Mac
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Re: [OpenSIPS-Users] Miliseconds precision for accounting module
Hi Ryan, Thanks for prompt reply. I am about to check this out. Mac. 2014-04-10 22:03 GMT+02:00 Ryan Mitchell r...@tcl.net: Using db_extra to stuff custom data into your acc table, use the $time var with a format such as %s.%N or similar. Or, as you suggested, do it on the database level with a trigger or auto-update column. On Thu, Apr 10, 2014 at 10:01 AM, Maciej Bylica mb...@gazeta.pl wrote: Hello I just want to know how to achieve miliseconds precision for accounting module. This is quite important while trying to sum up total traffic duration with the accuracy of hundred of ms. As i see there is no rounding feature implemented as well, but heaving ms precision it could be done directly on DB level. Could somebody give me a hand. Thanks in advanced Mac ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Ryan Mitchell r...@tcl.net Telecom Logic, LLC ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Miliseconds precision for accounting module
Ryan, One more question. Currently i have some db extra attrs setup. My acc table looks like following: ++--+--+-+-++ | Field | Type | Null | Key | Default | Extra | ++--+--+-+-++ | id | int(10) unsigned | NO | PRI | NULL| auto_increment | | method | char(16) | NO | | || | from_tag | char(64) | NO | | || | to_tag | char(64) | NO | | || | callid | char(64) | NO | MUL | || | sip_code | char(3) | NO | | || | sip_reason | char(32) | NO | | || | time | datetime | NO | | NULL|| | duration | int(11) unsigned | NO | | 0 || | setuptime | int(11) unsigned | NO | | 0 || | SourceAddr | char(30) | NO | | NULL|| | DestAddr | char(30) | NO | | NULL|| | Anum | char(30) | NO | | NULL|| | Bnum_rU| char(30) | NO | | NULL|| | Bnum_tU| char(30) | NO | | NULL|| | created| datetime | YES | | NULL|| ++--+--+-+-++ modparam(acc, db_extra, SourceAddr=$si; DestAddr=$rd; Anum=$fU; Bnum_rU=$rU; Bnum_tU=$tU) Now using additional data like $time will give me the exact moment the call is ended, nothing more, am i right? To have detailed call duration i need to know exact answer and disconnect timestamps. Btw: i am using OpenSIPS (1.9.1-notls (x86_64/linux)) Thanks, Mac 2014-04-10 22:03 GMT+02:00 Ryan Mitchell r...@tcl.net: Using db_extra to stuff custom data into 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 mb...@gazeta.pl wrote: 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 directly on DB level. Could somebody give me a hand. Thanks in advanced Mac ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Ryan Mitchell r...@tcl.net Telecom Logic, LLC ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Miliseconds precision for accounting module
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 directly on DB level. Could somebody give me a hand. Thanks in advanced Mac ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] drouting get_group_id behavior
Hi Alexander, Thanks for feedback. 2014-02-27 3:09 GMT+01:00 Alexander Mustafin mustafin.aleksa...@gmail.com: Hi! Bacause drouting do not accept .* in dr_rules - you may use dialplan module for this. Just catch domain name with regex and $avp(dest) will store name of rule for drouting, as example. С уважением, Александр Мустафин mustafin.aleksa...@gmail.com 26 февр. 2014 г., в 20:45, Maciej Bylica mb...@gazeta.pl написал(а): Hello, Thanks for reply. Yeah i did it by asking db for.. avp_db_query(SELECT groupid FROM dr_groups WHERE domain = '$fd',$avp(i:600)); and then using exactly the same avp for do_routing. It works, but i am still wondering how to match domain different way ( do_routing()) Thanks. 2014-02-25 18:52 GMT+01:00 steph...@shimaore.net: Hello, I have the same problem on 1.9 rel. | id | username | domain | groupid | description | | 4 | .* | 10.10.10.5 | 0 | TEST If you don't need to match on username why not pass directly the groupid to `do_routing` ? do_routing(0); If you need to dynamically map between a domain and a groupid, use e.g. do_routing($avp(10)); and an AVP table which maps from domains to groupid. S. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] drouting get_group_id behavior
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 18:52 GMT+01:00 steph...@shimaore.net: Hello, I have the same problem on 1.9 rel. | id | username | domain | groupid | description | | 4 | .* | 10.10.10.5 | 0 | TEST If you don't need to match on username why not pass directly the groupid to `do_routing` ? do_routing(0); If you need to dynamically map between a domain and a groupid, use e.g. do_routing($avp(10)); and an AVP table which maps from domains to groupid. S. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] drouting get_group_id behavior
Hello, I have the same problem on 1.9 rel. ++-+--+---++ | id | username | domain | groupid | description | ++-+--+---++ | 4 | .* | 10.10.10.5| 0 | TEST | ++-++-+---+ unfortunately i am getting ERROR:drouting:get_group_id: no group for user 221112233@10.10.10.5 db is asking for select groupid from dr_groups where username='221112233' AND domain='10.10.10.5' How to define ANY username to have this working? Thanks Mac. 2013-09-26 19:48 GMT+02:00 Александр Мустафин mustafin.aleksa...@gmail.com : I'm using 1.8.1 version. I read docs for 1.10.x version and don't see any difference. Best regards, Alexander Mustafin mustafin.aleksa...@gmail.com 26.09.2013, в 18:52, Nick Cameo sym...@gmail.com написал(а): Not a bug. Which version of OpenSIPS are you using? If it is 1.8/9, it does not support regular expressions for dr_groups. dr_groups maps users to routes. Not really groups Nick. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] OK message inspection inside the dialog
Thanks Muhammad, My t_on_reply pointed to wrong onreply_route, that was the problem. Again thank you for advice, so i can get back to that part of my script. Mac. 2013/7/14 Muhammad Shahzad shaherya...@gmail.com In reply_route where all replies from endpoints are receive, you should be able to filter 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 mb...@gazeta.pl wrote: 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 there any way to get into 180/183/200 messages? Thanks, Mac ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Mit freundlichen Grüßen Muhammad Shahzad --- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +49 176 99 83 10 85 MSN: shari_78...@hotmail.com Email: shaherya...@googlemail.com ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] OK message inspection inside the dialog
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 there any way to get into 180/183/200 messages? Thanks, Mac ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Call Generator
Thanks Adrian,
I will take a look on this...
Mac.
2013/6/18 Adrian Georgescu a...@ag-projects.com
See sipclient package, it contains this tool that does pretty much all you
are looking for:
http://sipsimpleclient.org/projects/sipsimpleclient/wiki/Sip_audio_session
Adrian
On Jun 18, 2013, at 7:03 PM, Maciej Bylica mb...@gazeta.pl 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 + RTP calls. There must be many .wav or mp3 files
possible to be used
- heaving random call duration
- heaving a possibility to set Called and Called numbers random in
specified ranges (like 1122233[0-9]{4} for instance).
Have you tested any call generator that has aforementioned functionality
implemented?
I know that Opensips could be used for this purpose, but i am looking for
the ready-to-run product.
Thanks,
Mac.
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[OpenSIPS-Users] Call Generator
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 duration
- heaving a possibility to set Called and Called numbers random in
specified ranges (like 1122233[0-9]{4} for instance).
Have you tested any call generator that has aforementioned functionality
implemented?
I know that Opensips could be used for this purpose, but i am looking for
the ready-to-run product.
Thanks,
Mac.
