[SR-Users] Kamailio delayed reponse in case of database load
Hello,
I have Kamailio installed as SIP redirect for an SBC to make routing decisions.
I'm using SQLOps module with postgresql 11.5 where I have multiple tables
containing each around 6M record of routing codes:
Code options
392342 sup1|sup2|sup3
Where code field is of prefix_range data type and has a gist index.
I'm sending traffic of 300 CPS to SBC which is forwarding it to Kamailio, and
Kamailio respond with 300 multiple choice with the routing decision.
In normal cases, everything is going smoothly where Kamailio repsonds up to 50
ms.
But during the process of loading data to the routing tables, Kamailio response
to SBC is dramatically delayed , where for some calls it takes up to 7 seconds
to respond.
Noting that during this process the data are loaded to staging tables and not
to live tables used by Kamailio.
Kamailio is installed on a VM with 32GB RAM and 16 vcores.
Following configuration is used in Kamailio:
fork=yes
children=10
following are the customized parameters in postgresql.conf
max_connections = 100
shared_buffers = 8GB
effective_cache_size = 24GB
maintenance_work_mem = 2GB
checkpoint_completion_target = 0.7
wal_buffers = 16MB
default_statistics_target = 100
random_page_cost = 1.1
effective_io_concurrency = 200
work_mem = 5242kB
min_wal_size = 1GB
max_wal_size = 2GB
max_worker_processes = 16
max_parallel_workers_per_gather = 8
max_parallel_workers = 16
Following is the routing logic I'm using:
route{
if (is_method("INVITE")) {
$var(inc) = $(ct{param.value,tgrp});
xlog("L_INFO","CONTACT : $var(inc)" );
xlog("L_INFO","from number : $rU" );
if($var(inc)!=0)
route(customer);
route(block);
}
}
route[customer]
{
sql_query("cb", "select caid from customeridentification where
trunk='$var(inc)'", "ra");
$var(cid)=$dbr(ra=>[0,0]);
sql_result_free("ra");
xlog("L_INFO","cid: $var(cid) \n");
sql_query("cb", "select routetablename from carrieraccount where
caid=$var(cid)", "ra");
$var(tid)=$dbr(ra=>[0,0]);
sql_result_free("ra");
if($var(cid)!=0)
route(query);
else
route(block);
}
route[query]
{
#sql_query("cb", "select supplierid from route_$var(cid) where prefix @>'$rU'
order by prefix limit 1", "ra");
sql_query("cb", "select get_routing('$var(tid)','$rU')", "ra");
$var(sup)=$dbr(ra=>[0,0]);
sql_result_free("ra");
xlog("L_INFO","supplieris: $var(sup) \n");
#$var(x)=$(var(sup){s.replace,|,}{s.len})/4;
$var(a)=$(var(sup){s.len});
$var(z)=$(var(sup){s.replace,|,});
$var(b)=$(var(z){s.len});
$var(suplen)=$var(a) - $var(b);
if($var(sup)!=0)
route(send);
else
route(block);
}
route[send]
{
xlog("L_INFO","Inside Send \n");
$var(i)=0;
append_to_reply("Contact:");
while($var(i)<$var(suplen) + 1)
{
$var(s)=$(var(sup){s.select,$var(i),|});
$var(s)=$(var(s){s.select,0,;});
append_to_reply("");
if($var(i)<$var(suplen)) append_to_reply(",");
$var(i)=$var(i)+1;
}
append_to_reply("\r\n");
sl_send_reply("300","Multiple Choices");
exit;
}
route[block]
{
xlog("L_INFO","INSIDE BLOCK" );
append_to_reply("Contact:\r\n");
sl_send_reply("300","Multiple Choices");
exit;
}
How can we enhance the Kamailio response in case of load on the postgresql
server.
Regards,
Ali Taher
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[SR-Users] Kamailio delayed reponse in case of database load
Hello, I have Kamailio installed as SIP redirect for an SBC to make routing decisions. I'm using SQLOps module with postgresql 11.5 where I have multiple tables containing each around 6M record of routing codes: Code options --- 392342 sup1|sup2|sup3 Where code field is of prefix_range data type and has a gist index. I'm sending traffic of 300 CPS to SBC which is forwarding it to Kamailio, and Kamailio respond with 300 multiple choice with the routing decision. In normal cases, everything is going smoothly where Kamailio repsonds up to 50 ms. But during the process of loading data to the routing tables, Kamailio response to SBC is dramatically delayed , where for some calls it takes up to 7 seconds to respond. Noting that during this process the data are loaded to staging tables and not to live tables used by Kamailio. Kamailio is installed on a VM with 32GB RAM and 16 vcores. Following configuration is used in Kamailio: fork=yes children=10 following are the customized parameters in postgresql.conf max_connections = 100 shared_buffers = 8GB effective_cache_size = 24GB maintenance_work_mem = 2GB checkpoint_completion_target = 0.7 wal_buffers = 16MB default_statistics_target = 100 random_page_cost = 1.1 effective_io_concurrency = 200 work_mem = 5242kB min_wal_size = 1GB max_wal_size = 2GB max_worker_processes = 16 max_parallel_workers_per_gather = 8 max_parallel_workers = 16 ___ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Kamailio delayed reponse in case of database load
Hello, It normally takes few milliseconds to execute the query. The issue is occurring only at specific times where another application is inserting data into routing tables (not the live tables used by Kamailio, which will be swapped later to the live tables) , where 300 CPS is flowing into kamailio. Regards, Ali Taher From: sr-users On Behalf Of David Villasmil Sent: Tuesday, December 10, 2019 5:04 PM To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Kamailio delayed reponse in case of database load Hello, Have you checked yourself how long PGSQL takes to answer a query from a different host? This really doesn't seem Kamalio-related, but PGSQL-related. Regards, David Villasmil email: david.villasmil.w...@gmail.com<mailto:david.villasmil.w...@gmail.com> phone: +34669448337 On Tue, Dec 10, 2019 at 12:41 PM Ali Taher mailto:ata...@vanrise.com>> wrote: Hello, I have Kamailio installed as SIP redirect for an SBC to make routing decisions. I’m using SQLOps module with postgresql 11.5 where I have multiple tables containing each around 6M record of routing codes: Code options --- 392342 sup1|sup2|sup3 Where code field is of prefix_range data type and has a gist index. I’m sending traffic of 300 CPS to SBC which is forwarding it to Kamailio, and Kamailio respond with 300 multiple choice with the routing decision. In normal cases, everything is going smoothly where Kamailio repsonds up to 50 ms. But during the process of loading data to the routing tables, Kamailio response to SBC is dramatically delayed , where for some calls it takes up to 7 seconds to respond. Noting that during this process the data are loaded to staging tables and not to live tables used by Kamailio. Kamailio is installed on a VM with 32GB RAM and 16 vcores. Following configuration is used in Kamailio: fork=yes children=10 following are the customized parameters in postgresql.conf max_connections = 100 shared_buffers = 8GB effective_cache_size = 24GB maintenance_work_mem = 2GB checkpoint_completion_target = 0.7 wal_buffers = 16MB default_statistics_target = 100 random_page_cost = 1.1 effective_io_concurrency = 200 work_mem = 5242kB min_wal_size = 1GB max_wal_size = 2GB max_worker_processes = 16 max_parallel_workers_per_gather = 8 max_parallel_workers = 16 ___ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org<mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users ___ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Kamailio delayed reponse in case of database load
Thanks Alex for the detailed explanation, I will check the provided link. Meanwhile, i would like to add that the application loading data into the postges tables uses 5 parallel processes to do so. I’m worrying that the delay in response could be due to insufficient workers available for kamailio to process more calls. For that I increased max_parallel_workers to 32, and kamailio children to 22. I noticed a positive enhancement after this change , but still in some cases I have delayed response up to 5 seconds. Here is a statistics of calls response during the table loading: Response time Number of calls 5 secs 2 ~1 sec157 >500ms 231 >100ms 427 <100ms 110k I also noticed that the delayed responses occurs mostly at a specific second,while during the whole process of table loading , response is resonnable. So I’m wondering what could be happening during this specific second to make this delay in response, knowing that the data loading rythm is same during the whole process. Regards, Ali Sent from my iPhone On Dec 10, 2019, at 5:53 PM, Alex Balashov mailto:abalas...@evaristesys.com>> wrote: Ali, If Kamailio is performing a database-bound workload, there's no way to make it respond faster if the database is being slow. Excluding things like caching of route responses, how would that be logically possible? :-) Since we use PostgreSQL (and prefix_range) very extensively, I can say that such extreme slow-downs when loading data are abnormal; relational databases in general, and Postgres's MVCC in particular, are specifically designed to deal with concurrently servicing intensive read operations amidst bulk writes. Speculating purely a priori, these delays are probably caused by very high I/O demand of a slow storage subsystem; you can attempt to ascertain that by running 'iostat -x 1' while loading the new data and looking at percentage utilisation on the relevant storage interface, or with 'iotop' or similar tools. But if the database responds slowly due to high background I/O load, you can't make Kamailio render an answer any faster. About the only thing you can do is to try mitigate the negative effects of this on the SIP level: (1) Send a '100 Trying' pre-emptively before initiating the query; this will dampen the retransmissions that otherwise occur, and whose proliferation could cause a positive feedback loop and deepen your problems in a database slow-down scenario; (2) Do asynchronous processing of the SQL query with t_suspend() / t_continue() -- though, you should carefully weigh the downsides of this given the (small) additional overhead of suspending/resuming every transaction under normal operating conditions. These approaches will prevent an overall request processing stall, but certainly not lower response time. For more background on this topic, consider a look at my blog post on the subject from some years ago: http://www.evaristesys.com/blog/tuning-kamailio-for-high-throughput-and-performance/ Hope that helps! Cheers, -- Alex PS. You may wish to escape the value of '$rU' and not lift it directly into your SQL queries, e.g. https://www.kamailio.org/wiki/cookbooks/5.3.x/transformations#sescapecommon https://www.kamailio.org/wiki/cookbooks/5.3.x/transformations#sqlval Otherwise, you may be exposing yourself to a possible SQL injection attack, i.e. if I get creative with what I put in the user part of the request URI in a way that doesn't jam Kamailio's SIP parser. — Sent from mobile, with due apologies for brevity and errors. On Dec 10, 2019, at 7:04 AM, David Villasmil mailto:david.villasmil.w...@gmail.com>> wrote: Hello, Have you checked yourself how long PGSQL takes to answer a query from a different host? This really doesn't seem Kamalio-related, but PGSQL-related. Regards, David Villasmil email: david.villasmil.w...@gmail.com<mailto:david.villasmil.w...@gmail.com> phone: +34669448337 On Tue, Dec 10, 2019 at 12:41 PM Ali Taher mailto:ata...@vanrise.com>> wrote: Hello, I have Kamailio installed as SIP redirect for an SBC to make routing decisions. I’m using SQLOps module with postgresql 11.5 where I have multiple tables containing each around 6M record of routing codes: Code options --- 392342 sup1|sup2|sup3 Where code field is of prefix_range data type and has a gist index. I’m sending traffic of 300 CPS to SBC which is forwarding it to Kamailio, and Kamailio respond with 300 multiple choice with the routing decision. In normal cases, everything is going smoothly where Kamailio repsonds up to 50 ms. But during the process of loading data to the routing tables, Kamailio response to SBC is dramatically delayed , where for some calls it takes up to 7 seconds to
Re: [SR-Users] Kamailio delayed reponse in case of database load
Thanks Alex for drawing my attention regarding escaping $rU before using it in
sql queries. But I'm not sure where to use sql.val in this case ? Do you mean I
can use either {s.escape.common} or {sql.val} ?
Regarding the 100 trying , should I put modparam("tm", "auto_inv_100_reason",
"Trying") in the beginning of if (is_method("INVITE")) block ?
Regards,
Ali Taher
-Original Message-
From: sr-users On Behalf Of Alex Balashov
Sent: Tuesday, December 10, 2019 11:21 AM
To: Kamailio (SER) - Users Mailing List
Subject: Re: [SR-Users] Kamailio delayed reponse in case of database load
Ali,
If Kamailio is performing a database-bound workload, there's no way to make it
respond faster if the database is being slow. Excluding things like caching of
route responses, how would that be logically possible?
:-)
Since we use PostgreSQL (and prefix_range) very extensively, I can say that
such extreme slow-downs when loading data are abnormal; relational databases in
general, and Postgres's MVCC in particular, are specifically designed to deal
with concurrently servicing intensive read operations amidst bulk writes.
Speculating purely a priori, these delays are probably caused by very high I/O
demand of a slow storage subsystem; you can attempt to ascertain that by
running 'iostat -x 1' while loading the new data and looking at percentage
utilisation on the relevant storage interface, or with 'iotop' or similar tools.
But if the database responds slowly due to high background I/O load, you can't
make Kamailio render an answer any faster. About the only thing you can do is
to try mitigate the negative effects of this on the SIP level:
(1) Send a '100 Trying' pre-emptively before initiating the query; this will
dampen the retransmissions that otherwise occur, and whose proliferation could
cause a positive feedback loop and deepen your problems in a database slow-down
scenario;
(2) Do asynchronous processing of the SQL query with t_suspend() /
t_continue() -- though, you should carefully weigh the downsides of this given
the (small) additional overhead of suspending/resuming every transaction under
normal operating conditions.
For more background on this topic, consider a look at my blog post on the
subject from some years ago:
http://www.evaristesys.com/blog/tuning-kamailio-for-high-throughput-and-performance/
Hope that helps!
Cheers,
-- Alex
PS. You may wish to escape the value of '$rU' and not lift it directly into
your SQL queries, e.g.
https://www.kamailio.org/wiki/cookbooks/5.3.x/transformations#sescapecommon
https://www.kamailio.org/wiki/cookbooks/5.3.x/transformations#sqlval
Otherwise, you may be exposing yourself to a possible SQL injection attack,
i.e. if I get creative with what I put in the user part of the request URI in a
way that doesn't jam Kamailio's SIP parser.
--
Alex Balashov | Principal | Evariste Systems LLC
Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
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Re: [SR-Users] Kamailio delayed reponse in case of database load
Sorry I meant putting modparam("tm", "auto_inv_100_reason", "Trying") as
parameter.
As per tm module , auto_inv_100 parameter is by default set as 1 , but still
Kamailio is not sending it, is there some function that trigger it ?
-Original Message-
From: sr-users On Behalf Of Ali Taher
Sent: Thursday, December 12, 2019 4:39 PM
To: Kamailio (SER) - Users Mailing List
Subject: Re: [SR-Users] Kamailio delayed reponse in case of database load
Thanks Alex for drawing my attention regarding escaping $rU before using it in
sql queries. But I'm not sure where to use sql.val in this case ? Do you mean I
can use either {s.escape.common} or {sql.val} ?
Regarding the 100 trying , should I put modparam("tm", "auto_inv_100_reason",
"Trying") in the beginning of if (is_method("INVITE")) block ?
Regards,
Ali Taher
-Original Message-
From: sr-users On Behalf Of Alex Balashov
Sent: Tuesday, December 10, 2019 11:21 AM
To: Kamailio (SER) - Users Mailing List
Subject: Re: [SR-Users] Kamailio delayed reponse in case of database load
Ali,
If Kamailio is performing a database-bound workload, there's no way to make it
respond faster if the database is being slow. Excluding things like caching of
route responses, how would that be logically possible?
:-)
Since we use PostgreSQL (and prefix_range) very extensively, I can say that
such extreme slow-downs when loading data are abnormal; relational databases in
general, and Postgres's MVCC in particular, are specifically designed to deal
with concurrently servicing intensive read operations amidst bulk writes.
Speculating purely a priori, these delays are probably caused by very high I/O
demand of a slow storage subsystem; you can attempt to ascertain that by
running 'iostat -x 1' while loading the new data and looking at percentage
utilisation on the relevant storage interface, or with 'iotop' or similar tools.
But if the database responds slowly due to high background I/O load, you can't
make Kamailio render an answer any faster. About the only thing you can do is
to try mitigate the negative effects of this on the SIP level:
(1) Send a '100 Trying' pre-emptively before initiating the query; this will
dampen the retransmissions that otherwise occur, and whose proliferation could
cause a positive feedback loop and deepen your problems in a database slow-down
scenario;
(2) Do asynchronous processing of the SQL query with t_suspend() /
t_continue() -- though, you should carefully weigh the downsides of this given
the (small) additional overhead of suspending/resuming every transaction under
normal operating conditions.
For more background on this topic, consider a look at my blog post on the
subject from some years ago:
http://www.evaristesys.com/blog/tuning-kamailio-for-high-throughput-and-performance/
Hope that helps!
Cheers,
-- Alex
PS. You may wish to escape the value of '$rU' and not lift it directly into
your SQL queries, e.g.
https://www.kamailio.org/wiki/cookbooks/5.3.x/transformations#sescapecommon
https://www.kamailio.org/wiki/cookbooks/5.3.x/transformations#sqlval
Otherwise, you may be exposing yourself to a possible SQL injection attack,
i.e. if I get creative with what I put in the user part of the request URI in a
way that doesn't jam Kamailio's SIP parser.
--
Alex Balashov | Principal | Evariste Systems LLC
Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
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[SR-Users] get first occurrence value of a param in a string
Hello,
I need to get the value of first tgrp param in this string :
$var(x) =
I'm trying $var(y) = $(var(x){param.value,tgrp}); but it's getting the last
value of tgrp which is "237>".
Noting that first tgrp is not always in the second index , some other
parameters can be added to the string.
How to get the value of first occurrence of tgrp param ?
Regards,
Ali Taher
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[SR-Users] SIP Redirect CANCEL handling
Hello, I'm using Kamailio as sip redirect server where SBC forwards the calls to Kamailio which sends the routing back to SBC in 3xx response. Now, I need to handle CANCEL requests sent from SBC to Kamailio. Here is the scenario: the client sends CANCEL request to SBC which responds with 200 OK and 487 Request Terminated and then sends CANCEL request to Kamailio. I'm not sure how Kamailio should handle the CANCEL request here, should it send only 200 OK back to SBC or also should send 487 request terminated? Regards, Ali Taher ___ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] SIP Redirect CANCEL handling
Hello Alex,
Kamailio sent 300 multiple choice back to SBC even after the SBC sent the
CANCEL request , knowing that SBC sent the CANCEL 4 times as Kamailio didn’t
respond to it , which is normal as in the routing logic I’m only handling
INVITE packets (if (is_method("INVITE"))).
So I’m confused whether I have to handle it or not.
Regards,
Ali Taher
From: sr-users On Behalf Of Alex Balashov
Sent: Tuesday, January 28, 2020 3:37 PM
To: Kamailio (SER) - Users Mailing List
Subject: Re: [SR-Users] SIP Redirect CANCEL handling
Unless you have created a transaction in the course of processing the redirect,
there is no transaction to terminate, and accordingly no 487 to generate.
However, these decisions are made by the default CANCEL handling (with hooks
into TM) from the stock Kamailio config. You should not need any special logic
of your own here.
—
Sent from my iPad
On Jan 28, 2020, at 8:29 AM, Ali Taher
mailto:ata...@vanrise.com>> wrote:
Hello,
I’m using Kamailio as sip redirect server where SBC forwards the calls to
Kamailio which sends the routing back to SBC in 3xx response.
Now, I need to handle CANCEL requests sent from SBC to Kamailio. Here is the
scenario:
the client sends CANCEL request to SBC which responds with 200 OK and 487
Request Terminated and then sends CANCEL request to Kamailio.
I’m not sure how Kamailio should handle the CANCEL request here, should it send
only 200 OK back to SBC or also should send 487 request terminated?
Regards,
Ali Taher
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Re: [SR-Users] SIP Redirect CANCEL handling
Hello Alex,
You mean handle the cancel from Kamailio side ? can you give me hint how to do
it ?
Regards,
Ali Taher
From: sr-users On Behalf Of Alex Balashov
Sent: Tuesday, January 28, 2020 4:07 PM
To: Kamailio (SER) - Users Mailing List
Subject: Re: [SR-Users] SIP Redirect CANCEL handling
You should handle the cancel.
—
Sent from mobile, with due apologies for brevity and errors.
On Jan 28, 2020, at 8:45 AM, Ali Taher
mailto:ata...@vanrise.com>> wrote:
Hello Alex,
Kamailio sent 300 multiple choice back to SBC even after the SBC sent the
CANCEL request , knowing that SBC sent the CANCEL 4 times as Kamailio didn’t
respond to it , which is normal as in the routing logic I’m only handling
INVITE packets (if (is_method("INVITE"))).
So I’m confused whether I have to handle it or not.
Regards,
Ali Taher
From: sr-users
mailto:sr-users-boun...@lists.kamailio.org>>
On Behalf Of Alex Balashov
Sent: Tuesday, January 28, 2020 3:37 PM
To: Kamailio (SER) - Users Mailing List
mailto:sr-users@lists.kamailio.org>>
Subject: Re: [SR-Users] SIP Redirect CANCEL handling
Unless you have created a transaction in the course of processing the redirect,
there is no transaction to terminate, and accordingly no 487 to generate.
However, these decisions are made by the default CANCEL handling (with hooks
into TM) from the stock Kamailio config. You should not need any special logic
of your own here.
—
Sent from my iPad
On Jan 28, 2020, at 8:29 AM, Ali Taher
mailto:ata...@vanrise.com>> wrote:
Hello,
I’m using Kamailio as sip redirect server where SBC forwards the calls to
Kamailio which sends the routing back to SBC in 3xx response.
Now, I need to handle CANCEL requests sent from SBC to Kamailio. Here is the
scenario:
the client sends CANCEL request to SBC which responds with 200 OK and 487
Request Terminated and then sends CANCEL request to Kamailio.
I’m not sure how Kamailio should handle the CANCEL request here, should it send
only 200 OK back to SBC or also should send 487 request terminated?
Regards,
Ali Taher
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[SR-Users] sqlops multiple databases
Hello,
Is there a way in sqlops module to use multiple postgres databases based on
routing logic ?
Here is what I'm using right now:
modparam("sqlops","sqlcon","cb=>postgres://postgres:postgres@localhost:5432/db1")
Thanks
Ali Taher
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[SR-Users] change module parameter in routing logic
Hi, Is it possible to change a module parameter inside the routing logic ? For example I want to change sqlcon parameter of sqlops module based on the originating IP. Regards, ___ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] change module parameter in routing logic
Hi Alex, I just realized that I can set sqlcon multiple times which will solve my problem. The scenario I’m trying to achieve is that based on the originating IP of the invite packet I will run some queries on the correspondent database and then use the result in Kamailio routing logic. Should I be worried about using same sqlres (ra) while fetching the result from the database? Regards, Ali Taher From: sr-users On Behalf Of Alex Balashov Sent: Tuesday, April 28, 2020 1:30 PM To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] change module parameter in routing logic It is not. — Sent from mobile, with due apologies for brevity and errors. On Apr 28, 2020, at 6:26 AM, Ali Taher mailto:ata...@vanrise.com>> wrote: Hi, Is it possible to change a module parameter inside the routing logic ? For example I want to change sqlcon parameter of sqlops module based on the originating IP. Regards, ___ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org<mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users ___ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] change module parameter in routing logic
Hi Alex, What do you mean by multiple SQL results concurrently? I will have multiple calls coming at the same time which need to go to one of the databases based on the originating IP. -Original Message- From: sr-users On Behalf Of Alex Balashov Sent: Tuesday, April 28, 2020 4:01 PM To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] change module parameter in routing logic On Tue, Apr 28, 2020 at 12:57:16PM +, Ali Taher wrote: > Should I be worried about using same sqlres (ra) while fetching the > result from the database? No, as long you sql_result_free() it between operations. If you need multiple SQL results concurrently, you should use different result set identifiers. -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/ ___ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users ___ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] worker processes
Hi, I just want to ask regarding the children number and listening interfaces set in Kamailio configuration. I am given to understand that the total number of worker processes there will be (children_setting * listeners) , so assuming I have set children=12 and listen=udp:172.16.1.12:5060 (which is the interface IP where the SIP packets as received), in this case there will be 12 worker processes and 12 database connections to postgresql (I'm using SQLOPS). My question is if I set listen=udp:0.0.0.0:5060 , knowing that I have 3 interfaces, in this case the number of worker processes will be 36. Is this good configuration ? will the other 24 processes be used or they are reserved to the 2 other interfaces which are not receiving any SIP packets and then will be useless ? Regards, Ali Taher ___ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] worker processes
Thanks for your answer. Please note that I have set 3 connection strings in SQLOPS module for 3 Postgres databases. How many connections per database does Kamailio open? (knowing that children=12 and only one listening interface). Regards, Ali Taher From: Daniel-Constantin Mierla Sent: Tuesday, September 1, 2020 1:18 PM To: Kamailio (SER) - Users Mailing List ; Ali Taher Subject: Re: [SR-Users] worker processes Hello, note that other processes are created by kamailio, not only sip worker processes. For example, there will be timer processes. The dedicated worker processes are only for UDP sockets, not for tcp/tls (here is a single pool of workers for all tcp/tls sockets). If you do not receive traffic on a specific udp socket, then do not listen to it, you have a group of processes not doing anything. If you need to listen on a socket, but have less traffic (e.g., used for monitoring), then use workers parameter before listen, so you get less worker processes for that udp socket. Cheers, Daniel On 01.09.20 12:03, Ali Taher wrote: Hi, I just want to ask regarding the children number and listening interfaces set in Kamailio configuration. I am given to understand that the total number of worker processes there will be (children_setting * listeners) , so assuming I have set children=12 and listen=udp:172.16.1.12:5060 (which is the interface IP where the SIP packets as received), in this case there will be 12 worker processes and 12 database connections to postgresql (I'm using SQLOPS). My question is if I set listen=udp:0.0.0.0:5060 , knowing that I have 3 interfaces, in this case the number of worker processes will be 36. Is this good configuration ? will the other 24 processes be used or they are reserved to the 2 other interfaces which are not receiving any SIP packets and then will be useless ? Regards, Ali Taher ___ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org<mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla -- www.asipto.com<http://www.asipto.com> www.twitter.com/miconda<http://www.twitter.com/miconda> -- www.linkedin.com/in/miconda<http://www.linkedin.com/in/miconda> Funding: https://www.paypal.me/dcmierla ___ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] worker processes
Hi Daniel,
I meant that I'm defining 3 connection strings using sqlcon parameter of sqlops
module as shown below:
modparam("sqlops","sqlcon","cb=>postgres://postgres:kamailio@localhost:5432/cataleya")
modparam("sqlops","sqlcon","mi=>postgres://postgres:kamailio@localhost:5432/cataleya_mitto")
modparam("sqlops","sqlcon","sp=>postgres://postgres:kamailio@localhost:5432/cataleya_spactron")
regards,
Ali Taher
From: Daniel-Constantin Mierla
Sent: Tuesday, September 1, 2020 1:52 PM
To: Ali Taher ; Kamailio (SER) - Users Mailing List
Subject: Re: [SR-Users] worker processes
hello,
what do you mean by "3 connection strings"? Maybe you can give similar example
here.
Cheers,
Daniel
On 01.09.20 12:33, Ali Taher wrote:
Thanks for your answer.
Please note that I have set 3 connection strings in SQLOPS module for 3
Postgres databases. How many connections per database does Kamailio open?
(knowing that children=12 and only one listening interface).
Regards,
Ali Taher
From: Daniel-Constantin Mierla <mailto:mico...@gmail.com>
Sent: Tuesday, September 1, 2020 1:18 PM
To: Kamailio (SER) - Users Mailing List
<mailto:sr-users@lists.kamailio.org>; Ali Taher
<mailto:ata...@vanrise.com>
Subject: Re: [SR-Users] worker processes
Hello,
note that other processes are created by kamailio, not only sip worker
processes. For example, there will be timer processes.
The dedicated worker processes are only for UDP sockets, not for tcp/tls (here
is a single pool of workers for all tcp/tls sockets). If you do not receive
traffic on a specific udp socket, then do not listen to it, you have a group of
processes not doing anything. If you need to listen on a socket, but have less
traffic (e.g., used for monitoring), then use workers parameter before listen,
so you get less worker processes for that udp socket.
Cheers,
Daniel
On 01.09.20 12:03, Ali Taher wrote:
Hi,
I just want to ask regarding the children number and listening interfaces set
in Kamailio configuration.