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Re: [OpenSIPS-Users] Modify Via header
Thanks Dani for prompt feedback. I will take a look on this. Maciej. 2012/1/10 Dani Popa dani.p...@gmail.com: 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 mb...@gazeta.pl wrote: 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 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Dani Popa ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Modify Via header
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 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] SIP messages delayed
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.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] SIP messages delayed
, Maciej Bylica wrote: 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 127.0.0.16SIPpTag0017 WARNING:core:log_expiry: threshold exceeded : msg processing took too long - 927067 us.Source : INVITE sip:21133@X1.X1.X1.X1:5060 SIP/2.0 =rtpmap:0 PCMU/8000 087637 IN IP4 127.0.0.16SIPpTag0013 WARNING:core:log_expiry: threshold exceeded : msg processing took too long - 300760 us.Source : INVITE sip:21133@X1.X1.X1.X1:5060 SIP/2.0 =rtpmap:0 PCMU/8000 087637 IN IP4 127.0.0.16SIPpTag003: WARNING:core:log_expiry: threshold exceeded : msg processing took too long - 267328 us.Source : INVITE sip:21133@X1.X1.X1.X1:5060 SIP/2.0 =rtpmap:0 PCMU/8000 087637 IN IP4 127.0.0.16SIPpTag0018 WARNING:core:log_expiry: threshold exceeded : msg processing took too long - 860686 us.Source : INVITE sip:21133@X1.X1.X1.X1:5060 SIP/2.0 =rtpmap:0 PCMU/8000 087637 IN IP4 127.0.0.16SIPpTag0014 WARNING:core:log_expiry: threshold exceeded : msg processing took too long - 284254 us.Source : INVITE sip:21133@X1.X1.X1.X1:5060 SIP/2.0 Core parameters were as follows: exec_dns_threshold=6 exec_msg_threshold=6 What is more number of awaken processes were as below: opensipsctl fifo get_statistics udp:X1.X1.X1.X1:5060-load load:udp:X1.X1.X1.X1:5060-load = 100 opensipsctl fifo get_statistics udp:X1.X1.X1.X1:5060-load load:udp:X1.X1.X1.X14:5060-load = 100 opensipsctl fifo get_statistics udp:X1.X1.X1.X1:5060-load load:udp:X1.X1.X1.X1:5060-load = 100 opensipsctl fifo get_statistics udp:X1.X1.X1.X1:5060-load load:udp:X1.X1.X1.X1:5060-load = 100 opensipsctl fifo get_statistics udp:X1.X1.X1.X1:5060-load load:udp:X1.X1.X1.X1:5060-load = 25 opensipsctl fifo get_statistics udp:X1.X1.X1.X1:5060-load load:udp:X1.X1.X1.X1:5060-load = 0 CPS was more less four, so quite low. Where the problem might be located? :( Thanks, Maciej 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 omit dns names, the after effect was exactly the same. But i am about to proceed with core parameters as you described and give you feedback here. Thanks, Maciej Hello, Try to use the exec_dns_threshold [1] and the exec_msg_threshold [2] core parameters, as well as the exec_query_threshold parameter in the db_mysql module to try and see which component is determining the delay ( whether it's the DNS, MySQL or some other things in your config ). [1] http://www.opensips.org/Resources/DocsCoreFcn#toc49 [2] http://www.opensips.org/Resources/DocsCoreFcn#toc50 [3] http://www.opensips.org/html/docs/modules/devel/db_mysql.html#id249058 Regards, Vlad Paiu OpenSIPS Developer On 12/02/2011 09:13 PM, logan wrote: Are you using a DB back end for ACC? If so which one? I've seen instances where using MySQL under heavy load w/o optimizing your mysql config can cause messages to hang in OpenSIPs while it's waiting for a mysql resource to write down its existing ACC or Missed_Call records. -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/SIP-messages-delayed-tp7054793p7055831.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] SIP messages delayed
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 127.0.0.16SIPpTag0017 WARNING:core:log_expiry: threshold exceeded : msg processing took too long - 927067 us.Source : INVITE sip:21133@X1.X1.X1.X1:5060 SIP/2.0 =rtpmap:0 PCMU/8000 087637 IN IP4 127.0.0.16SIPpTag0013 WARNING:core:log_expiry: threshold exceeded : msg processing took too long - 300760 us.Source : INVITE sip:21133@X1.X1.X1.X1:5060 SIP/2.0 =rtpmap:0 PCMU/8000 087637 IN IP4 127.0.0.16SIPpTag003: WARNING:core:log_expiry: threshold exceeded : msg processing took too long - 267328 us.Source : INVITE sip:21133@X1.X1.X1.X1:5060 SIP/2.0 =rtpmap:0 PCMU/8000 087637 IN IP4 127.0.0.16SIPpTag0018 WARNING:core:log_expiry: threshold exceeded : msg processing took too long - 860686 us.Source : INVITE sip:21133@X1.X1.X1.X1:5060 SIP/2.0 =rtpmap:0 PCMU/8000 087637 IN IP4 127.0.0.16SIPpTag0014 WARNING:core:log_expiry: threshold exceeded : msg processing took too long - 284254 us.Source : INVITE sip:21133@X1.X1.X1.X1:5060 SIP/2.0 Core parameters were as follows: exec_dns_threshold=6 exec_msg_threshold=6 What is more number of awaken processes were as below: opensipsctl fifo get_statistics udp:X1.X1.X1.X1:5060-load load:udp:X1.X1.X1.X1:5060-load = 100 opensipsctl fifo get_statistics udp:X1.X1.X1.X1:5060-load load:udp:X1.X1.X1.X14:5060-load = 100 opensipsctl fifo get_statistics udp:X1.X1.X1.X1:5060-load load:udp:X1.X1.X1.X1:5060-load = 100 opensipsctl fifo get_statistics udp:X1.X1.X1.X1:5060-load load:udp:X1.X1.X1.X1:5060-load = 100 opensipsctl fifo get_statistics udp:X1.X1.X1.X1:5060-load load:udp:X1.X1.X1.X1:5060-load = 25 opensipsctl fifo get_statistics udp:X1.X1.X1.X1:5060-load load:udp:X1.X1.X1.X1:5060-load = 0 CPS was more less four, so quite low. Where the problem might be located? :( Thanks, Maciej 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 omit dns names, the after effect was exactly the same. But i am about to proceed with core parameters as you described and give you feedback here. Thanks, Maciej Hello, Try to use the exec_dns_threshold [1] and the exec_msg_threshold [2] core parameters, as well as the exec_query_threshold parameter in the db_mysql module to try and see which component is determining the delay ( whether it's the DNS, MySQL or some other things in your config ). [1] http://www.opensips.org/Resources/DocsCoreFcn#toc49 [2] http://www.opensips.org/Resources/DocsCoreFcn#toc50 [3] http://www.opensips.org/html/docs/modules/devel/db_mysql.html#id249058 Regards, Vlad Paiu OpenSIPS Developer On 12/02/2011 09:13 PM, logan wrote: Are you using a DB back end for ACC? If so which one? I've seen instances where using MySQL under heavy load w/o optimizing your mysql config can cause messages to hang in OpenSIPs while it's waiting for a mysql resource to write down its existing ACC or Missed_Call records. -- View this message in context: http://opensips-open-sip-server.1449251.n2.nabble.com/SIP-messages-delayed-tp7054793p7055831.html Sent from the OpenSIPS - Users mailing list archive at Nabble.com. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] SIP messages delayed
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 interface X1.X1.X1.X1 to ip address Z.Z.Z.Z through interface X2.X2.X2.X2. (Y.Y.Y.Y --Opensips(X1.X1.X1.X1)-Opensips(X2.X2.X2.X2)--Z.Z.Z.Z) The problem i have encountered is that quite frequently (30%-50% of all messages) sip messages are buffered and delayed. As a after-effect traffic origination server on ip Y.Y.Y.Y is resending messages (INVITE for instance). After 5-30secs Opensips is responding to aforementioned INVITEs to Y.Y.Y.Y. The same situation is between Opensips(X2.X2.X2.X2)--Z.Z.Z.Z Example: 1 - originating server 2 - opensips 3 - termination server 1- INVITE -2 2- INVITE -3 3- 100 Trying -2 1- INVITE -2 1- INVITE -2 1- INVITE -2 3- 183 Session Progress -2 1- INVITE -2 1- INVITE -2 1- INVITE -2 3- 200 OK -2 3- 200 OK -2 3- 200 OK -2 3- BYE -2 3- BYE -2 2- 100 Trying -1 2- 100 Trying -1 2- 100 Trying -1 2- 100 Trying -1 2- 183 Session Progress -1 2- 200 OK -1 2- 200 OK -1 2- 200 OK -1 2- 200 OK -1 Do you know where the problem might be located? Is it hw issue? CPS in peak might be around 30CPS. In config.h i have changed: PKG_MEM_POOL_SIZE to 1024x1024x8 SHM_MEM_SIZE 768 Regards, Maciej ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Opensips as a $si differentiator
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 trigger) and to make routing mechanism like following - if $si = X.X.X.X then t_relay the call by using second interface and its subinterface Z1 - if $si = Y.Y.Y.Y then t_relay the call by using second interface and its subinterface Z2 and so on. So in other words i need to have mechanism that is checking source address of all incoming sip messages and then make decision which subinterface to use for t_relay. Each subinterface will be heaving different ip address to achieve customer differentiation. I am puzzling over how to build such a mechanism, which modules to use? Could you please give me some hints here. Thanks, Maciej ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Opensips as a $si differentiator
Thanks for prompt feedback.