I am given to understand that the total number of worker processes there will
be (children_setting * listeners) , so assuming I have set children=12 and
listen=udp:172.16.1.12:5060 (which is the interface IP where the SIP packets as
received), in this case there will be 12 worker processes and 12 database
connections to postgresql (I'm using SQLOPS).
My question is if I set listen=udp:0.0.0.0:5060 , knowing that I have 3
interfaces, in this case the number of worker processes will be 36. Is this
good configuration ? will the other 24 processes be used or they are reserved
to the 2 other interfaces which are not receiving any SIP packets and then will
be useless ?
Regards,
Ali Taher
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Funding: https://www.paypal.me/dcmierla
--
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www.twitter.com/miconda<http://www.twitter.com/miconda> --
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Re: [SR-Users] worker processes
Hi Daniel,
So in my case :
* Server vcpus = 24
* Max_worker_processes in postgres = 24
* Max_parallel_workers in postgres = 24
* Children in Kamailio config = 12
* 1 udp listener
I noticed that there is 48 database connections established by Kamailio (16 x
3) , can this cause an issue knowing that maximum worker processes is 24 ?
Regards,
Ali Taher
From: Daniel-Constantin Mierla
Sent: Tuesday, September 1, 2020 6:48 PM
To: Ali Taher ; Kamailio (SER) - Users Mailing List
Subject: Re: [SR-Users] worker processes
Hello,
in this case there are 3 distinct connections per process, so you have to
multiply the number of processes by 3 to get the overall number of connections.
You can run 'kamctl ps' to see all the processes of a kamailio instance.
Actually a few kamailio processes may not open database connections, a matter
of their purpose if they are created by specific modules, but for a rough upper
limit, this is the formula.
Cheers,
Daniel
On 01.09.20 16:25, Ali Taher wrote:
Hi Daniel,
I meant that I'm defining 3 connection strings using sqlcon parameter of sqlops
module as shown below:
modparam("sqlops","sqlcon","cb=>postgres://postgres:kamailio@localhost:5432/cataleya")
modparam("sqlops","sqlcon","mi=>postgres://postgres:kamailio@localhost:5432/cataleya_mitto")
modparam("sqlops","sqlcon","sp=>postgres://postgres:kamailio@localhost:5432/cataleya_spactron")
regards,
Ali Taher
From: Daniel-Constantin Mierla <mailto:mico...@gmail.com>
Sent: Tuesday, September 1, 2020 1:52 PM
To: Ali Taher <mailto:ata...@vanrise.com>; Kamailio (SER) -
Users Mailing List
<mailto:sr-users@lists.kamailio.org>
Subject: Re: [SR-Users] worker processes
hello,
what do you mean by "3 connection strings"? Maybe you can give similar example
here.
Cheers,
Daniel
On 01.09.20 12:33, Ali Taher wrote:
Thanks for your answer.
Please note that I have set 3 connection strings in SQLOPS module for 3
Postgres databases. How many connections per database does Kamailio open?
(knowing that children=12 and only one listening interface).
Regards,
Ali Taher
From: Daniel-Constantin Mierla <mailto:mico...@gmail.com>
Sent: Tuesday, September 1, 2020 1:18 PM
To: Kamailio (SER) - Users Mailing List
<mailto:sr-users@lists.kamailio.org>; Ali Taher
<mailto:ata...@vanrise.com>
Subject: Re: [SR-Users] worker processes
Hello,
note that other processes are created by kamailio, not only sip worker
processes. For example, there will be timer processes.
The dedicated worker processes are only for UDP sockets, not for tcp/tls (here
is a single pool of workers for all tcp/tls sockets). If you do not receive
traffic on a specific udp socket, then do not listen to it, you have a group of
processes not doing anything. If you need to listen on a socket, but have less
traffic (e.g., used for monitoring), then use workers parameter before listen,
so you get less worker processes for that udp socket.
Cheers,
Daniel
On 01.09.20 12:03, Ali Taher wrote:
Hi,
I just want to ask regarding the children number and listening interfaces set
in Kamailio configuration.
I am given to understand that the total number of worker processes there will
be (children_setting * listeners) , so assuming I have set children=12 and
listen=udp:172.16.1.12:5060 (which is the interface IP where the SIP packets as
received), in this case there will be 12 worker processes and 12 database
connections to postgresql (I'm using SQLOPS).
My question is if I set listen=udp:0.0.0.0:5060 , knowing that I have 3
interfaces, in this case the number of worker processes will be 36. Is this
good configuration ? will the other 24 processes be used or they are reserved
to the 2 other interfaces which are not receiving any SIP packets and then will
be useless ?
Regards,
Ali Taher
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www.twitter.com/miconda<http://www.twitter.com/miconda> --
www.linkedin.com/in/miconda<http://www.linkedin.com/in/miconda>
Funding: https://www.paypal.me/dcmierla
--
Daniel-Constantin Mierla -- www.asipto.com<http://www.asipto.com>
www.twitter.com/miconda<http://www.twitter.com/miconda> --
www.linkedin.com/in/miconda<http://www.linkedin.com/in/miconda>
Funding: https://www.paypal.me/dcmierla
--
Daniel-Constantin Mierla -- www.asipto.com<http://www.asipto.com>
www.twitter.com/miconda<http://www.twitter.com/miconda> --
www.linkedin.com/in/miconda<http://www.linkedin.com/in/miconda>
Funding: https://www.paypal.me/dcmierla
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Re: [SR-Users] worker processes
Hi Alex,
That is clear. I decreased the max worker processes to 20 to be on the safe
side. But what is confusing me is what about the 18 worker processes ran by
Kamailio, will they overlap with postgres processes or they are the same ,
knowing that Kamailio role in my case is just getting routing decision by
running a query on the database and send it back to the SBC. Excuse my lack of
knowledge in how linux processes work.
Regards,
Ali Taher
-Original Message-
From: sr-users On Behalf Of Alex Balashov
Sent: Tuesday, September 1, 2020 8:22 PM
To: sr-users@lists.kamailio.org
Subject: Re: [SR-Users] worker processes
On 2020-09-01 12:08, Ali Taher wrote:
> I noticed that there is 48 database connections established by
> Kamailio
> (16 x 3) , can this cause an issue knowing that maximum worker
> processes is 24 ?
Postgres's ability to serve clients and accommodate queries isn't limited by
the number of max_worker_processes. The default is 8 if this value is unset,
yet any stock PostgreSQL instance can accommodate more than 8 concurrent
connections, clearly. max_worker_processes != max_connections.
As far as the wisdom of setting process limits to equal the number of available
CPU cores ("VCPUs"), there's no absolute answer to that. The general danger
with doing this is just that a particularly expensive or destructive statement
can consume all of the hardware resources, with no safety margins for other
processes, and perhaps not even enough for someone to log in and resolve the
problem. So, the general recommendation is not allow PostgreSQL to consume
every single VCPU or max out any other resource, and that's why most Linux
distributions'
server packages come with these types of settings conservatively tuned.
-- Alex
--
Alex Balashov | Principal | Evariste Systems LLC
Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
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[SR-Users] voltdb support
Hello, I just need to know if kamailio can communicate with Voltdb for sqlops operations for example. Regards, Ali Taher ___ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] voltdb support
Hi Alex, Thanks for your answer. How to proceed to let Kamailio talk manually to VoltDB ? Regards, Ali Taher From: sr-users On Behalf Of Alex Balashov Sent: Monday, December 14, 2020 6:38 PM To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] voltdb support Is VoltDB’s wire protocol binary-compatible with Postgres, MySQL, Oracle, or anything reachable via UnixODBC? From what I understand, no. So, you can’t use SQL-native modules to talk to it. You can, of course, get Kamailio to talk to it “manually” for select purposes. — Sent from mobile, with due apologies for brevity and errors. On Dec 14, 2020, at 11:28 AM, Ali Taher mailto:ata...@vanrise.com>> wrote: Hello, I just need to know if kamailio can communicate with Voltdb for sqlops operations for example. Regards, Ali Taher ___ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org<mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users ___ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] voltdb support
What I need is to fetch routing from VoltDB and send it back to switch using 3xx response. So I think the API solution is the best approach here. Thanks for all this useful info. -Original Message- From: sr-users On Behalf Of Alex Balashov Sent: Monday, December 14, 2020 10:16 PM To: sr-users@lists.kamailio.org Subject: Re: [SR-Users] voltdb support You'd have to devise some sort of API or use something that Kamailio can talk to, unless you're inclined to write a custom module that uses a library[1]. Otherwise, the usual integration paths are available, e.g. 1) You could put a REST API server in front of VoltDB and use: https://kamailio.org/docs/modules/5.4.x/modules/http_async_client.html https://kamailio.org/docs/modules/5.4.x/modules/http_client.html 2) Or, you could write a server of some kind which mediates these requests using EVAPI: https://kamailio.org/docs/modules/5.4.x/modules/evapi.html 3) Or, you could write a minimal SIP redirect service using a library such as PJSUA/PJSIP, or JAIN, https://www.oracle.com/technical-resources/articles/enterprise-architecture/introduction-jain-sip-part1.html which would allow Kamailio to consume 3xx redirects as a transport for information queries. That is a common pattern for external routing services, for example. It's also worth keeping in mind that even if you invent such a shim, you're not going to be able to use VoltDB to provide database backing for Kamailio's database-backed modules (e.g. htable, usrloc, dispatcher, carrierroute, etc). If your aim is to make VoltDB work transparently with those modules, then this isn't a viable project. What you can do is query an API for some limited purposes. For example, you could get authentication credentials from VoltDB and store them in pseudovariables so as to feed them into functions which can use them, such as: https://kamailio.org/docs/modules/5.4.x/modules/auth.html#auth.f.pv_auth_check But most Kamailio modules which extensively utilise a relational database don't expose capabilities like that. That would, after all, contradict the basic purpose for which a lot of them exist, which is to transparently manage database interactions for you while providing terse and easy-to-use syntactical sugar. So, it depends on what exactly you want to do. If it's something very specific, like getting routes or authentication credentials, you could potentially use an unsupported database if you devise a means of communicating with it indirectly, using a generic channel that Kamailio does support. But if you're asking if Kamailio supports VoltDB in a rather general sense, the answer is no. -- Alex [1] https://docs.voltdb.com/UsingVoltDB/ChapAppDesign.php On 12/14/20 2:57 PM, Ali Taher wrote: > Hi Alex, > > Thanks for your answer. How to proceed to let Kamailio talk manually > to VoltDB ? > > Regards, > > Ali Taher -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/ ___ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users ___ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] voltdb support
What do you recommend using : http sync or async client modules? -Original Message- From: sr-users On Behalf Of Alex Balashov Sent: Monday, December 14, 2020 10:31 PM To: sr-users@lists.kamailio.org Subject: Re: [SR-Users] voltdb support On 12/14/20 3:28 PM, Ali Taher wrote: > What I need is to fetch routing from VoltDB and send it back to switch > using 3xx response. So I think the API solution is the best approach > here. Thanks for all this useful info. > If Kamailio is sending the 3xx response, then it sounds like using the HTTP modules to interact with VoltDB's JSON interface might be the best way to go. You'll have to account for the latency inherent in such queries and make sure you have a sufficient number of worker processes to handle the load, whether you use the async or synchronous HTTP modules. For more information on that topic, you can read an old blog post of mine: http://www.evaristesys.com/blog/tuning-kamailio-for-high-throughput-and-performance/ -- Alex -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/ ___ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users ___ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] Kamailio bypassing RTP media
Hi everyone, I'm new to kamailio. I plan to setup a sip server cluster, Does someone can give me some suggestions if I can use kamailio as front server, which handle sip message and bypass rtp media message to backendserver, this mean the kamailio handle only sip message, the rtp packets are sent directed to backend server. The sip sequences is 1.softswitch send invite to kamailio 2.kamailio send invite to backend server. 3.backend server return 200 ok to kamailio 4.kamailio return 200 ok to softswitch, the ip and port in sip is from the backend server's msg, this make the softswitch send rtp direct to backend server. 5.rtp packets are transfered between softswitch and backend server. 6.backend server send bye to kamailio 7.kamailio send bye to softswitch. Any help is appreciated. Regards, Ali Taher ___ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] Kamailio bypassing RTP media
Hi Alex, What do I have to put in routing logic in this case ? Regards, Ali Taher From: sr-users On Behalf Of Alex Balashov Sent: Friday, May 3, 2019 5:45 PM To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] Kamailio bypassing RTP media Yes, this is possible. In fact, this is Kamailio’s default behaviour, since it is inherently a SIP proxy. It would not touch the SDP unless you asked it to. — Sent from mobile, with due apologies for brevity and errors. On May 3, 2019, at 4:39 PM, Ali Taher mailto:ata...@vanrise.com>> wrote: Hi everyone, I'm new to kamailio. I plan to setup a sip server cluster, Does someone can give me some suggestions if I can use kamailio as front server, which handle sip message and bypass rtp media message to backendserver, this mean the kamailio handle only sip message, the rtp packets are sent directed to backend server. The sip sequences is 1.softswitch send invite to kamailio 2.kamailio send invite to backend server. 3.backend server return 200 ok to kamailio 4.kamailio return 200 ok to softswitch, the ip and port in sip is from the backend server's msg, this make the softswitch send rtp direct to backend server. 5.rtp packets are transfered between softswitch and backend server. 6.backend server send bye to kamailio 7.kamailio send bye to softswitch. Any help is appreciated. Regards, Ali Taher ___ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org<mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users ___ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] Kamailio not responding in heavy load
Hello all, I'm using sqlops module to connect to postgres and get the routing of calls from postgresql database. In order to troubleshoot issues in case they happened, I added each call details to Kamailio log. In some cases, when there is heavy traffic load (around 100 cps), I don't find some calls in Kamailio log, knowing that I can find them on the switch log (switch sending the invite packets to Kamailio). When checking the traces on these calls taken by the switch, it's clear that Kamailio is not responding to the invite packets sent by the switch. It could be because there are no available process to handle the call at this moment (I have set fork=14) Is it possible to check if there are calls pending in the queue? Regards, __ Kamailio - Users Mailing List - Non Commercial Discussions * sr-users@lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe: * https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] packets exceeding MTU size
Hello all,
Lately I faced an issue that Kamailio is not responding to some invite packets
coming from SBC. When investigating I found that these packets exceeds MTU size.