That is what i was looking for.
Cheers
Maciej
Hello,
a solution would be checking for the sourceip and using force_send_socket()
to set a different interface, which will be used by t_relay.
if ($si=~^108\.109\.180\. || $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 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 trigger) and to make routing mechanism like following
- if $si = X.X.X.X then t_relay the call by using second interface and
its subinterface Z1
- if $si = Y.Y.Y.Y then t_relay the call by using second interface and
its subinterface Z2
and so on.
So in other words i need to have mechanism that is checking source
address of all incoming sip messages and then make decision which
subinterface to use for t_relay.
Each subinterface will be heaving different ip address to achieve
customer differentiation.
I am puzzling over how to build such a mechanism, which modules to use?
Could you please give me some hints here.
Thanks,
Maciej
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Re: [OpenSIPS-Users] New acc module for cdr generation over Radius
Hi Vlad, Thanks for info. I am about to work on this. Thanks, Maciej 2011/4/21 Vlad Paiu vladp...@opensips.org: Hello, The new CDRs type of accounting in OpenSIPS 1.6.4 only produces one entry per call for every type of backend, whether it's a DB, Radius or Syslog. So it's natural to only have a single STOP entry per call, and not two, a start and stop entry as in the old type of accounting. The STOP packet should also contain the Sip-Call-Duration and Sip-Call-Duration attributes, defined in the 'etc/dictionary.opensips' dictionary that comes 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 feature with Mysql.etc/dictionary.opensipset 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 invite. Once it is set there i do receive only STOP radius acc packet. In case i do not have setflag set anywere in my script Opensips produce START and STOP packet properly. Does anyone knows where to look for the problem? Last question does standard STOP packet incorporate call duration attr anyhow or should i use aaa_extra in my config. My STOP packet is as follows: Sun Apr 17 21:08:38 2011 Acct-Status-Type = Stop Service-Type = IAPP-Register Sip-Response-Code = 200 Sip-Method = Bye Event-Timestamp = \266:\253M\374\212\256 Sip-From-Tag = eb759c18 Sip-To-Tag = 00-07350-027f5afd-492940963 Acct-Session-Id = b0790b4443102642ZTMzOWZlNGU0Njg4MDMwM2EzZjI1NTY5NTllNWFiYjk. User-Name = 11122233@66.66.66.66 Calling-Station-Id = sip:11122233@66.66.66.66 Called-Station-Id = sip:999887766@66.66.66.66 Sip-Translated-Request-URI = sip:77.77.77.77:5060 User-Agent = X-Lite release 1003l stamp 30942 Contact = sip:11122233@10.119.204.184:15950 NAS-Port-Id = 5060 Acct-Delay-Time = 0 NAS-IP-Address = 127.0.0.1 Client-IP-Address = 127.0.0.1 Acct-Unique-Session-Id = 7e6e2ace14ff4970 Timestamp = 1303067318 I do have the latest OS 1.6.4-2-notls revision 7872. Thx in advance for help, Maciej. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] New acc module for cdr generation over Radius
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, there is a need to use cdr_flag and setflag in initial invite. Once it is set there i do receive only STOP radius acc packet. In case i do not have setflag set anywere in my script Opensips produce START and STOP packet properly. Does anyone knows where to look for the problem? Last question does standard STOP packet incorporate call duration attr anyhow or should i use aaa_extra in my config. My STOP packet is as follows: Sun Apr 17 21:08:38 2011 Acct-Status-Type = Stop Service-Type = IAPP-Register Sip-Response-Code = 200 Sip-Method = Bye Event-Timestamp = \266:\253M\374\212\256 Sip-From-Tag = eb759c18 Sip-To-Tag = 00-07350-027f5afd-492940963 Acct-Session-Id = b0790b4443102642ZTMzOWZlNGU0Njg4MDMwM2EzZjI1NTY5NTllNWFiYjk. User-Name = 11122233@66.66.66.66 Calling-Station-Id = sip:11122233@66.66.66.66 Called-Station-Id = sip:999887766@66.66.66.66 Sip-Translated-Request-URI = sip:77.77.77.77:5060 User-Agent = X-Lite release 1003l stamp 30942 Contact = sip:11122233@10.119.204.184:15950 NAS-Port-Id = 5060 Acct-Delay-Time = 0 NAS-IP-Address = 127.0.0.1 Client-IP-Address = 127.0.0.1 Acct-Unique-Session-Id = 7e6e2ace14ff4970 Timestamp = 1303067318 I do have the latest OS 1.6.4-2-notls revision 7872. Thx in advance for help, Maciej. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] New acc module for cdr generation over Radius
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 invite. Once it is set there i do receive only STOP radius acc packet. In case i do not have setflag set anywere in my script Opensips produce START and STOP packet properly. Does anyone knows where to look for the problem? Last question does standard STOP packet incorporate call duration attr anyhow or should i use aaa_extra in my config. My STOP packet is as follows: Sun Apr 17 21:08:38 2011 Acct-Status-Type = Stop Service-Type = IAPP-Register Sip-Response-Code = 200 Sip-Method = Bye Event-Timestamp = \266:\253M\374\212\256 Sip-From-Tag = eb759c18 Sip-To-Tag = 00-07350-027f5afd-492940963 Acct-Session-Id = b0790b4443102642ZTMzOWZlNGU0Njg4MDMwM2EzZjI1NTY5NTllNWFiYjk. User-Name = 11122233@66.66.66.66 Calling-Station-Id = sip:11122233@66.66.66.66 Called-Station-Id = sip:999887766@66.66.66.66 Sip-Translated-Request-URI = sip:77.77.77.77:5060 User-Agent = X-Lite release 1003l stamp 30942 Contact = sip:11122233@10.119.204.184:15950 NAS-Port-Id = 5060 Acct-Delay-Time = 0 NAS-IP-Address = 127.0.0.1 Client-IP-Address = 127.0.0.1 Acct-Unique-Session-Id = 7e6e2ace14ff4970 Timestamp = 1303067318 I do have the latest OS 1.6.4-2-notls revision 7872. Thx in advance for help, Maciej. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Opensips Server header - how to modify
Thanks a lot for prompt answer. Changed and its working :) Thanks, Maciej 2011/2/8 Duane Larson duane.lar...@gmail.com: http://www.opensips.org/Resources/DocsCoreFcn16#toc66 On Mon, Feb 7, 2011 at 6:49 PM, David J. da...@styleflare.com wrote: You dont have to recompile; Look in the docs; 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 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:sip:xxx@11.22.33.44;tag=164494841543104e5b53d388cae71165 To:sip:yyy@77.88.99.66 Call-ID: YTBkMjRhYmExDDE2ZGRkmjQ1MmU2ODU3NzI0ODA1NjU. CSeq: 200 INVITE Server: OpenSIPS (1.6.4-2-notls (i386/linux)) Content-Length: 0 Do i need to recompile Opensips to have OpenSIPS (1.6.4-2-notls (i386/linux)) modified? Thanks, Maciej. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- -- *--*--*--*--*--* Duane *--*--*--*--*--* -- ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Topology hiding - B2B_LOGIC
Hi Ovidiu, I will take a look on this for sure. Thx, Maciej. 2011/2/6 Ovidiu Sas o...@voipembedded.com: For now, best thing to do is to separate functionality: - one server doing topology hiding; - one server doing routing, accounting, rtp proxy, etc. Regards, Ovidiu Sas On Sun, Feb 6, 2011 at 9:23 AM, Maciej Bylica mb...@gazeta.pl 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 information of Operator_2 to Operator_1 like Contact, User-Agent headers and so on and to do rtp proxying. For rtp proxying i've installed rtpproxy and it works fine. But still the question is about signalization and SDP (o= part) I ran through a few posts and found out that the answer is B2B functionality here - so B2B_LOGIC. Are there any other wayouts or this is the only way i may follow. One more question. Should I place b2bua separately or could i combine that functionality with my current Opensips installation? I am asking because as i understand there might be some problems with proper call accounting (no radius is used in my case). If positive then my scenario will look like following: Operator_1 -- my Opensips (billing) -- Opensips b2bua (top hiding) -- OS RTP Proxy -- Operator_2. or it is wrong assumption. Thanks, Maciej. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Topology hiding - B2B_LOGIC
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've installed rtpproxy and it works fine. But still the question is about signalization and SDP (o= part) I ran through a few posts and found out that the answer is B2B functionality here - so B2B_LOGIC. Are there any other wayouts or this is the only way i may follow. One more question. Should I place b2bua separately or could i combine that functionality with my current Opensips installation? I am asking because as i understand there might be some problems with proper call accounting (no radius is used in my case). If positive then my scenario will look like following: Operator_1 -- my Opensips (billing) -- Opensips b2bua (top hiding) -- OS RTP Proxy -- Operator_2. or it is wrong assumption. Thanks, Maciej. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Topology hiding - B2B_LOGIC
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've installed rtpproxy and it works fine. But still the question is about signalization and SDP (o= part) I ran through a few posts and found out that the answer is B2B functionality here - so B2B_LOGIC. Are there any other wayouts or this is the only way i may follow. Thanks in advance, Maciej. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Opensips just stops responding
Hi Bogdan,
Thanks for prompt answer.