Could this be the issue? Knowing that in Kamailio configuration I have a
condition on is_method("INVITE"). Could be that in case of packet exceeding MTU
size it's not considered as invite packet by Kamailio?
I have Kamailio 5.2.5.
Regards,
Ali Taher
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[SR-Users] Randomize calling number
Hello, What is the best way to change the calling number to a random number (based on specific pattern, like number of digits for example) in an invite packet without using a database. So the scenario is : * receive an invite packet from certain source * change the calling number * send back the invite packet including the new calling number to the original source. Regards, Ali Taher __ Kamailio - Users Mailing List - Non Commercial Discussions To unsubscribe send an email to sr-users-le...@lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe:
[SR-Users] Re: Randomize calling number
Hi Henning, I used the uac_replace_from() function to replace the fU, but the value is still the same when i printed it using xlog. I also dont know how to send the invite back to the switch. What function to use? Thanks, Sent from my iPhone On Jul 6, 2023, at 5:31 PM, Henning Westerholt wrote: Hello, Not 100% sure if I understood your question correctly. There are probably many ways. One option is to use the $RANDOM pseudo-variable that can generate (pseudo-)random numbers. You can get it to the pattern you like with some script operations in the cfg. Just rewrite the request URI user of the INVITE in Kamailio to the new generated number and send it out again. Cheers, Henning -- Henning Westerholt – https://skalatan.de/blog/ Kamailio services – https://gilawa.com<https://gilawa.com/> From: Ali Taher Sent: Donnerstag, 6. Juli 2023 15:15 To: sr-users@lists.kamailio.org Subject: [SR-Users] Randomize calling number Hello, What is the best way to change the calling number to a random number (based on specific pattern, like number of digits for example) in an invite packet without using a database. So the scenario is : * receive an invite packet from certain source * change the calling number * send back the invite packet including the new calling number to the original source. Regards, Ali Taher __ Kamailio - Users Mailing List - Non Commercial Discussions To unsubscribe send an email to sr-users-le...@lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe:
[SR-Users] Re: Randomize calling number
Thanks Alex, Yes I'm aware of that, this is just for testing purposes. Regards, -Original Message- From: Alex Balashov Sent: Thursday, July 6, 2023 6:37 PM To: Kamailio (SER) - Users Mailing List Subject: [SR-Users] Re: Randomize calling number You can just modify the domain portion of the RURI, i.e. by manipulating $rd. For example, $rd = "some.ip"; Then, t_relay(). However, beware that sending the call back to the same gateway from which it came is likely to cause call loops. Proxies don't create new call legs, they are just part of the same logical call leg; call leg "A" comes in, and call leg "A" goes back out, with the same Call-ID, CSeq, etc. If you send the INVITE back to the same gateway that originated the request, it will likely reject it since it is literally the same request it itself created. -- Alex > On Jul 6, 2023, at 11:02 AM, Ali Taher wrote: > > Hi Henning, > > I used the uac_replace_from() function to replace the fU, but the value is > still the same when i printed it using xlog. > > I also dont know how to send the invite back to the switch. What function to > use? > > Thanks, > > Sent from my iPhone > >> On Jul 6, 2023, at 5:31 PM, Henning Westerholt wrote: >> >> Hello, >> Not 100% sure if I understood your question correctly. >> There are probably many ways. One option is to use the $RANDOM >> pseudo-variable that can generate (pseudo-)random numbers. You can get it to >> the pattern you like with some script operations in the cfg. >> Just rewrite the request URI user of the INVITE in Kamailio to the new >> generated number and send it out again. >> Cheers, >> Henning >> -- Henning Westerholt – https://skalatan.de/blog/ Kamailio services >> – https://gilawa.com >> From: Ali Taher >> Sent: Donnerstag, 6. Juli 2023 15:15 >> To: sr-users@lists.kamailio.org >> Subject: [SR-Users] Randomize calling number Hello, What is the >> best way to change the calling number to a random number (based on specific >> pattern, like number of digits for example) in an invite packet without >> using a database. >> So the scenario is : >> • receive an invite packet from certain source >> • change the calling number >> • send back the invite packet including the new calling number to the >> original source. >> Regards, >> Ali Taher > > __ > Kamailio - Users Mailing List - Non Commercial Discussions To > unsubscribe send an email to sr-users-le...@lists.kamailio.org > Important: keep the mailing list in the recipients, do not reply only to the > sender! > Edit mailing list options or unsubscribe: -- Alex Balashov Principal Consultant Evariste Systems LLC Web: https://evaristesys.com Tel: +1-706-510-6800 __ Kamailio - Users Mailing List - Non Commercial Discussions To unsubscribe send an email to sr-users-le...@lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe: __ Kamailio - Users Mailing List - Non Commercial Discussions To unsubscribe send an email to sr-users-le...@lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe:
[SR-Users] Re: Randomize calling number
Hi Henning, Is there a way to change From URI user part without touching the domain ? As I can see in uac_replace_from there is an option to either change the display part or the whole URI part. Regards, From: Henning Westerholt Sent: Thursday, July 6, 2023 6:42 PM To: Kamailio (SER) - Users Mailing List Cc: Ali Taher Subject: RE: Randomize calling number Hello, ah, you want to change the From URI user. Yes, uac_replace_from is a common approach for that. This function will change the From URI when you send out the message, you will not see it with a xlog in the cfg. https://www.kamailio.org/wikidocs/tutorials/faq/main/#why-changes-made-to-headers-or-body-are-not-visible-immediately Regarding sending out the SIP message, a common approach is to use rewrite the request URI to the new destination and then use “t_relay” to send it. Cheers, Henning -- Henning Westerholt – https://skalatan.de/blog/ Kamailio services – https://gilawa.com<https://gilawa.com/> From: Ali Taher mailto:ata...@vanrise.com>> Sent: Donnerstag, 6. Juli 2023 17:02 To: Henning Westerholt mailto:h...@gilawa.com>> Cc: Kamailio (SER) - Users Mailing List mailto:sr-users@lists.kamailio.org>> Subject: Re: Randomize calling number Hi Henning, I used the uac_replace_from() function to replace the fU, but the value is still the same when i printed it using xlog. I also dont know how to send the invite back to the switch. What function to use? Thanks, Sent from my iPhone On Jul 6, 2023, at 5:31 PM, Henning Westerholt mailto:h...@gilawa.com>> wrote: Hello, Not 100% sure if I understood your question correctly. There are probably many ways. One option is to use the $RANDOM pseudo-variable that can generate (pseudo-)random numbers. You can get it to the pattern you like with some script operations in the cfg. Just rewrite the request URI user of the INVITE in Kamailio to the new generated number and send it out again. Cheers, Henning -- Henning Westerholt – https://skalatan.de/blog/ Kamailio services – https://gilawa.com<https://gilawa.com/> From: Ali Taher mailto:ata...@vanrise.com>> Sent: Donnerstag, 6. Juli 2023 15:15 To: sr-users@lists.kamailio.org<mailto:sr-users@lists.kamailio.org> Subject: [SR-Users] Randomize calling number Hello, What is the best way to change the calling number to a random number (based on specific pattern, like number of digits for example) in an invite packet without using a database. So the scenario is : 1. receive an invite packet from certain source 2. change the calling number 3. send back the invite packet including the new calling number to the original source. Regards, Ali Taher __ Kamailio - Users Mailing List - Non Commercial Discussions To unsubscribe send an email to sr-users-le...@lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe:
[SR-Users] Random number
Hello, As per Kamailio documentation, $RANDOM returns a random value from the [0 - 2^31) range. But when testing it it's always returning a random number of 9 or 10 digits. I'm not sure if it's always the case or if it can also returns numbers with 1 or 2 digits as well. Another question, how can I force it to return a number of specific digits starting with certain prefix. Regards, __ Kamailio - Users Mailing List - Non Commercial Discussions To unsubscribe send an email to sr-users-le...@lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe:
[SR-Users] 180 and 183 SIP messages
Hello everyone, I'm working on freeswitch , where I have SIP calls coming from A-Party to my freeswitch and I'm routing it to B-Party. I'm facing an issue , when I'm getting back the 183 message from the supplier(B-Party) and forwarding it to the customer(A-Party); the customer is not hearing the RBT and asking to send 180 rather than 183 . (the supplier is only sending 183 without 180). Is it possible using kamailio to replace this 183 message with 180 message , or even add 180 message while keeping the same 183 message ? Regards, Ali Taher ___ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] 180 and 183 SIP messages
Hi Arsen, I know that such thing can be done through freeswitch , but I need to do it using Kamailio for specific reason. Regards, Ali Taher From: sr-users [mailto:sr-users-boun...@lists.kamailio.org] On Behalf Of Arsen Sent: Thursday, March 29, 2018 6:49 PM To: Kamailio (SER) - Users Mailing List Subject: Re: [SR-Users] 180 and 183 SIP messages Hi, It should be possible with freeswitch as well, google for freeswitch cause code substitution and you will see example. Regards, Arsen Semionov www.eurolan.info <http://www.eurolan.info> cell: +442035198881 On Thu, Mar 29, 2018 at 6:44 PM, Ali Taher mailto:ata...@vanrise.com> > wrote: Hello everyone, I’m working on freeswitch , where I have SIP calls coming from A-Party to my freeswitch and I’m routing it to B-Party. I’m facing an issue , when I’m getting back the 183 message from the supplier(B-Party) and forwarding it to the customer(A-Party); the customer is not hearing the RBT and asking to send 180 rather than 183 . (the supplier is only sending 183 without 180). Is it possible using kamailio to replace this 183 message with 180 message , or even add 180 message while keeping the same 183 message ? Regards, Ali Taher ___ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org <mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users ___ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] 180 and 183 SIP messages
Thank you Alex for your response.
So if I want to add a header to the INVITE packet , and also detect if the
message is 183 and then change it to 180 then forward it back , does is do the
job :
route{
if (is_method("INVITE"))
{
append_hf("Test: this is a test");
}
forward(127.0.0.1,5060);
}
reply_route {
if (status == "183") {
change_reply_status(180, "Ringing");
exit;
}
}
Regards,
Ali Taher
-Original Message-
From: sr-users [mailto:sr-users-boun...@lists.kamailio.org] On Behalf Of Alex
Balashov
Sent: Thursday, March 29, 2018 9:02 PM
To: Kamailio (SER) - Users Mailing List
Subject: Re: [SR-Users] 180 and 183 SIP messages
You can absolutely do it:
https://kamailio.org/docs/modules/5.1.x/modules/textopsx.html#textopsx.f.change_reply_status
You can even remove the embedded SDP if desired, using the various textops
functions:
https://kamailio.org/docs/modules/5.1.x/modules/textops.html
Whether you _should_ do this just because you _can_, on the other hand, is a
question I would consider very carefully.
On Thu, Mar 29, 2018 at 06:56:52PM +0300, Ali Taher wrote:
> Hi Arsen,
>
>
>
> I know that such thing can be done through freeswitch , but I need to do it
> using Kamailio for specific reason.
>
>
>
> Regards,
>
> Ali Taher
>
>
>
> From: sr-users [mailto:sr-users-boun...@lists.kamailio.org] On Behalf
> Of Arsen
> Sent: Thursday, March 29, 2018 6:49 PM
> To: Kamailio (SER) - Users Mailing List
> Subject: Re: [SR-Users] 180 and 183 SIP messages
>
>
>
> Hi,
>
>
>
> It should be possible with freeswitch as well, google for freeswitch cause
> code substitution and you will see example.
>
>
>
>
>
> Regards,
>
>
>
>
>
>
>
>
> Arsen Semionov
>
> www.eurolan.info <http://www.eurolan.info>
>
> cell: +442035198881
>
>
>
> On Thu, Mar 29, 2018 at 6:44 PM, Ali Taher <mailto:ata...@vanrise.com> > wrote:
>
> Hello everyone,
>
>
>
> I’m working on freeswitch , where I have SIP calls coming from A-Party to my
> freeswitch and I’m routing it to B-Party.
>
>
>
> I’m facing an issue , when I’m getting back the 183 message from the
> supplier(B-Party) and forwarding it to the customer(A-Party); the customer is
> not hearing the RBT and asking to send 180 rather than 183 . (the supplier is
> only sending 183 without 180).
>
>
>
> Is it possible using kamailio to replace this 183 message with 180 message ,
> or even add 180 message while keeping the same 183 message ?
>
>
>
> Regards,
>
> Ali Taher
>
>
>
>
> ___
> Kamailio (SER) - Users Mailing List
> sr-users@lists.kamailio.org <mailto:sr-users@lists.kamailio.org>
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>
>
>
> ___
> Kamailio (SER) - Users Mailing List
> sr-users@lists.kamailio.org
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
--
Alex Balashov | Principal | Evariste Systems LLC
Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
___
Kamailio (SER) - Users Mailing List
sr-users@lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
___
Kamailio (SER) - Users Mailing List
sr-users@lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] 180 and 183 SIP messages
Hello,
As I understand now , append_hf function must be called from either
REQUEST_ROUTE, ONREPLY_ROUTE, FAILURE_ROUTE or BRANCH_ROUTE.
I'm not sure if this what I need in my case , where I want to insert a new
header to the INVITE packet then forward it .
Any help would be appreciated.
Regards,
Ali Taher
-Original Message-
From: Ali Taher [mailto:ata...@vanrise.com]
Sent: Friday, March 30, 2018 10:02 AM
To: 'Kamailio (SER) - Users Mailing List'
Cc: 'Ali Taher~Vanrise Technical Support'
Subject: RE: [SR-Users] 180 and 183 SIP messages
Thank you Alex for your response.
So if I want to add a header to the INVITE packet , and also detect if the
message is 183 and then change it to 180 then forward it back , does is do the
job :
route{
if (is_method("INVITE"))
{
append_hf("Test: this is a test");
}
forward(127.0.0.1,5060);
}
reply_route {
if (status == "183") {
change_reply_status(180, "Ringing");
exit;
}
}
Regards,
Ali Taher
-Original Message-
From: sr-users [mailto:sr-users-boun...@lists.kamailio.org] On Behalf Of Alex
Balashov
Sent: Thursday, March 29, 2018 9:02 PM
To: Kamailio (SER) - Users Mailing List
Subject: Re: [SR-Users] 180 and 183 SIP messages
You can absolutely do it:
https://kamailio.org/docs/modules/5.1.x/modules/textopsx.html#textopsx.f.change_reply_status
You can even remove the embedded SDP if desired, using the various textops
functions:
https://kamailio.org/docs/modules/5.1.x/modules/textops.html
Whether you _should_ do this just because you _can_, on the other hand, is a
question I would consider very carefully.
On Thu, Mar 29, 2018 at 06:56:52PM +0300, Ali Taher wrote:
> Hi Arsen,
>
>
>
> I know that such thing can be done through freeswitch , but I need to do it
> using Kamailio for specific reason.
>
>
>
> Regards,
>
> Ali Taher
>
>
>
> From: sr-users [mailto:sr-users-boun...@lists.kamailio.org] On Behalf
> Of Arsen
> Sent: Thursday, March 29, 2018 6:49 PM
> To: Kamailio (SER) - Users Mailing List
> Subject: Re: [SR-Users] 180 and 183 SIP messages
>
>
>
> Hi,
>
>
>
> It should be possible with freeswitch as well, google for freeswitch cause
> code substitution and you will see example.
>
>
>
>
>
> Regards,
>
>
>
>
>
>
>
>
> Arsen Semionov
>
> www.eurolan.info <http://www.eurolan.info>
>
> cell: +442035198881
>
>
>
> On Thu, Mar 29, 2018 at 6:44 PM, Ali Taher <mailto:ata...@vanrise.com> > wrote:
>
> Hello everyone,
>
>
>
> I’m working on freeswitch , where I have SIP calls coming from A-Party to my
> freeswitch and I’m routing it to B-Party.
>
>
>
> I’m facing an issue , when I’m getting back the 183 message from the
> supplier(B-Party) and forwarding it to the customer(A-Party); the customer is
> not hearing the RBT and asking to send 180 rather than 183 . (the supplier is
> only sending 183 without 180).
>
>
>
> Is it possible using kamailio to replace this 183 message with 180 message ,
> or even add 180 message while keeping the same 183 message ?
>
>
>
> Regards,
>
> Ali Taher
>
>
>
>
> ___
> Kamailio (SER) - Users Mailing List
> sr-users@lists.kamailio.org <mailto:sr-users@lists.kamailio.org>
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>
>
>
> ___
> Kamailio (SER) - Users Mailing List
> sr-users@lists.kamailio.org
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
--
Alex Balashov | Principal | Evariste Systems LLC
Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
___
Kamailio (SER) - Users Mailing List
sr-users@lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
___
Kamailio (SER) - Users Mailing List
sr-users@lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
[SR-Users] SIP re-invite Packets
Hello,
I'm using below routing logic to change the 183 reply to 180 and send it
back to the origination.
Noting that Kamailio is listening on 5065.
route{
if (is_method("INVITE")) {
xlog("L_INFO","INSIDE request route $si \n");
insert_hf("X-AUTH-IP: $si\r\n");
}
route(FWD);
}
reply_route {
#xlog("L_INFO","INSDE reply \n");
if (status == "183") {
change_reply_status(180, "Ringing");
exit;
}
}
route[FWD] {
#xlog("L_INFO","INSIDE FWD \n");
forward(x.x.x.x,5060);
}
Below is the scenario , where the customer is sending packets to Kamailio ,
and Kamailio is forwarding them the supplier (with IP x.x.x.x) , then the
supplier reply back to kamailio and kamailio relay the reply to the
customer
Customer --> Kamailio ---> ---Supplier---
I have a case where the supplier sends in some cases an in-dialog invite
(Re-Invite) , which is considered I think as SIP-Request and thus as per
above routing logic , kamailio is sending it back to the supplier , yet I
need to send it to the customer.
How can I solve this issue?
Thanks
Ali Taher
___
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sr-users@lists.kamailio.org
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Re: [SR-Users] SIP re-invite Packets
Hi Alex,
So loose_route() is only responsible of In-dialog requests ?
Can you please send me a simplified config that suit my case ?
Noting that I'm using Kamailio only as proxy between my customer and my
supplier.
Much appreciated.
Ali Taher
-Original Message-
From: sr-users On Behalf Of Alex Balashov
Sent: Tuesday, July 17, 2018 3:44 PM
To: Kamailio (SER) - Users Mailing List
Cc: sr-us...@lists.sip-router.org
Subject: Re: [SR-Users] SIP re-invite Packets
Hi,
In-dialog requests must be handled using loose_route(). You can find the proper
boilerplate in the stock configuration file:
https://github.com/kamailio/kamailio/blob/master/etc/kamailio.cfg#L587
To note, your config is not sufficient to handle all aspects of a call between
endpoints in an adequate way. You will want to consider the main request_route
in the main config file and adapt its most essential aspects accordingly:
https://github.com/kamailio/kamailio/blob/master/etc/kamailio.cfg#L455
-- Alex
On Tue, Jul 17, 2018 at 03:40:58PM +0300, Ali Taher wrote:
> Hello,
>
>
>
> I'm using below routing logic to change the 183 reply to 180 and send
> it back to the origination.
>
>
>
> Noting that Kamailio is listening on 5065.
>
> route{
>
> if (is_method("INVITE")) {
>
> xlog("L_INFO","INSIDE request route $si \n");
>
> insert_hf("X-AUTH-IP: $si\r\n");
>
> }
>
> route(FWD);
>
> }
>
>
>
> reply_route {
>
> #xlog("L_INFO","INSDE reply \n");
>
> if (status == "183") {
>
> change_reply_status(180, "Ringing");
>
> exit;
>
> }
>
> }
>
>
>
> route[FWD] {
>
> #xlog("L_INFO","INSIDE FWD \n");
>
> forward(x.x.x.x,5060);
>
> }
>
>
>
>
>
> Below is the scenario , where the customer is sending packets to
> Kamailio , and Kamailio is forwarding them the supplier (with IP
> x.x.x.x) , then the supplier reply back to kamailio and kamailio
> relay the reply to the customer
>
>
>
> Customer --> Kamailio ---> ---Supplier---
>
>
>
> I have a case where the supplier sends in some cases an in-dialog
> invite
> (Re-Invite) , which is considered I think as SIP-Request and thus as
> per above routing logic , kamailio is sending it back to the supplier
> , yet I need to send it to the customer.
>
>
>
> How can I solve this issue?
>
>
>
> Thanks
>
> Ali Taher
>
> ___
> Kamailio (SER) - Users Mailing List
> sr-users@lists.kamailio.org
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
--
Alex Balashov | Principal | Evariste Systems LLC
Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
___
Kamailio (SER) - Users Mailing List
sr-users@lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
___
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sr-users@lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
Re: [SR-Users] SIP re-invite Packets
Hi Alex,
Thank you for your reply.
Does below config forward all sip requests coming from the customer to the
supplier , and relay sip replies back to customer ?
Where re-invite packets are handled in the below config file?
Thank you
Ali Taher
-Original Message-
From: sr-users On Behalf Of Alex Balashov
Sent: Tuesday, July 17, 2018 4:25 PM
To: Kamailio (SER) - Users Mailing List
Cc: sr-us...@lists.sip-router.org
Subject: Re: [SR-Users] SIP re-invite Packets
On Tue, Jul 17, 2018 at 04:21:25PM +0300, Ali Taher wrote:
> Can you please send me a simplified config that suit my case ?
Try this for a main request route:
---
route {
if(!mf_process_maxfwd_header("10")) {
sl_send_reply("483", "Too Many Hops");
exit;
}
# Maybe some sanity_check() here.
if(has_totag()) {
if(loose_route()) {
if(!t_relay())
sl_reply_error();
exit;
} else {
if(is_method("ACK")) {
if(t_check_trans())
t_relay();
} else
sl_send_reply("403", "Forbidden");
}
exit;
}
# CANCELs / branches.
if(is_method("CANCEL")) {
if(!t_relay_cancel()) {
sl_send_reply("500", "Server Internal Error");
exit;
}
exit;
}
t_check_trans();
if(is_method("OPTIONS")) {
sl_send_reply("200", "OK");
exit;
}
else if(is_method("INVITE")) {
# Route your calls.
exit;
}
sl_send_reply("403", "Method unsupported");
exit;
}
--
Alex Balashov | Principal | Evariste Systems LLC
Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
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Re: [SR-Users] SIP re-invite Packets
Thank you for your reply.
I tried the config file shared by Alex below , but I noticed that kamailio is
not dealing with CANCEL requests.
the scenario in my case is that I’m using Kamailio behind a switch to alter
some packets (precisely 183 packets) , so I need to forward all packets coming
from my customer to the switch , which will deal with it (send it to
appropriate supplier).
So what I need is the config file that will forward all requests coming to
Kamailio from my customer , to forward them to my switch , then reply back to
customer the reply coming from the switch. 9after altering 183 packet to 180) ,
handling the in-dialog invites sent by my switch , which needs to be sent to
the customer by kamailio.
Thanks
Ali Taher
From: sr-users On Behalf Of
y...@yu-boot.ru
Sent: Tuesday, July 17, 2018 9:41 PM
To: Kamailio (SER) - Users Mailing List
Cc: sr-us...@lists.sip-router.org
Subject: Re: [SR-Users] SIP re-invite Packets
ReInvites like all in-dialog requests are processed in has_totag()/loose_route
part. Default kamailio.cfg has almost identical part for this purpose.
Main idea is that an initial request (non in-dialog) hasn't To tag already. It
appears only when your request "hits" remote side (called or callee, as you
wish) and goes back to you. Simplified, but I hope you got an idea :)
--
sent from myMail for Android
вторник, 17 июля 2018г., 17:16 +03:00 от Ali Taher ata...@vanrise.com
<mailto:ata...@vanrise.com> :
Hi Alex,
Thank you for your reply.
Does below config forward all sip requests coming from the customer to the
supplier , and relay sip replies back to customer ?
Where re-invite packets are handled in the below config file?
Thank you
Ali Taher
-Original Message-
From: sr-users mailto:sr-users-boun...@lists.kamailio.org> > On Behalf Of Alex Balashov
Sent: Tuesday, July 17, 2018 4:25 PM
To: Kamailio (SER) - Users Mailing List mailto:sr-users@lists.kamailio.org> >
Cc: sr-us...@lists.sip-router.org <mailto:sr-us...@lists.sip-router.org>
Subject: Re: [SR-Users] SIP re-invite Packets
On Tue, Jul 17, 2018 at 04:21:25PM +0300, Ali Taher wrote:
> Can you please send me a simplified config that suit my case ?
Try this for a main request route:
---
route {
if(!mf_process_maxfwd_header("10")) {
sl_send_reply("483", "Too Many Hops");
exit;
}
# Maybe some sanity_check() here.
if(has_totag()) {
if(loose_route()) {
if(!t_relay())
sl_reply_error();
exit;
} else {
if(is_method("ACK")) {
if(t_check_trans())
t_relay();
} else
sl_send_reply("403", "Forbidden");
}
exit;
}
# CANCELs / branches.
if(is_method("CANCEL")) {
if(!t_relay_cancel()) {
sl_send_reply("500", "Server Internal Error");
exit;
}
exit;
}
t_check_trans();
if(is_method("OPTIONS")) {
sl_send_reply("200", "OK");
exit;
}
else if(is_method("INVITE")) {
# Route your calls.
exit;
}
sl_send_reply("403", "Method unsupported");
exit;
}
--
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Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free)
Web: http://www.evaristesys.com/, http://www.csrpswitch.com/
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[SR-Users] in-dialog requests
Hello,
I'm using Kamailio as a proxy behind freeswitch to alter some packets (183
packets precisely).
Here is my scenario:
(---Customer---) --> ((port 5065)---Kamailio---) -->
((port5060)---Freeswitch---) --> (---Supplier---)
Here is my request route :
route{
if (is_method("INVITE")) {
xlog("L_INFO","INSIDE request route $si \n");
insert_hf("X-AUTH-IP: $si\r\n");
}
route(FWD);
}
reply_route {
#xlog("L_INFO","INSDE reply \n");
if (status == "183") {
change_reply_status(180, "Ringing");
exit;
}
}
route[FWD] {
#xlog("L_INFO","INSIDE FWD \n");
forward(X.X.X.X,5060);
}
Where X.X.X.X is the Freeswitch IP
This script is working normally as long as the customer is sending the
requests(INVITE,BYE,ACK,CANCEL..) . But when the supplier or freeswitch
initiated the request , above script causes loop where the packet is bounced
between kamailio and freeswitch.
As I know , loose_route is used to handle such requests , but I'm not sure
how to forward these packets to the customer after receiving them from
freeswitch.
I'm not sure also if what's I'm doing is statefull or stateless.
Any help is appreciated.
Thanks,
Ali Taher
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Re: [SR-Users] in-dialog requests
Hi Alex,
Yes I did, but I noticed that all in-dialog requests sent by freeswitch to
kamailio were ignored and no packet was generated from kamailio in this regard.
I'm not sure if t_relay is working as expected at my side.
Thanks.
Ali Taher
-Original Message-
From: sr-users On Behalf Of Alex Balashov
Sent: Wednesday, July 18, 2018 4:34 PM
To: Kamailio (SER) - Users Mailing List
Subject: Re: [SR-Users] in-dialog requests
Have you tried the config I suggested to you yesterday?
On July 18, 2018 9:32:01 AM EDT, Ali Taher wrote:
>Hello,
>
>
>
>I'm using Kamailio as a proxy behind freeswitch to alter some packets
>(183
>packets precisely).
>
>
>
>Here is my scenario:
>
>
>
>(---Customer---) --> ((port 5065)---Kamailio---) -->
>((port5060)---Freeswitch---) --> (---Supplier---)
>
>
>
>Here is my request route :
>
>
>
>route{
>
>
>
>if (is_method("INVITE")) {
>
>xlog("L_INFO","INSIDE request route $si \n");
>
> insert_hf("X-AUTH-IP: $si\r\n");
>
>}
>
>
>
>route(FWD);
>
>}
>
>
>
>
>
>reply_route {
>
>#xlog("L_INFO","INSDE reply \n");
>
>if (status == "183") {
>
>change_reply_status(180, "Ringing");
>
>exit;
>
>}
>
>}
>
>
>
>route[FWD] {
>
>#xlog("L_INFO","INSIDE FWD \n");
>
>forward(X.X.X.X,5060);
>
>}
>
>Where X.X.X.X is the Freeswitch IP
>
>
>
>This script is working normally as long as the customer is sending the
>requests(INVITE,BYE,ACK,CANCEL..) . But when the supplier or freeswitch
>initiated the request , above script causes loop where the packet is
>bounced between kamailio and freeswitch.