Port 5060 is open, but thanks to your advice i tried to originate the
INVITE directly from the server and the result was that Opensips
responded properly.
Then i found out that centos has Selinux enabled :)
Problem is solved,
Once again many thanks,
Maciej.
2010/12/22 Bogdan-Andrei Iancu bog...@voice-system.ro:
Hi Maciej,
What opensips version are you using ? 4448 is quite old rev number...Current
1.6.4 has 7611 rev number...
Anyhow, are you sure you are sending the traffic on a port which is used by
opensips ? try with netstat -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 latest revision installed $Revision: 4448 in my server.
Opensips is starting itself properly:
# ps -ef | grep opensips
root 20982 6115 0 02:01 pts/1 00:00:00 gdb
/usr/local/sbin/opensips
root 21326 1 0 02:14 ? 00:00:00
/usr/local/sbin/opensips -P /var/run/opensips/opensips.pid
root 21328 21326 0 02:14 ? 00:00:00
/usr/local/sbin/opensips -P /var/run/opensips/opensips.pid
root 21329 21326 0 02:14 ? 00:00:00
/usr/local/sbin/opensips -P /var/run/opensips/opensips.pid
root 21330 21326 0 02:14 ? 00:00:00
/usr/local/sbin/opensips -P /var/run/opensips/opensips.pid
root 21331 21326 0 02:14 ? 00:00:00
/usr/local/sbin/opensips -P /var/run/opensips/opensips.pid
root 21332 21326 0 02:14 ? 00:00:00
/usr/local/sbin/opensips -P /var/run/opensips/opensips.pid
root 21333 21326 0 02:14 ? 00:00:00
/usr/local/sbin/opensips -P /var/run/opensips/opensips.pid
root 21334 21326 0 02:14 ? 00:00:00
/usr/local/sbin/opensips -P /var/run/opensips/opensips.pid
root 21335 21326 0 02:14 ? 00:00:00
/usr/local/sbin/opensips -P /var/run/opensips/opensips.pid
root 21336 21326 0 02:14 ? 00:00:00
/usr/local/sbin/opensips -P /var/run/opensips/opensips.pid
root 21337 21326 0 02:14 ? 00:00:00
/usr/local/sbin/opensips -P /var/run/opensips/opensips.pid
root 21338 21326 0 02:14 ? 00:00:00
/usr/local/sbin/opensips -P /var/run/opensips/opensips.pid
root 21339 21326 0 02:14 ? 00:00:00
/usr/local/sbin/opensips -P /var/run/opensips/opensips.pid
root 21340 21326 0 02:14 ? 00:00:00
/usr/local/sbin/opensips -P /var/run/opensips/opensips.pid
root 21341 21326 0 02:14 ? 00:00:00
/usr/local/sbin/opensips -P /var/run/opensips/opensips.pid
root 21342 21326 0 02:14 ? 00:00:00
/usr/local/sbin/opensips -P /var/run/opensips/opensips.pid
root 21343 21326 0 02:14 ? 00:00:00
/usr/local/sbin/opensips -P /var/run/opensips/opensips.pid
root 21392 6258 0 02:18 pts/2 00:00:00 grep opensips
and there is a opensips.pid file generated.
The problem is that opensips is not responding to any request, there
are no debug information at all (default opensips.conf file)
I even created simple script route { log... } and the effect is the
same.
Here is how opensips is starting (debug 5):
Dec 22 02:25:50 mac opensips: WARNING:core:fix_socket_list: could not
rev. resolve 62.29.162.76
Dec 22 02:25:50 mac opensips: WARNING:core:fix_socket_list: could not
rev. resolve 62.29.162.76
Dec 22 02:25:50 mac opensips: INFO:core:init_tcp: using epoll_lt as
the TCP io watch method (auto detected)
Dec 22 02:25:50 mac /usr/local/sbin/opensips[21463]: NOTICE:core:main:
version: opensips 1.6.4-notls (i386/linux)
Dec 22 02:25:50 mac /usr/local/sbin/opensips[21463]: INFO:core:main:
using 32 Mb shared memory
Dec 22 02:25:50 mac /usr/local/sbin/opensips[21463]: INFO:core:main:
using 1 Mb private memory per process
Dec 22 02:25:50 mac /usr/local/sbin/opensips[21463]:
NOTICE:signaling:mod_init: initializing module ...
Dec 22 02:25:50 mac /usr/local/sbin/opensips[21463]: INFO:sl:mod_init:
Initializing StateLess engine
Dec 22 02:25:50 mac /usr/local/sbin/opensips[21463]: INFO:tm:mod_init:
TM - initializing...
Dec 22 02:25:50 mac /usr/local/sbin/opensips[21463]: INFO:rr:mod_init:
rr - initializing
Dec 22 02:25:50 mac /usr/local/sbin/opensips[21463]:
INFO:maxfwd:mod_init: initializing...
Dec 22 02:25:50 mac /usr/local/sbin/opensips[21463]:
INFO:usrloc:ul_init_locks: locks array size 512
Dec 22 02:25:50 mac /usr/local/sbin/opensips[21463]:
INFO:registrar:mod_init: initializing...
Dec 22 02:25:50 mac /usr/local/sbin/opensips[21463]:
INFO:textops:mod_init: initializing...
Dec 22 02:25:50 mac /usr/local/sbin/opensips[21463]:
INFO:acc:mod_init: initializing...
Dec 22 02:25:50 mac /usr/local/sbin/opensips[21463]:
INFO:core:probe_max_receive_buffer: using a UDP receive buffer of 255
kb
more info from gdb /usr/local/sbin/opensips
(gdb)
No stack.
and the output from opensipsctl fifo ps
Process:: ID=0 PID=21326 Type
Re: [OpenSIPS-Users] Opensips just stops responding
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 21326 1 0 02:14 ?00:00:00
/usr/local/sbin/opensips -P /var/run/opensips/opensips.pid
root 21328 21326 0 02:14 ?00:00:00
/usr/local/sbin/opensips -P /var/run/opensips/opensips.pid
root 21329 21326 0 02:14 ?00:00:00
/usr/local/sbin/opensips -P /var/run/opensips/opensips.pid
root 21330 21326 0 02:14 ?00:00:00
/usr/local/sbin/opensips -P /var/run/opensips/opensips.pid
root 21331 21326 0 02:14 ?00:00:00
/usr/local/sbin/opensips -P /var/run/opensips/opensips.pid
root 21332 21326 0 02:14 ?00:00:00
/usr/local/sbin/opensips -P /var/run/opensips/opensips.pid
root 21333 21326 0 02:14 ?00:00:00
/usr/local/sbin/opensips -P /var/run/opensips/opensips.pid
root 21334 21326 0 02:14 ?00:00:00
/usr/local/sbin/opensips -P /var/run/opensips/opensips.pid
root 21335 21326 0 02:14 ?00:00:00
/usr/local/sbin/opensips -P /var/run/opensips/opensips.pid
root 21336 21326 0 02:14 ?00:00:00
/usr/local/sbin/opensips -P /var/run/opensips/opensips.pid
root 21337 21326 0 02:14 ?00:00:00
/usr/local/sbin/opensips -P /var/run/opensips/opensips.pid
root 21338 21326 0 02:14 ?00:00:00
/usr/local/sbin/opensips -P /var/run/opensips/opensips.pid
root 21339 21326 0 02:14 ?00:00:00
/usr/local/sbin/opensips -P /var/run/opensips/opensips.pid
root 21340 21326 0 02:14 ?00:00:00
/usr/local/sbin/opensips -P /var/run/opensips/opensips.pid
root 21341 21326 0 02:14 ?00:00:00
/usr/local/sbin/opensips -P /var/run/opensips/opensips.pid
root 21342 21326 0 02:14 ?00:00:00
/usr/local/sbin/opensips -P /var/run/opensips/opensips.pid
root 21343 21326 0 02:14 ?00:00:00
/usr/local/sbin/opensips -P /var/run/opensips/opensips.pid
root 21392 6258 0 02:18 pts/200:00:00 grep opensips
and there is a opensips.pid file generated.
The problem is that opensips is not responding to any request, there
are no debug information at all (default opensips.conf file)
I even created simple script route { log... } and the effect is the same.
Here is how opensips is starting (debug 5):
Dec 22 02:25:50 mac opensips: WARNING:core:fix_socket_list: could not
rev. resolve 62.29.162.76
Dec 22 02:25:50 mac opensips: WARNING:core:fix_socket_list: could not
rev. resolve 62.29.162.76
Dec 22 02:25:50 mac opensips: INFO:core:init_tcp: using epoll_lt as
the TCP io watch method (auto detected)
Dec 22 02:25:50 mac /usr/local/sbin/opensips[21463]: NOTICE:core:main:
version: opensips 1.6.4-notls (i386/linux)
Dec 22 02:25:50 mac /usr/local/sbin/opensips[21463]: INFO:core:main:
using 32 Mb shared memory
Dec 22 02:25:50 mac /usr/local/sbin/opensips[21463]: INFO:core:main:
using 1 Mb private memory per process
Dec 22 02:25:50 mac /usr/local/sbin/opensips[21463]:
NOTICE:signaling:mod_init: initializing module ...
Dec 22 02:25:50 mac /usr/local/sbin/opensips[21463]: INFO:sl:mod_init:
Initializing StateLess engine
Dec 22 02:25:50 mac /usr/local/sbin/opensips[21463]: INFO:tm:mod_init:
TM - initializing...
Dec 22 02:25:50 mac /usr/local/sbin/opensips[21463]: INFO:rr:mod_init:
rr - initializing
Dec 22 02:25:50 mac /usr/local/sbin/opensips[21463]:
INFO:maxfwd:mod_init: initializing...
Dec 22 02:25:50 mac /usr/local/sbin/opensips[21463]:
INFO:usrloc:ul_init_locks: locks array size 512
Dec 22 02:25:50 mac /usr/local/sbin/opensips[21463]:
INFO:registrar:mod_init: initializing...
Dec 22 02:25:50 mac /usr/local/sbin/opensips[21463]:
INFO:textops:mod_init: initializing...
Dec 22 02:25:50 mac /usr/local/sbin/opensips[21463]:
INFO:acc:mod_init: initializing...
Dec 22 02:25:50 mac /usr/local/sbin/opensips[21463]:
INFO:core:probe_max_receive_buffer: using a UDP receive buffer of 255
kb
more info from gdb /usr/local/sbin/opensips
(gdb)
No stack.
and the output from opensipsctl fifo ps
Process:: ID=0 PID=21326 Type=attendant
Process:: ID=1 PID=21328 Type=MI FIFO
Process:: ID=2 PID=21329 Type=SIP receiver udp:127.0.0.1:5060
Process:: ID=3 PID=21330 Type=SIP receiver udp:127.0.0.1:5060
Process:: ID=4 PID=21331 Type=SIP receiver udp:127.0.0.1:5060
Process:: ID=5 PID=21332 Type=SIP receiver udp:127.0.0.1:5060
Process:: ID=6 PID=21333 Type=SIP receiver udp:62.29.162.76:5060
Process:: ID=7 PID=21334 Type=SIP receiver udp:62.29.162.76:5060
Process:: ID=8 PID=21335 Type=SIP receiver udp:62.29.162.76:5060
Process:: ID=9 PID=21336 Type=SIP receiver udp:62.29.162.76:5060
Process:: ID=10 PID=21337 Type=time_keeper
Process:: ID=11 PID=21338 Type=timer
Process:: ID=12 PID=21339 Type=TCP receiver
Process:: ID=13 PID=21340 Type=TCP receiver
Process:: ID=14 PID=21341 Type=TCP receiver
Process:: ID=15
Re: [OpenSIPS-Users] CANCELing the connection - no totag in ACK
Thank You Bogdan.
This is the way i am going to follow.
Maciej.
Maciej,
skip auth challenge for the ACK requests as you cannot send replies for an
ACK
So you have 2 options:
1) if you do not want to auth ACK at all, simple skip them from auth
2) 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 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 proxy auth only INVITE methods - at this moment i do have
if (!(method==REGISTER) from_uri==myself) /*no multidomain
version*/ {
if (!proxy_authorize(, subscriber)) {
xlog(L_INFO,proxy auth);
proxy_challenge(, 0);
exit;
}
so there wont be any problem to filter that out, but how to inspect
branch, CSeq - isn't that functionality hardcoded?
Thx,
Maciej.
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OpenSIPS Bootcamp
15 - 19 November 2010, Edison, New Jersey, USA
www.voice-system.ro
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Re: [OpenSIPS-Users] CANCELing the connection - no totag in ACK
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_username: Call {www,proxy}_authorize before calling
check_* functions!
After closer look it turnes out that it is generated due to lack of
totag in ACK method as a response to 487 Request Terminated.
ACK is omitting has_totag() part of configuration and then again is
asked for proxy auth.
The call is generated by UA registered with Opensips, then t_relayed
to OPERATOR_1 and his MGW to PSTN.
UA--OPENSIPS-OPERATOR_1_SIPPROXYMGW
The proper call flow should be (A) is UA, B is OPERATOR_1_SIPPROXY
1. (A)INVITE -(B)
2. (A)--180 RIGING--(B)
3. (A)CANCEL---(B)
4. (A)--OK(B)
5. (A)-487 Request Terminated---(B)
6. (A)ACK-(B)
and it looks the same, but:
- CANCEL should be sent by (A) without To tag
- OK should be sent by (B) with To tag
- 487 with the same To tag
- ACK should be sent by (A) with exactly the same To tag.
Unfortunately it is not my case :(
- I am fine with CANCEL
- I am receiving proper OK with To tag
- and here is the source of my problem. 487 is sent by (B) without
totag proposed in OK message previously sent.
- ACK is obviously using the same totag as OK, so im my case no totag
is incorporated into ACK method.
The after-effect is that ACK is asked for proxy auth.
I am asking you guys to tell me how to cope with the cases like above.