>
>
>
>As I know , loose_route is used to handle such requests , but I'm not
>sure how to forward these packets to the customer after receiving them
>from freeswitch.
>
>
>
>I'm not sure also if what's I'm doing is statefull or stateless.
>
>
>
>Any help is appreciated.
>
>
>
>Thanks,
>
>Ali Taher
-- Alex
--
Sent via mobile, please forgive typos and brevity.
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Re: [SR-Users] in-dialog requests
Hi Alex,
How can I troubleshoot the issue in this case ?
Noting that I'm using Kamailio 4.2.0
Thanks
Ali Taher
-Original Message-
From: sr-users On Behalf Of Alex Balashov
Sent: Wednesday, July 18, 2018 4:50 PM
To: Kamailio (SER) - Users Mailing List
Subject: Re: [SR-Users] in-dialog requests
That may be, but it's still the correct config to use. Your scaled-down version
simply will not work.
On July 18, 2018 9:47:15 AM EDT, Ali Taher wrote:
>Hi Alex,
>
>Yes I did, but I noticed that all in-dialog requests sent by freeswitch
>to kamailio were ignored and no packet was generated from kamailio in
>this regard.
>
>I'm not sure if t_relay is working as expected at my side.
>
>Thanks.
>Ali Taher
>
>-Original Message-
>From: sr-users On Behalf Of Alex
>Balashov
>Sent: Wednesday, July 18, 2018 4:34 PM
>To: Kamailio (SER) - Users Mailing List
>Subject: Re: [SR-Users] in-dialog requests
>
>Have you tried the config I suggested to you yesterday?
>
>On July 18, 2018 9:32:01 AM EDT, Ali Taher wrote:
>>Hello,
>>
>>
>>
>>I'm using Kamailio as a proxy behind freeswitch to alter some packets
>>(183
>>packets precisely).
>>
>>
>>
>>Here is my scenario:
>>
>>
>>
>>(---Customer---) --> ((port 5065)---Kamailio---) -->
>>((port5060)---Freeswitch---) --> (---Supplier---)
>>
>>
>>
>>Here is my request route :
>>
>>
>>
>>route{
>>
>>
>>
>>if (is_method("INVITE")) {
>>
>>xlog("L_INFO","INSIDE request route $si \n");
>>
>> insert_hf("X-AUTH-IP: $si\r\n");
>>
>>}
>>
>>
>>
>>route(FWD);
>>
>>}
>>
>>
>>
>>
>>
>>reply_route {
>>
>>#xlog("L_INFO","INSDE reply \n");
>>
>>if (status == "183") {
>>
>>change_reply_status(180, "Ringing");
>>
>>exit;
>>
>>}
>>
>>}
>>
>>
>>
>>route[FWD] {
>>
>>#xlog("L_INFO","INSIDE FWD \n");
>>
>>forward(X.X.X.X,5060);
>>
>>}
>>
>>Where X.X.X.X is the Freeswitch IP
>>
>>
>>
>>This script is working normally as long as the customer is sending the
>>requests(INVITE,BYE,ACK,CANCEL..) . But when the supplier or
>freeswitch
>>initiated the request , above script causes loop where the packet is
>>bounced between kamailio and freeswitch.
>>
>>
>>
>>As I know , loose_route is used to handle such requests , but I'm not
>>sure how to forward these packets to the customer after receiving them
>
>>from freeswitch.
>>
>>
>>
>>I'm not sure also if what's I'm doing is statefull or stateless.
>>
>>
>>
>>Any help is appreciated.
>>
>>
>>
>>Thanks,
>>
>>Ali Taher
>
>
>-- Alex
>
>--
>Sent via mobile, please forgive typos and brevity.
>
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>
>
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-- Alex
--
Sent via mobile, please forgive typos and brevity.
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[SR-Users] NATHELPER issue
Hello,
I'm using Kamailio 4.2 as proxy with nathelper enabled.
Yet , the ACK packet sent from the proxy to the origination's private IP.
The ACK is sent as reply on the following 200 OK sent from the origination :
8m2EJN41BN/6WSIP/2.0 200 OK
From: ;tag=XQBQNjvjgp4Ze
To: ;tag=12033368836000
Via: SIP/2.0/UDP
X.X.X.X:5065;branch=z9hG4bK2959.233ecbc5eff949f946d8763ce25e5e6d.0;received=
X.X.X.X,SIP/2.0/UDP
X.X.X.X;received=X.X.X.X;rport=5060;branch=z9hG4bKN93cXvv26vDDN
Record-Route:
Call-ID: CbeX8453909200habfGhEfElPce@BC00.XX
CSeq: 125698370 INVITE
Accept: application/sdp
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,UPDATE
P-Charging-Vector:
icid-value=B0912C3D70-0720-09394507;icid-generated-at=BC00.XX.XX
;orig-ioi=MXX
Content-Type: application/sdp
Contact:
Content-Length: 268
v=0
o=- 5838243 5838244 IN IP4 BC00.XX
s=-
c=IN IP4 172.16.45.144
t=0 0
a=sendrecv
m=audio 47588 RTP/AVP 18 96
c=IN IP4 172.16.45.144
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=maxptime:20
Following is the header of the sent ACK packet:
Request-Line: ACK sip:172.16.45.65:5060;transport=UDP SIP/2.0
Record-Route:
Via: SIP/2.0/UDP
X.X.X.X:5065;branch=z9hG4bK2959.871535fd341bbe3099d0bf60d6460e18.0
Via: SIP/2.0/UDP
X.X.X.X;received=X.X.X.X;rport=5060;branch=z9hG4bKQUpy0jy90etjc
Max-Forwards: 69
From: ;tag=XQBQNjvjgp4Ze
To: ;tag=12033368836000
Call-ID: CbeX8453909200habfGhEfElPce@BC00.XX
CSeq: 125698370 ACK
Content-Length: 0
Where X.X.X.X is Kamailio server public IP.
Following is part of my config file :
route {
route(NATDETECT);
record_route();
if(!mf_process_maxfwd_header("10")) {
sl_send_reply("483", "Too Many Hops");
exit;
}
# Maybe some sanity_check() here.
if(has_totag()) {
if(loose_route()) {
route(DLGURI);
if(!t_relay())
sl_reply_error();
exit;
} else {
if(is_method("ACK")) {
route(DLGURI);
if(t_check_trans()) {
t_relay();
}
} else
sl_send_reply("403", "Forbidden");
}
exit;
}
.
}
route[NATDETECT] {
#!ifdef WITH_NAT
force_rport();
if (nat_uac_test("19")) {
if (is_method("REGISTER")) {
fix_nated_register();
} else {
add_contact_alias();
}
setflag(FLT_NATS);
}
#!endif
return;
}
route[DLGURI] {
#!ifdef WITH_NAT
if(!isdsturiset()) {
handle_ruri_alias();
}
#!endif
return;
}
Can you please check why the ACK is still sent on private IP ?
Thanks
Ali Taher
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Re: [SR-Users] NATHELPER issue
Thank you for your reply.
ACK is now sent correctly to public IP , yet I have another issue with RBT
where it’s still sent to private IP .Yet the call media is sent correctly to
the public IP.
There is something strange , as I hear two short different ringing tones then
complete silence till the called party answer the call and RTP flow normally.
How can I handle this case?
Thanks,
Ali
From: Konstantin Polyakov
Sent: Friday, July 20, 2018 3:01 PM
To: sr-users@lists.kamailio.org; Ali Taher
Subject: Re: [SR-Users] NATHELPER issue
Hello Ali,
ACK is sent by UAC to the Contact which is received in OK from UAS.
So from my point of view you need to fix that Contact in OK on your proxy.
My NATDETECT looks like:
if (nat_uac_test("19")) {
fix_nated_contact();
}
I call it for requests and responses.
Best regards.
Konstantin
Message: 23
Date: Fri, 20 Jul 2018 11:13:42 +0300
From: "Ali Taher" mailto:ata...@vanrise.com> >
To: mailto:sr-us...@lists.sip-router.org> >
Subject: [SR-Users] NATHELPER issue
Message-ID: <070901d42001$939be2b0$bad3a810$@vanrise.com
<mailto:070901d42001$939be2b0$bad3a810$@vanrise.com> >
Content-Type: text/plain; charset="utf-8"
Hello,
I'm using Kamailio 4.2 as proxy with nathelper enabled.
Yet , the ACK packet sent from the proxy to the origination's private IP.
The ACK is sent as reply on the following 200 OK sent from the origination :
8m2EJN41BN/6WSIP/2.0 200 OK
From: ;tag=XQBQNjvjgp4Ze
To: ;tag=12033368836000
Via: SIP/2.0/UDP
X.X.X.X:5065;branch=z9hG4bK2959.233ecbc5eff949f946d8763ce25e5e6d.0;received=
X.X.X.X,SIP/2.0/UDP
X.X.X.X;received=X.X.X.X;rport=5060;branch=z9hG4bKN93cXvv26vDDN
Record-Route:
Call-ID: CbeX8453909200habfGhEfElPce@BC00.XX
<mailto:CbeX8453909200habfGhEfElPce@BC00.XX>
CSeq: 125698370 INVITE
Accept: application/sdp
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,UPDATE
P-Charging-Vector:
icid-value=B0912C3D70-0720-09394507;icid-generated-at=BC00.XX.XX
;orig-ioi=MXX
Content-Type: application/sdp
Contact:
Content-Length: 268
v=0
o=- 5838243 5838244 IN IP4 BC00.XX
s=-
c=IN IP4 172.16.45.144
t=0 0
a=sendrecv
m=audio 47588 RTP/AVP 18 96
c=IN IP4 172.16.45.144
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=maxptime:20
Following is the header of the sent ACK packet:
Request-Line: ACK sip:172.16.45.65:5060;transport=UDP SIP/2.0
Record-Route:
Via: SIP/2.0/UDP
X.X.X.X:5065;branch=z9hG4bK2959.871535fd341bbe3099d0bf60d6460e18.0
Via: SIP/2.0/UDP
X.X.X.X;received=X.X.X.X;rport=5060;branch=z9hG4bKQUpy0jy90etjc
Max-Forwards: 69
From: ;tag=XQBQNjvjgp4Ze
To: ;tag=12033368836000
Call-ID: CbeX8453909200habfGhEfElPce@BC00.XX
<mailto:CbeX8453909200habfGhEfElPce@BC00.XX>
CSeq: 125698370 ACK
Content-Length: 0
Where X.X.X.X is Kamailio server public IP.
Following is part of my config file :
route {
route(NATDETECT);
record_route();
if(!mf_process_maxfwd_header("10")) {
sl_send_reply("483", "Too Many Hops");
exit;
}
# Maybe some sanity_check() here.
if(has_totag()) {
if(loose_route()) {
route(DLGURI);
if(!t_relay())
sl_reply_error();
exit;
} else {
if(is_method("ACK")) {
route(DLGURI);
if(t_check_trans()) {
t_relay();
}
} else
sl_send_reply("403", "Forbidden");
}
exit;
}
.
}
route[NATDETECT] {
#!ifdef WITH_NAT
force_rport();
if (nat_uac_test("19")) {
if (is_method("REGISTER")) {
fix_nated_register();
} else {
add_contact_alias();
}
setflag(FLT_NATS);
}
#!endif
return;
}
route[DLGURI] {
#!ifdef WITH_NAT
if(!isdsturiset()) {
handle_ruri_alias();
}
#!endif
return;
}
Can you please check why the ACK is still sent on private IP ?
Thanks
Ali Taher
"Ali Taher" https://e.mail.ru/compose?To=ata...@vanrise.com> >
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Re: [SR-Users] NATHELPER issue
Hi Konstantin,
INVITE:
INVITE sip:9613159370@X.X.X.X:5065 SIP/2.0
Record-Route:
X-AUTH-IP: Y.Y.Y.Y
Via: SIP/2.0/UDP
X.X.X.X:5065;branch=z9hG4bK0f8e.68a6417bb4b72bec0eb7c44427dc8731.0
Via: SIP/2.0/UDP
192.168.26.3:31556;received=Y.Y.Y.Y;branch=z9hG4bK-d87543-6a165215a2441724-1--d87543-;rport=31556
Max-Forwards: 69
Contact:
To: "9613159370"
From: "+967123456";tag=5269152d
Call-ID: c10f30782b42be0fNjNiNTdjNGM1MWIzNDlhNTM4MzAwYWI0NWY2NGRmOTM.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE,
INFO
Content-Type: application/sdp
User-Agent: eyeBeam release 1003s stamp 31159
Content-Length: 434
v=0
o=- 5 2 IN IP4 192.168.26.3
s=CounterPath eyeBeam 1.5
c=IN IP4 192.168.26.3
t=0 0
m=audio 8602 RTP/AVP 0 8 101
a=alt:1 4 : F4Vpy7HX BRVjd5rl 192.168.26.3 8602
a=alt:2 3 : ZTQ/VQpO 5uDGgHMJ 192.168.164.1 8602
a=alt:3 2 : ZZPaUsAW LVjbM9qy 192.168.74.1 8602
a=alt:4 1 : 5lOKe54U yTwyWfpC 11.11.0.122 8602
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv
a=x-rtp-session-id:0CDBBC1F22564B72BDA5AD0B07C2AECF
200 OK
200 OK
Via: SIP/2.0/UDP
192.168.26.3:31556;received=Y.Y.Y.Y;branch=z9hG4bK-d87543-6a165215a2441724-1--d87543-;rport=31556
Record-Route:
From: "+967123456" ;tag=5269152d
To: "9613159370" ;tag=agtHmrp3Dccrg
Call-ID: c10f30782b42be0fNjNiNTdjNGM1MWIzNDlhNTM4MzAwYWI0NWY2NGRmOTM.
CSeq: 1 INVITE
Contact:
User-Agent: FIKAR
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog,
line-seize, call-info, sla, include-session-description, presence.winfo,
message-summary, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 200
Remote-Party-ID: "Outbound Call"
;party=calling;privacy=off;screen=no
v=0
o=FIKAR 4050780586 4050780588 IN IP4 X.X.X.X
s=FIKAR
c=IN IP4 X.X.X.X
t=0 0
m=audio 24496 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
Regards,
Ali
From: Konstantin Polyakov
Sent: Monday, July 23, 2018 12:08 PM
To: Ali Taher
Subject: Re[2]: [SR-Users] NATHELPER issue
Hello Ali,
Could you show your latest INVITE and OK?
Check you SDP bodies in those messages, if you still have there private IPs
then you need to solve usual NAT traversal task in your media software (not
proxy).
Kamailio doesn't process media as you know.
Of course you can try fix_nated_sdp function from nathelper module but it will
not work for symmetric NAT because you fix SDP with public IP toward proxy but
not toward UA.
Your media software or hardware should support STUN or ICE protocols to work
correctly via NAT.
Best regards.
Понедельник, 23 июля 2018, 11:41 +03:00 от Ali Taher mailto:ata...@vanrise.com> >:
Thank you for your reply.
ACK is now sent correctly to public IP , yet I have another issue with RBT
where it’s still sent to private IP .Yet the call media is sent correctly to
the public IP.
There is something strange , as I hear two short different ringing tones then
complete silence till the called party answer the call and RTP flow normally.
How can I handle this case?
Thanks,
Ali
From: Konstantin Polyakov mailto:piligrim...@mail.ru> >
Sent: Friday, July 20, 2018 3:01 PM
To: sr-users@lists.kamailio.org <mailto:sr-users@lists.kamailio.org> ; Ali
Taher mailto:ata...@vanrise.com> >
Subject: Re: [SR-Users] NATHELPER issue
Hello Ali,
ACK is sent by UAC to the Contact which is received in OK from UAS.
So from my point of view you need to fix that Contact in OK on your proxy.
My NATDETECT looks like:
if (nat_uac_test("19")) {
fix_nated_contact();
}
I call it for requests and responses.
Best regards.
Konstantin
Message: 23
Date: Fri, 20 Jul 2018 11:13:42 +0300
From: "Ali Taher" >
To:
>
Subject: [SR-Users] NATHELPER issue
Message-ID: <070901d42001$939be2b0$bad3a810$@vanrise.com
>
Content-Type: text/plain; charset="utf-8"
Hello,
I'm using Kamailio 4.2 as proxy with nathelper enabled.
Yet , the ACK packet sent from the proxy to the origination's private IP.
The ACK is sent as reply on the following 200 OK sent from the origination :
8m2EJN41BN/6WSIP/2.0 200 OK
From: ;tag=XQBQNjvjgp4Ze
To: ;tag=12033368836000
Via: SIP/2.0/UDP
X.X.X.X:5065;branch=z9hG4bK2959.233ecbc5eff949f946d8763ce25e5e6d.0;received=
X.X.X.X,SIP/2.0/UDP
X.X.X.X;received=X.X.X.X;rport=5060;branch=z9hG4bKN93cXvv26vDDN
Record-Route:
Call-ID: CbeX8453909200habfGhEfElPce@BC00.XX
CSeq: 125698370 INVITE
Accept: application/sdp
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,UPDATE
P-Charging-Vector:
icid-value=B0912C3D70-0720-09394507;icid-generated-at=BC00.XX.X
Re: [SR-Users] NATHELPER issue
Hi Konstantin,
INVITE:
INVITE sip:9613159370@X.X.X.X:5065 SIP/2.0
Record-Route:
X-AUTH-IP: Y.Y.Y.Y
Via: SIP/2.0/UDP
X.X.X.X:5065;branch=z9hG4bK0f8e.68a6417bb4b72bec0eb7c44427dc8731.0
Via: SIP/2.0/UDP
192.168.26.3:31556;received=Y.Y.Y.Y;branch=z9hG4bK-d87543-6a165215a2441724-1--d87543-;rport=31556
Max-Forwards: 69
Contact:
To: "9613159370"
From: "+967123456";tag=5269152d
Call-ID: c10f30782b42be0fNjNiNTdjNGM1MWIzNDlhNTM4MzAwYWI0NWY2NGRmOTM.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE,
INFO
Content-Type: application/sdp
User-Agent: eyeBeam release 1003s stamp 31159
Content-Length: 434
v=0
o=- 5 2 IN IP4 192.168.26.3
s=CounterPath eyeBeam 1.5
c=IN IP4 192.168.26.3
t=0 0
m=audio 8602 RTP/AVP 0 8 101
a=alt:1 4 : F4Vpy7HX BRVjd5rl 192.168.26.3 8602
a=alt:2 3 : ZTQ/VQpO 5uDGgHMJ 192.168.164.1 8602
a=alt:3 2 : ZZPaUsAW LVjbM9qy 192.168.74.1 8602
a=alt:4 1 : 5lOKe54U yTwyWfpC 11.11.0.122 8602
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv
a=x-rtp-session-id:0CDBBC1F22564B72BDA5AD0B07C2AECF
200 OK
200 OK
Via: SIP/2.0/UDP
192.168.26.3:31556;received=Y.Y.Y.Y;branch=z9hG4bK-d87543-6a165215a2441724-1--d87543-;rport=31556
Record-Route:
From: "+967123456" ;tag=5269152d
To: "9613159370" ;tag=agtHmrp3Dccrg
Call-ID: c10f30782b42be0fNjNiNTdjNGM1MWIzNDlhNTM4MzAwYWI0NWY2NGRmOTM.
CSeq: 1 INVITE
Contact:
User-Agent: FIKAR
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog,
line-seize, call-info, sla, include-session-description, presence.winfo,
message-summary, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 200
Remote-Party-ID: "Outbound Call"
;party=calling;privacy=off;screen=no
v=0
o=FIKAR 4050780586 4050780588 IN IP4 X.X.X.X
s=FIKAR
c=IN IP4 X.X.X.X
t=0 0
m=audio 24496 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
where:
X.X.X.X is Kamailio IP
Y.Y.Y.Y is origination public IP
Regards,
Ali
From: Konstantin Polyakov
Sent: Monday, July 23, 2018 12:08 PM
To: Ali Taher
Subject: Re[2]: [SR-Users] NATHELPER issue
Hello Ali,
Could you show your latest INVITE and OK?
Check you SDP bodies in those messages, if you still have there private IPs
then you need to solve usual NAT traversal task in your media software (not
proxy).
Kamailio doesn't process media as you know.
Of course you can try fix_nated_sdp function from nathelper module but it will
not work for symmetric NAT because you fix SDP with public IP toward proxy but
not toward UA.
Your media software or hardware should support STUN or ICE protocols to work
correctly via NAT.
Best regards.
Понедельник, 23 июля 2018, 11:41 +03:00 от Ali Taher mailto:ata...@vanrise.com> >:
Thank you for your reply.
ACK is now sent correctly to public IP , yet I have another issue with RBT
where it’s still sent to private IP .Yet the call media is sent correctly to
the public IP.
There is something strange , as I hear two short different ringing tones then
complete silence till the called party answer the call and RTP flow normally.
How can I handle this case?
Thanks,
Ali
From: Konstantin Polyakov mailto:piligrim...@mail.ru> >
Sent: Friday, July 20, 2018 3:01 PM
To: sr-users@lists.kamailio.org <mailto:sr-users@lists.kamailio.org> ; Ali
Taher mailto:ata...@vanrise.com> >
Subject: Re: [SR-Users] NATHELPER issue
Hello Ali,
ACK is sent by UAC to the Contact which is received in OK from UAS.
So from my point of view you need to fix that Contact in OK on your proxy.
My NATDETECT looks like:
if (nat_uac_test("19")) {
fix_nated_contact();
}
I call it for requests and responses.
Best regards.
Konstantin
Message: 23
Date: Fri, 20 Jul 2018 11:13:42 +0300
From: "Ali Taher" >
To:
>
Subject: [SR-Users] NATHELPER issue
Message-ID: <070901d42001$939be2b0$bad3a810$@vanrise.com
>
Content-Type: text/plain; charset="utf-8"
Hello,
I'm using Kamailio 4.2 as proxy with nathelper enabled.
Yet , the ACK packet sent from the proxy to the origination's private IP.
The ACK is sent as reply on the following 200 OK sent from the origination :
8m2EJN41BN/6WSIP/2.0 200 OK
From: ;tag=XQBQNjvjgp4Ze
To: ;tag=12033368836000
Via: SIP/2.0/UDP
X.X.X.X:5065;branch=z9hG4bK2959.233ecbc5eff949f946d8763ce25e5e6d.0;received=
X.X.X.X,SIP/2.0/UDP
X.X.X.X;received=X.X.X.X;rport=5060;branch=z9hG4bKN93cXvv26vDDN
Record-Route:
Call-ID: CbeX8453909200habfGhEfElPce@BC00.XX
CSeq: 125698370 INVITE
Accept: application/sdp
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,UPDATE
P-Charging-Vector:
icid-value=
Re: [SR-Users] NATHELPER issue
Thank you Konstantin,
What I can’t understand is why media packets are sent correctly to the public
IP , while early media (RBT) are sent on private.
Regards,
Ali
From: Konstantin Polyakov
Sent: Monday, July 23, 2018 1:38 PM
To: Ali Taher
Subject: Re[4]: [SR-Users] NATHELPER issue
Original INVITE contains private IP in c= parameter, so you need to use STUN
or TURN or ICE, something which is supported in your system or in your
softphone.
See my comments in the previous mail.
Best regrads
Понедельник, 23 июля 2018, 12:25 +03:00 от Ali Taher mailto:ata...@vanrise.com> >:
Hi Konstantin,
INVITE:
INVITE sip:9613159370@X.X.X.X:5065 SIP/2.0
Record-Route:
X-AUTH-IP: Y.Y.Y.Y
Via: SIP/2.0/UDP
X.X.X.X:5065;branch=z9hG4bK0f8e.68a6417bb4b72bec0eb7c44427dc8731.0
Via: SIP/2.0/UDP
192.168.26.3:31556;received=Y.Y.Y.Y;branch=z9hG4bK-d87543-6a165215a2441724-1--d87543-;rport=31556
Max-Forwards: 69
Contact:
To: "9613159370"
From: "+967123456";tag=5269152d
Call-ID: c10f30782b42be0fNjNiNTdjNGM1MWIzNDlhNTM4MzAwYWI0NWY2NGRmOTM.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE,
INFO
Content-Type: application/sdp
User-Agent: eyeBeam release 1003s stamp 31159
Content-Length: 434
v=0
o=- 5 2 IN IP4 192.168.26.3
s=CounterPath eyeBeam 1.5
c=IN IP4 192.168.26.3
t=0 0
m=audio 8602 RTP/AVP 0 8 101
a=alt:1 4 : F4Vpy7HX BRVjd5rl 192.168.26.3 8602
a=alt:2 3 : ZTQ/VQpO 5uDGgHMJ 192.168.164.1 8602
a=alt:3 2 : ZZPaUsAW LVjbM9qy 192.168.74.1 8602
a=alt:4 1 : 5lOKe54U yTwyWfpC 11.11.0.122 8602
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendrecv
a=x-rtp-session-id:0CDBBC1F22564B72BDA5AD0B07C2AECF
200 OK
200 OK
Via: SIP/2.0/UDP
192.168.26.3:31556;received=Y.Y.Y.Y;branch=z9hG4bK-d87543-6a165215a2441724-1--d87543-;rport=31556
Record-Route:
From: "+967123456" ;tag=5269152d
To: "9613159370" ;tag=agtHmrp3Dccrg
Call-ID: c10f30782b42be0fNjNiNTdjNGM1MWIzNDlhNTM4MzAwYWI0NWY2NGRmOTM.
CSeq: 1 INVITE
Contact:
User-Agent: FIKAR
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog,
line-seize, call-info, sla, include-session-description, presence.winfo,
message-summary, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 200
Remote-Party-ID: "Outbound Call" mailto:506251539613159370@X.X.X.X> >;party=calling;privacy=off;screen=no
v=0
o=FIKAR 4050780586 4050780588 IN IP4 X.X.X.X
s=FIKAR
c=IN IP4 X.X.X.X
t=0 0
m=audio 24496 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
where:
X.X.X.X is Kamailio IP
Y.Y.Y.Y is origination public IP
Regards,
Ali
From: Konstantin Polyakov mailto:piligrim...@mail.ru> >
Sent: Monday, July 23, 2018 12:08 PM
To: Ali Taher mailto:ata...@vanrise.com> >
Subject: Re[2]: [SR-Users] NATHELPER issue
Hello Ali,
Could you show your latest INVITE and OK?
Check you SDP bodies in those messages, if you still have there private IPs
then you need to solve usual NAT traversal task in your media software (not
proxy).
Kamailio doesn't process media as you know.
Of course you can try fix_nated_sdp function from nathelper module but it will
not work for symmetric NAT because you fix SDP with public IP toward proxy but
not toward UA.
Your media software or hardware should support STUN or ICE protocols to work
correctly via NAT.
Best regards.
Понедельник, 23 июля 2018, 11:41 +03:00 от Ali Taher >:
Thank you for your reply.
ACK is now sent correctly to public IP , yet I have another issue with RBT
where it’s still sent to private IP .Yet the call media is sent correctly to
the public IP.
There is something strange , as I hear two short different ringing tones then
complete silence till the called party answer the call and RTP flow normally.
How can I handle this case?
Thanks,
Ali
From: Konstantin Polyakov >
Sent: Friday, July 20, 2018 3:01 PM
To: sr-users@lists.kamailio.org
; Ali Taher >
Subject: Re: [SR-Users] NATHELPER issue
Hello Ali,
ACK is sent by UAC to the Contact which is received in OK from UAS.
So from my point of view you need to fix that Contact in OK on your proxy.
My NATDETECT looks like:
if (nat_uac_test("19")) {
fix_nated_contact();
}
I call it for requests and responses.
Best regards.