Thanks in advance,
Maciej
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Re: [OpenSIPS-Users] CANCELing the connection - no totag in ACK
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 to build transaction inspection? Thanks, Maciej. The call is generated by UA registered with Opensips, then t_relayed to OPERATOR_1 and his MGW to PSTN. UA--OPENSIPS-OPERATOR_1_SIPPROXYMGW The proper call flow should be (A) is UA, B is OPERATOR_1_SIPPROXY 1. (A)INVITE -(B) 2. (A)--180 RIGING--(B) 3. (A)CANCEL---(B) 4. (A)--OK(B) 5. (A)-487 Request Terminated---(B) 6. (A)ACK-(B) and it looks the same, but: - CANCEL should be sent by (A) without To tag - OK should be sent by (B) with To tag - 487 with the same To tag Wrong. 200 OK for CANCEL is sent by the proxy (CANCEL is hop by hop). However 487 is sent by the UAS (not by the proxy) and of course the UAS doesn't know which To tag has chosen the proxy for the 200 (CANCEL). Also, there could be multiple UAS's (parallel forking). - ACK should be sent by (A) with exactly the same To tag. Just a final 487 will be delivered by the proxy to the UAC (even in case there is parallel forking) so the ACK must contain the same Totag than the 487 received. Unfortunately it is not my case :( - I am fine with CANCEL - I am receiving proper OK with To tag - and here is the source of my problem. 487 is sent by (B) without totag proposed in OK message previously sent. And that is correct. - ACK is obviously using the same totag as OK, That is wrong. It should be the same Totag as the UAC receives in the 487. so im my case no totag is incorporated into ACK method. The after-effect is that ACK is asked for proxy auth. The proxy should not be dialog aware (Totag value aware). It should inspect just the transaction identifier (Via's branch and CSeq). -- Iñaki Baz Castillo i...@aliax.net ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] CANCELing the connection - no totag in ACK
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 proxy auth only INVITE methods - at this moment i do have
if (!(method==REGISTER) from_uri==myself) /*no multidomain
version*/ {
if (!proxy_authorize(, subscriber)) {
xlog(L_INFO,proxy auth);
proxy_challenge(, 0);
exit;
}
so there wont be any problem to filter that out, but how to inspect
branch, CSeq - isn't that functionality hardcoded?
Thx,
Maciej.
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Re: [OpenSIPS-Users] Dispatcher radius messages are not valid
Nevertheless, thank You Bogdan. Has Anybody more less similiar problem like me? Thx, Maciej. 2010/11/12 Bogdan-Andrei Iancu bog...@voice-system.ro: I see... unfortunately I cannot help you with media-dispatchernever used it :-/ Regards, Bogdan Maciej Bylica wrote: Ups sorry for not being so precise. I am talking about media-dispacher (not dispacher module) which is installed on the same server using radiusclient-ng Additionally there is media-relay running different ip. Thx, Maciej. 2010/11/11 Bogdan-Andrei Iancu bog...@voice-system.ro: lost meif dispatcher 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 same secret (the same radiusclient.conf file) and are located on the same server. There are only one entry provided in radius server clients file describing ip address (the same for opensips and dispatcher) and secret (the same for opensips and dispatcher). So if opensips had permission to sent messages then in the same way dispatcher should be able to massage radius server. Thx, Maciej Hi Maciej, Sounds quite clear (from the err message) that the secrets on radius server 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 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 same radiusclient.conf file. The problem is that radius messages generated from dispatcher are not taken into account while i have no problem with opensips radius messages (secred for dispatcher and opensips is the same) Here is an output from radius server Waking up in 0.10 seconds. Thread 9 got semaphore Thread 9 handling request 121, (13 handled so far) [thread] Received Accounting-Request packet from client 10.1.1.229 with invalid signature! (Shared secret is incorrect.) Dropping packet without response. I've already tested freeradius-xs from debian pkg with same effect. I am running 32bit os linux debian lenny. Has anybody similiar problem. Could you guys pls point me what should i check? Thx in advance, Maciej ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Bogdan-Andrei Iancu OpenSIPS Bootcamp 15 - 19 November 2010, Edison, New Jersey, USA www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Bogdan-Andrei Iancu OpenSIPS Bootcamp 15 - 19 November 2010, Edison, New Jersey, USA www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Bogdan-Andrei Iancu OpenSIPS Bootcamp 15 - 19 November 2010, Edison, New Jersey, USA www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Dispatcher radius messages are not valid
Ups sorry for not being so precise. I am talking about media-dispacher (not dispacher module) which is installed on the same server using radiusclient-ng Additionally there is media-relay running different ip. Thx, Maciej. 2010/11/11 Bogdan-Andrei Iancu bog...@voice-system.ro: lost meif dispatcher 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 same secret (the same radiusclient.conf file) and are located on the same server. There are only one entry provided in radius server clients file describing ip address (the same for opensips and dispatcher) and secret (the same for opensips and dispatcher). So if opensips had permission to sent messages then in the same way dispatcher should be able to massage radius server. Thx, Maciej Hi Maciej, Sounds quite clear (from the err message) that the secrets on radius server 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 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 same radiusclient.conf file. The problem is that radius messages generated from dispatcher are not taken into account while i have no problem with opensips radius messages (secred for dispatcher and opensips is the same) Here is an output from radius server Waking up in 0.10 seconds. Thread 9 got semaphore Thread 9 handling request 121, (13 handled so far) [thread] Received Accounting-Request packet from client 10.1.1.229 with invalid signature! (Shared secret is incorrect.) Dropping packet without response. I've already tested freeradius-xs from debian pkg with same effect. I am running 32bit os linux debian lenny. Has anybody similiar problem. Could you guys pls point me what should i check? Thx in advance, Maciej ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Bogdan-Andrei Iancu OpenSIPS Bootcamp 15 - 19 November 2010, Edison, New Jersey, USA www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Bogdan-Andrei Iancu OpenSIPS Bootcamp 15 - 19 November 2010, Edison, New Jersey, USA www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Dispatcher radius messages are not valid
Hi Bogdan, From my point of view it is not so clear, because opensips and dispatcher use the same secret (the same radiusclient.conf file) and are located on the same server. There are only one entry provided in radius server clients file describing ip address (the same for opensips and dispatcher) and secret (the same for opensips and dispatcher). So if opensips had permission to sent messages then in the same way dispatcher should be able to massage radius server. Thx, Maciej Hi Maciej, Sounds quite clear (from the err message) that the secrets on radius server 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 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 same radiusclient.conf file. The problem is that radius messages generated from dispatcher are not taken into account while i have no problem with opensips radius messages (secred for dispatcher and opensips is the same) Here is an output from radius server Waking up in 0.10 seconds. Thread 9 got semaphore Thread 9 handling request 121, (13 handled so far) [thread] Received Accounting-Request packet from client 10.1.1.229 with invalid signature! (Shared secret is incorrect.) Dropping packet without response. I've already tested freeradius-xs from debian pkg with same effect. I am running 32bit os linux debian lenny. Has anybody similiar problem. Could you guys pls point me what should i check? Thx in advance, Maciej ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Bogdan-Andrei Iancu OpenSIPS Bootcamp 15 - 19 November 2010, Edison, New Jersey, USA www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Dispatcher radius messages are not valid
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 same radiusclient.conf file. The problem is that radius messages generated from dispatcher are not taken into account while i have no problem with opensips radius messages (secred for dispatcher and opensips is the same) Here is an output from radius server Waking up in 0.10 seconds. Thread 9 got semaphore Thread 9 handling request 121, (13 handled so far) [thread] Received Accounting-Request packet from client 10.1.1.229 with invalid signature! (Shared secret is incorrect.) Dropping packet without response. I've already tested freeradius-xs from debian pkg with same effect. I am running 32bit os linux debian lenny. Has anybody similiar problem. Could you guys pls point me what should i check? Thx in advance, Maciej ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Services management - question about proper module
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 determine the service to be fetched by the script? That is the the proper module for handling generic attribute. Uisng AVPops module you can load from db, for a certain user, a certain attribute (via avp_db_load ). You may use different attributes (AVPs) for different services - like one attreibute to be URI for permanent call fwd other for being URI for busy redirect. 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. keep in mind that certain ops can be done on opensips (like call fwd), you do not need asterisk. Regards, Bogdan Is this proper point of view? Thx, Maciej. 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 that value opensips should do the magic :) The question is what kind of module is the best to follow. Is it AVPops or maybe there is another way to achieve my goal. What are pros and cons. Thx in advance, Maciej. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Bogdan-Andrei Iancu OpenSIPS Bootcamp 15 - 19 November 2010, Edison, New Jersey, USA www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] $fU read-only, calling number modification problem
Guys any advice please... Thx. Just prompt explaination: - no modparams in config no uac modparams in config. - route[0] is responsible for basic routing - route[5] is for proper call distribution by using lookup(location) information. In the same route i have implemented calling number modification. Just before the end of route i am arming t_relay with failure route (in case of busy for instance). - failure_route[105] is to do_routing the call to VM service outside the opensips. But just before t_relaying here i need to restore the original $fU. According to http://www.opensips.org/html/docs/modules/1.6.x/uac.html#id292928 i should use uac_restore_from() command (with default restore_mode modparam). Unfortunately the calling number once replaced cannot be restored in my case. Below you may find a snippet from debug Number replacing is generating vsf param /sbin/opensips[28268]: DBG:uac:w_replace_from: dsp=0xffd3f38c (len=0) , uri=0xffd3f394 (len=29) /sbin/opensips[28268]: DBG:uac:replace_uri: removing display [unknown] /sbin/opensips[28268]: DBG:uac:replace_uri: uri to replace [sip:48222114...@11.22.33.44] /sbin/opensips[28268]: DBG:uac:replace_uri: replacement uri is [sip:222114...@11.22.33.44] /sbin/opensips[28268]: DBG:uac:replace_uri: encode is=AAIJAgUMBAEAD3AHeQUXBR8FHwUaCRoKHAsyOQ-- len=44 /sbin/opensips[28268]: DBG:rr:add_rr_param: adding (;vsf=AAIJAgUMBAEAD3AHeQUXBR8FHwUaCRoKHAsyOQ--) 0x81c2828 then opensips constucts Busy message and in the same time without any uac_restore_from() command: /sbin/opensips[28795]: DBG:uac:restore_uri_reply: removing sip:222114...@11.22.33.44 /sbin/opensips[28795]: DBG:uac:restore_uri_reply: inserting unknown sip:48222114...@11.22.33.44 then the call failes to failure_route[105] and uac_restore_from() is generating following debug /sbin/opensips[28797]: DBG:uac:restore_uri: getting 'vsf' Route param /sbin/opensips[28797]: DBG:uac:restore_uri: route param 'vsf' not found just after that the call is hitting do_routing and t_relay to VM server. Of course calling number was not restored to original one. Could You please point me where the problem is located? Just one more info - calling number modification part of config is located in separated route[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 bog...@voice-system.ro: To be more precise: http://www.opensips.org/html/docs/modules/1.6.x/uac.html#id228582 Regards, Bogdan Stefano Pisani wrote: 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/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 for me, because what im struggling with is to achieve proper calling number presentation. In my scenario all endpoints located in subscriber table do have full username with country code, so there are for instance: - 48111223344 (48 country code) - 49222334455 (49 country code) - 44333445566 (44 country code) ... If there is a national call inside the 48 country code the calling number should be changed by striping first two digits (48) - 48999887766---999887766 In case of international call, i should add two digits (00) - 49222334455---0049222334455. I am using diaplan module in this case and following entry gives me the error I mentioned. dp_translate(2, $fU/$fU); If there are any workaround. Any help would be highly appreaciated. Thanks, Maciej ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Bogdan-Andrei Iancu OpenSIPS Bootcamp 15 - 19 November 2010, Edison, New Jersey, USA www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] $fU read-only, calling number modification problem
Guys one more question. I have some problems to force opensips to restore oryginal uri that was previously replaced. I do have: - no modparams in config - route[0] is responsible for basic routing - route[5] is for proper call distribution by using lookup(location) information. In the same route i have implemented calling number modification. Just before the end of route i am arming t_relay with failure route (in case of busy for instance). - failure_route[105] is to do_routing the call to VM service outside the opensips. But just before t_relaying here i need to restore the original $fU. According to http://www.opensips.org/html/docs/modules/1.6.x/uac.html#id292928 i should use uac_restore_from() command (with default restore_mode modparam). Unfortunately the calling number once replaced cannot be restored in my case. Below you may find a snippet from debug Number replacing is generating vsf param /sbin/opensips[28268]: DBG:uac:w_replace_from: dsp=0xffd3f38c (len=0) , uri=0xffd3f394 (len=29) /sbin/opensips[28268]: DBG:uac:replace_uri: removing display [unknown] /sbin/opensips[28268]: DBG:uac:replace_uri: uri to replace [sip:48222114...@11.22.33.44] /sbin/opensips[28268]: DBG:uac:replace_uri: replacement uri is [sip:222114...@11.22.33.44] /sbin/opensips[28268]: DBG:uac:replace_uri: encode is=AAIJAgUMBAEAD3AHeQUXBR8FHwUaCRoKHAsyOQ-- len=44 /sbin/opensips[28268]: DBG:rr:add_rr_param: adding (;vsf=AAIJAgUMBAEAD3AHeQUXBR8FHwUaCRoKHAsyOQ--) 0x81c2828 then opensips constucts Busy message and in the same time without any uac_restore_from() command: /sbin/opensips[28795]: DBG:uac:restore_uri_reply: removing sip:222114...@11.22.33.44 /sbin/opensips[28795]: DBG:uac:restore_uri_reply: inserting unknown sip:48222114...@11.22.33.44 then the call failes to failure_route[105] and uac_restore_from() is generating following debug /sbin/opensips[28797]: DBG:uac:restore_uri: getting 'vsf' Route param /sbin/opensips[28797]: DBG:uac:restore_uri: route param 'vsf' not found just after that the call is hitting do_routing and t_relay to VM server. Of course calling number was not restored to original one. Could You please point me where the problem is located? Just one more info - calling number modification part of config is located in separated route[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 bog...@voice-system.ro: To be more precise: http://www.opensips.org/html/docs/modules/1.6.x/uac.html#id228582 Regards, Bogdan Stefano Pisani wrote: 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/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 for me, because what im struggling with is to achieve proper calling number presentation. In my scenario all endpoints located in subscriber table do have full username with country code, so there are for instance: - 48111223344 (48 country code) - 49222334455 (49 country code) - 44333445566 (44 country code) ... If there is a national call inside the 48 country code the calling number should be changed by striping first two digits (48) - 48999887766---999887766 In case of international call, i should add two digits (00) - 49222334455---0049222334455. I am using diaplan module in this case and following entry gives me the error I mentioned. dp_translate(2, $fU/$fU); If there are any workaround. Any help would be highly appreaciated. Thanks, Maciej ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Bogdan-Andrei Iancu OpenSIPS Bootcamp 15 - 19 November 2010, Edison, New Jersey, USA www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] $fU read-only, calling number modification problem
Just prompt explaination: - no modparams in config no uac modparams in config. - route[0] is responsible for basic routing - route[5] is for proper call distribution by using lookup(location) information. In the same route i have implemented calling number modification. Just before the end of route i am arming t_relay with failure route (in case of busy for instance). - failure_route[105] is to do_routing the call to VM service outside the opensips. But just before t_relaying here i need to restore the original $fU. According to http://www.opensips.org/html/docs/modules/1.6.x/uac.html#id292928 i should use uac_restore_from() command (with default restore_mode modparam). Unfortunately the calling number once replaced cannot be restored in my case. Below you may find a snippet from debug Number replacing is generating vsf param /sbin/opensips[28268]: DBG:uac:w_replace_from: dsp=0xffd3f38c (len=0) , uri=0xffd3f394 (len=29) /sbin/opensips[28268]: DBG:uac:replace_uri: removing display [unknown] /sbin/opensips[28268]: DBG:uac:replace_uri: uri to replace [sip:48222114...@11.22.33.44] /sbin/opensips[28268]: DBG:uac:replace_uri: replacement uri is [sip:222114...@11.22.33.44] /sbin/opensips[28268]: DBG:uac:replace_uri: encode is=AAIJAgUMBAEAD3AHeQUXBR8FHwUaCRoKHAsyOQ-- len=44 /sbin/opensips[28268]: DBG:rr:add_rr_param: adding (;vsf=AAIJAgUMBAEAD3AHeQUXBR8FHwUaCRoKHAsyOQ--) 0x81c2828 then opensips constucts Busy message and in the same time without any uac_restore_from() command: /sbin/opensips[28795]: DBG:uac:restore_uri_reply: removing sip:222114...@11.22.33.44 /sbin/opensips[28795]: DBG:uac:restore_uri_reply: inserting unknown sip:48222114...@11.22.33.44 then the call failes to failure_route[105] and uac_restore_from() is generating following debug /sbin/opensips[28797]: DBG:uac:restore_uri: getting 'vsf' Route param /sbin/opensips[28797]: DBG:uac:restore_uri: route param 'vsf' not found just after that the call is hitting do_routing and t_relay to VM server. Of course calling number was not restored to original one. Could You please point me where the problem is located? Just one more info - calling number modification part of config is located in separated route[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 bog...@voice-system.ro: To be more precise: http://www.opensips.org/html/docs/modules/1.6.x/uac.html#id228582 Regards, Bogdan Stefano Pisani wrote: 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/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 for me, because what im struggling with is to achieve proper calling number presentation. In my scenario all endpoints located in subscriber table do have full username with country code, so there are for instance: - 48111223344 (48 country code) - 49222334455 (49 country code) - 44333445566 (44 country code) ... If there is a national call inside the 48 country code the calling number should be changed by striping first two digits (48) - 48999887766---999887766 In case of international call, i should add two digits (00) - 49222334455---0049222334455. I am using diaplan module in this case and following entry gives me the error I mentioned. dp_translate(2, $fU/$fU); If there are any workaround. Any help would be highly appreaciated. Thanks, Maciej ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Bogdan-Andrei Iancu OpenSIPS Bootcamp 15 - 19 November 2010, Edison, New Jersey, USA www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] Services management - question about proper module
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. 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 that value opensips should do the magic :) The question is what kind of module is the best to follow. Is it AVPops or maybe there is another way to achieve my goal. What are pros and cons. Thx in advance, Maciej. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] Services management - question about proper module
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 that value opensips should do the magic :) The question is what kind of module is the best to follow. Is it AVPops or maybe there is another way to achieve my goal. What are pros and cons. Thx in advance, Maciej. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] $fU read-only, calling number modification problem
Bogdan, Stefano, Its working as is should :) Thanks for pointing me to the right function. Maciej. 2010/10/11 Bogdan-Andrei Iancu bog...@voice-system.ro: To be more precise: http://www.opensips.org/html/docs/modules/1.6.x/uac.html#id228582 Regards, Bogdan Stefano Pisani wrote: 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/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 for me, because what im struggling with is to achieve proper calling number presentation. In my scenario all endpoints located in subscriber table do have full username with country code, so there are for instance: - 48111223344 (48 country code) - 49222334455 (49 country code) - 44333445566 (44 country code) ... If there is a national call inside the 48 country code the calling number should be changed by striping first two digits (48) - 48999887766---999887766 In case of international call, i should add two digits (00) - 49222334455---0049222334455. I am using diaplan module in this case and following entry gives me the error I mentioned. dp_translate(2, $fU/$fU); If there are any workaround. Any help would be highly appreaciated. Thanks, Maciej ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Bogdan-Andrei Iancu OpenSIPS Bootcamp 15 - 19 November 2010, Edison, New Jersey, USA www.voice-system.ro ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] $fU read-only, calling number modification problem
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 for me, because what im struggling with is to achieve proper calling number presentation. In my scenario all endpoints located in subscriber table do have full username with country code, so there are for instance: - 48111223344 (48 country code) - 49222334455 (49 country code) - 44333445566 (44 country code) ... If there is a national call inside the 48 country code the calling number should be changed by striping first two digits (48) - 48999887766---999887766 In case of international call, i should add two digits (00) - 49222334455---0049222334455. I am using diaplan module in this case and following entry gives me the error I mentioned. dp_translate(2, $fU/$fU); If there are any workaround. Any help would be highly appreaciated. Thanks, Maciej ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] DRouting - routeid, prefix questions
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. Thanks for the rest info. Regards, Maciej ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] DRouting - routeid, prefix questions
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. ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
[OpenSIPS-Users] One-way audio problem with Opensips and Mediaproxy
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 {
# -
# Sanity Check Section
# -
if (!mf_process_maxfwd_header(10)) {
sl_send_reply(483, Too Many Hops);
exit;
};
if (msg:len max_len) {
sl_send_reply(513, Message Overflow);
exit;
};
# -
# Record Route Section
# -
if (method==INVITE nat_uac_test(3)) {
record_route_preset(xx.yy.zz.vv:5060;nat=yes);
} else if (method!=REGISTER) {
record_route();
};
# -
# Call Tear Down Section
# -
if (method==BYE || method==CANCEL) {
end_media_session();
};
# -
# Loose Route Section
# -
if (loose_route()) {
if ((method==INVITE || method==REFER) !has_totag()) {
sl_send_reply(403, Forbidden);
exit;
};
if (method==INVITE) {
if (!proxy_authorize(,subscriber)) {
proxy_challenge(,0);
exit;
} else if (!db_check_from()) {
sl_send_reply(403, Use From=ID);
exit;
};
consume_credentials();
if (nat_uac_test(3) || search(^Route:.*;nat=yes)) {
setflag(6);
use_media_proxy();
};
};
route(1);
exit;
};
# -
# Call Type Processing Section
# -
if (uri!=myself) {
route(4);
route(1);
exit;
};
if (method==ACK) {
route(1);
exit;
} else if (method==CANCEL) {
route(1);
exit;
} else if (method==INVITE) {
route(3);
exit;
} else if (method==REGISTER) {
route(2);
exit;
};
lookup(aliases);
if (uri!=myself) {
route(4);
route(1);
exit;
};
if (!lookup(location)) {
sl_send_reply(404, User Not Found);
exit;
};
route(1);
}
route[1] {
# -
# Default Message Handler
# -
t_on_reply(1);
if (!t_relay()) {
if (method==INVITE || method==ACK) {
end_media_session();
};
sl_reply_error();
};
}
route[2] {
# -
# REGISTER Message Handler
#
sl_send_reply(100, Trying);
if (!search(^Contact:[ ]*\*) nat_uac_test(31)) {
setflag(6);
fix_nated_register();
force_rport();
};
if (!www_authorize(,subscriber)) {
www_challenge(,0);
exit;
};
if (!db_check_to()) {
sl_send_reply(401, Unauthorized);
exit;
};
consume_credentials();
if (!save(location)) {
sl_reply_error();
};
}
route[3] {
# -
# INVITE Message Handler
# -
if (nat_uac_test(3)) {
setflag(7);
force_rport();
fix_nated_contact();
};
if (!proxy_authorize(,subscriber)) {
proxy_challenge(,0);
exit;
} else if (!db_check_from()) {
sl_send_reply(403, Use From=ID);
exit;
};
consume_credentials();
lookup(aliases);
if (uri!=myself) {
route(4);
route(1);
exit;
};
if (!lookup(location)) {
sl_send_reply(404, User Not Found);
exit;
};
route(4);
route(1);
}
route[4] {
# -
# NAT Traversal Section
# -
if (isflagset(6) || isflagset(7)) {
if (!isflagset(8)) {
setflag(8);
use_media_proxy();
};
};
}
onreply_route[1] {
if ((isflagset(6) || isflagset(7)) (status=~(180)|(183)|2[0-9][0-9])) {
if (!search(^Content-Length:[ ]*0)) {
$avp(s:media_relay) = xx.yy.zz.vv;
use_media_proxy();
};
};
if (nat_uac_test(1)) {
fix_nated_contact();
};
}
Here is my call flow:
UA1(behindNAT)---Opensips,mediaproxy--Asterisk(publicip)UA2(behind
NAT)
If the call is originated from UA1 side, there is a both-ways audio.
The problem occurs in opposite scenario, if UA2 is calling UA1.
In db there are following entries:
| 101 | UA1number | NULL |