Konstantin
Message: 23
Date: Fri, 20 Jul 2018 11:13:42 +0300
From: "Ali Taher" mailto:ata...@vanrise.com> >
To: mailto:sr-us...@lists.sip-router.org> >
Subject: [SR-Users] NATHELPER issue
Message-ID: <070901d42001$939be2b0$bad3a810$@vanrise.com
<mailto:070901d42001$939be2b0$bad3a810$@vanrise.com> >
Content-Type: text/plain; charset="utf-8"
[SR-Users] change A number using a pool of numbers
Hello, I'm looking to change the A number (from number) to a number selected randomly from a pool of existing numbers (hundreds) and then relay the call to a certain gateway. What is the best and more efficient way to do it in kamailio? Thanks in advance __ Kamailio - Users Mailing List - Non Commercial Discussions To unsubscribe send an email to sr-users-le...@lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe:
[SR-Users] Re: change A number using a pool of numbers
Thanks Kaufman, Relying on DB is of course one way to do it, but I thought that there are some modules that could be used and are faster, like app_lua but I'm not sure if it can be used to select a number randomly from a pool saved in a file. Regards, From: Kaufman Sent: Wednesday, August 2, 2023 7:36 PM To: Kamailio (SER) - Users Mailing List Subject: [SR-Users] Re: change A number using a pool of numbers There's functionally two parts to this question: 1. How to change a number 2. How to select a random number For selection of the number it's probably easiest to rely on DB lookup. With one column as the original number and another column as the possible replacement numbers, then just select with an order by random to get the replacement number. As far as changing the number, it can depend on specifically what number you're trying to change. The RURI can be changed just by setting $rU. Values in the To: or From: headers should be changed with the functions from the UAC module so that replies can have the original values restored. For other miscellaneous headers like P-Asserted-Identity, just remove the old header and append a new one. Regards, Kaufman From: Ali Taher mailto:ata...@vanrise.com>> Sent: Wednesday, August 2, 2023 1:20 AM To: Kamailio (SER) - Users Mailing List mailto:sr-users@lists.kamailio.org>> Subject: [SR-Users] change A number using a pool of numbers CAUTION: This email originated from outside the organization. Do not click links or open attachments unless you recognize the sender and know the content is safe. Hello, I'm looking to change the A number (from number) to a number selected randomly from a pool of existing numbers (hundreds) and then relay the call to a certain gateway. What is the best and more efficient way to do it in kamailio? Thanks in advance __ Kamailio - Users Mailing List - Non Commercial Discussions To unsubscribe send an email to sr-users-le...@lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe:
[SR-Users] Modify To URI in Multiple Choices
Hello,
I'm using Kamailio as SIP redirect where SBC sends INVITE packet to kamailio
and "302","Multiple Choices" is sent back to SBC.
What I want is to modify the URI part in the To header of the Multiple Choices
packet sent from kamailio. I tried to use
uac_replace_to("","sip:$var(cdpn)@$td") where $var(cdpn) is already defined in
the routing logic, but it's not reflected and the original TO URI is sent
instead.
Any help to handle this issue is much appreciated.
Regards,
__
Kamailio - Users Mailing List - Non Commercial Discussions
To unsubscribe send an email to sr-users-le...@lists.kamailio.org
Important: keep the mailing list in the recipients, do not reply only to the
sender!
Edit mailing list options or unsubscribe:
[SR-Users] Re: Modify To URI in Multiple Choices
I used append_to_reply function to rebuild the To header, but the problem is
that the old To header stays there even if I use remove_hf method to remove it.
What I’m doing wrong here?
My logic is set inside a route block.
Regards,
Sent from my iPhone
On Aug 9, 2023, at 2:55 PM, Ali Taher wrote:
Hello,
I’m using Kamailio as SIP redirect where SBC sends INVITE packet to kamailio
and "302","Multiple Choices" is sent back to SBC.
What I want is to modify the URI part in the To header of the Multiple Choices
packet sent from kamailio. I tried to use
uac_replace_to("”,"sip:$var(cdpn)@$td") where $var(cdpn) is already defined in
the routing logic, but it’s not reflected and the original TO URI is sent
instead.
Any help to handle this issue is much appreciated.
Regards,
__
Kamailio - Users Mailing List - Non Commercial Discussions
To unsubscribe send an email to sr-users-le...@lists.kamailio.org
Important: keep the mailing list in the recipients, do not reply only to the
sender!
Edit mailing list options or unsubscribe:
[SR-Users] kamailio-postgres on RHEL 9
Hello, I'm trying to install latest Kamailio version on RHEL 9.2. All is good, but I couldn't find kamailio-postgres package related to postgres module, knowing that I have installed Postgresql 15. Is there a rpm file that I can download to install kamailio-postgres ? Regards, __ Kamailio - Users Mailing List - Non Commercial Discussions To unsubscribe send an email to sr-users-le...@lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe:
[SR-Users] Set a License for Kamailio
Hello, Is it possible to put a kind of license for Kamailio, in a way that it works for 1 week for example, and after that it will not process new requests? Regards, __ Kamailio - Users Mailing List - Non Commercial Discussions To unsubscribe send an email to sr-users-le...@lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe:
[SR-Users] Re: Set a License for Kamailio
Hi Alex, Can you please put me on track how to proceed? I'm not familiar with JWT. Regards, -Original Message- From: Alex Balashov Sent: Tuesday, September 5, 2023 4:38 PM To: Kamailio (SER) - Users Mailing List Subject: [SR-Users] Re: Set a License for Kamailio A JWT would be a reasonably stateless way to do that. > On Sep 5, 2023, at 8:45 AM, Ali Taher wrote: > > Hello, > Is it possible to put a kind of license for Kamailio, in a way that it works > for 1 week for example, and after that it will not process new requests? > Regards, > __ > Kamailio - Users Mailing List - Non Commercial Discussions To > unsubscribe send an email to sr-users-le...@lists.kamailio.org > Important: keep the mailing list in the recipients, do not reply only to the > sender! > Edit mailing list options or unsubscribe: -- Alex Balashov Principal Consultant Evariste Systems LLC Web: https://evaristesys.com Tel: +1-706-510-6800 __ Kamailio - Users Mailing List - Non Commercial Discussions To unsubscribe send an email to sr-users-le...@lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe: __ Kamailio - Users Mailing List - Non Commercial Discussions To unsubscribe send an email to sr-users-le...@lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe:
[SR-Users] 0.0.0.0 in VIA header
Hello, I'm using Kamailio as SIP proxy to receive INVITE packets, randomize A-number and then relay it to a certain gateway. I noticed that a Via header is added to the INVITE sent by Kamailio to the gateway with 0.0.0.0 as address as shown below: Via: SIP/2.0/UDP 0.0.0.0;branch=z9hG4bK2d97.61de75dd58fa7ca7450cc8b9b01db62f.0 As far as I know, the IP that should be used in the new added VIA header must be Kamailio IP, but it's not the case here, knowing that Kamailio is not behind NAT. Regards, Ali Taher __ Kamailio - Users Mailing List - Non Commercial Discussions To unsubscribe send an email to sr-users-le...@lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe:
[SR-Users] Kamailio as SIP proxy
Hello,
I'm using Kamailio as SIP proxy between Session Manager and SBC
So here is the scenario :
Session Manager sends a call (Invite) to Kamailio, kamailio needs to relay it
to SBC after doing some modification.
(-Session Manager-) --- > (Kamailio) --- >
(-SBC-)
My routing logic is similar to the below:
route{
if ((method==OPTIONS) && (! uri=~"sip:.*[@]+.*")) {
options_reply();
}
if (is_method("INVITE")) {
sql_query("cb", "select number from pool1 order by random() limit 1", "ra");
$var(rand)=$dbr(ra=>[0,0]);
uac_replace_from($var(rand),"sip:$var(rand)@192.168.1.1");
xlog("L_INFO","Random: $var(rand)");
$td = "192.168.1.10";
$rd = "192.168.1.10";
t_relay();
}
}
Where:
192.168.1.1 is Kamailio IP
192.168.1.10 is SBC IP
The issue is that I'm not sure to handle the other SIP packets like 183,CANCEL,
BYE etc that can be received from both SBC and session Manager?
What to add to my configuration to handle it properly?
Regards,
__
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[SR-Users] 183 Session Progress Acknowledgment
Hello, Is there a way to send PRACK as reply to a 183 Session Progress (with Require: 100rel) sent to Kamailio sip proxy? Regards, __ Kamailio - Users Mailing List - Non Commercial Discussions To unsubscribe send an email to sr-users-le...@lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe:
[SR-Users] Re: 183 Session Progress Acknowledgment
Hi Alex, That means that kamailio should forward the 183 session progress received from UAS as it to the UAC, which in their turn must send PRACK and kamailio forwards it back to UAS? > On Oct 13, 2023, at 4:29 PM, Alex Balashov via sr-users > wrote: > > PRACKs are up to the UAs to send. > >> On 13 Oct 2023, at 08:30, Ali Taher via sr-users >> wrote: >> >> Hello, >> Is there a way to send PRACK as reply to a 183 Session Progress (with >> Require: 100rel) sent to Kamailio sip proxy? >> Regards, >> __ >> Kamailio - Users Mailing List - Non Commercial Discussions >> To unsubscribe send an email to sr-users-le...@lists.kamailio.org >> Important: keep the mailing list in the recipients, do not reply only to the >> sender! >> Edit mailing list options or unsubscribe: > > > -- > Alex Balashov > Principal Consultant > Evariste Systems LLC > Web: https://evaristesys.com > Tel: +1-706-510-6800 > > __ > Kamailio - Users Mailing List - Non Commercial Discussions > To unsubscribe send an email to sr-users-le...@lists.kamailio.org > Important: keep the mailing list in the recipients, do not reply only to the > sender! > Edit mailing list options or unsubscribe: __ Kamailio - Users Mailing List - Non Commercial Discussions To unsubscribe send an email to sr-users-le...@lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe:
[SR-Users] Kamailio SIP proxy role
Hello,
I'm using Kamailio as SIP proxy between Session Manager and SBC
So here is the scenario :
Session Manager sends a call (Invite) to Kamailio, kamailio needs to relay it
to SBC after doing some modification.
(-Session Manager-) --- > (Kamailio) --- >
(-SBC-)
My routing logic is similar to the below:
route{
if ((method==OPTIONS) && (! uri=~"sip:.*[@]+.*")) {
options_reply();
}
if (is_method("INVITE")) {
sql_query("cb", "select number from pool1 order by random() limit 1", "ra");
$var(rand)=$dbr(ra=>[0,0]);
uac_replace_from($var(rand),"sip:$var(rand)@192.168.1.1");
xlog("L_INFO","Random: $var(rand)");
$var(ip) = $(ct{s.select,1,@}{s.select,0,;}{s.replace,>,});
remove_hf("Contact");
insert_hf("Contact: $var(rand) \r\n");
remove_hf("P-Asserted-Identity");
remove_hf("Route");
insert_hf("Route: \r\n","Route");
record_route();
t_relay();
$td = "192.168.1.10";
$rd = "192.168.1.10";
t_relay();
}
}
Where:
192.168.1.1 is Kamailio IP
192.168.1.10 is SBC IP
My question here is that Kamailio handle all in dialog messages between Session
Manager and SBC?
For example, after forwarding the invite packet to SBC, SBC sends 183 Session
Progress to Kamailio, which in its turn relay it to the session manager,
knowing that this is not mentioned in my routing logic, so how Kamailio knew
that it must relay it to session manager in this case?
Also I need to add to the 183 Session Progress sent to session manager by
Kamailio record route and Via headers , how to do it ?
Regards,
__
Kamailio - Users Mailing List - Non Commercial Discussions
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Important: keep the mailing list in the recipients, do not reply only to the
sender!
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[SR-Users] SIP Proxy : BYE next hop
Hello all, I'm using Kamailio as SIP proxy between session manager and SBC. I came across an issue in BYE request where the SBC sent in the header list of Routes where first route is Kamailio IP, and the second is SM IP, in addition to other routes (last route is the caller party) What Kamailio did is to route the BYE to the address set in the Request-URI, which is the caller party. What should be done instead is that Kamailio should remove the Route header including its own IP, and route the BYE to the next hop which is in this case the SM IP. How to accomplish that? Following is the Route header in the BYE request coming from SBC: [truncated]Route: ,,,sips:127.0.0.2:15061;ibmsid=local.xx_xx_;lr;ibmdrr, sip:SM1@SM_IP;transport=tls;lr;av-asset-uid=xx, sip:CALLER_PARTY_IP:5061;transport=tls;lr Regards, __ Kamailio - Users Mailing List - Non Commercial Discussions To unsubscribe send an email to sr-users-le...@lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe:
[SR-Users] Re: SIP Proxy : BYE next hop
Hi James,
Yes I'm using loose_route() and it's returning 1, but BYE message is not routed
anywhere.
Frame 6213: 673 bytes on wire (5384 bits), 673 bytes captured (5384 bits)
Ethernet II, Src: Cisco_f8:19:ff (00:22:bd:f8:19:ff), Dst: VMware_bf:27:0b
(00:50:56:bf:27:0b)
Internet Protocol Version 4, Src: 192.168.1.5, Dst: 192.168.1.2
User Datagram Protocol, Src Port: 16368, Dst Port: 5060
Session Initiation Protocol (BYE)
Request-Line: BYE sip:192.168.1.1:5060;transport=udp SIP/2.0
Message Header
Via: SIP/2.0/UDP
192.168.1.5:16368;branch=z9hG4bK-d87543-c147891e677a1f19-1--d87543-;rport
Max-Forwards: 70
Route:
Route:
Contact:
To: "121";tag=gtyrzghr-CC-1047-OFC-39
From: "69079514";tag=7d4a7c3c
Call-ID: 085d9048b81abc1eNDUwODVhMjI2NDk5NDM3YmZiZWZiYzI0YWZkOTU1Njc.
[Generated Call-ID:
085d9048b81abc1eNDUwODVhMjI2NDk5NDM3YmZiZWZiYzI0YWZkOTU1Njc.]
CSeq: 2 BYE
User-Agent: eyeBeam release 1003s stamp 31159
Reason: SIP;description="User Hung Up"
Content-Length: 0
Here is routing block related to BYE:
if (is_method("BYE")) {
xlog("L_INFO","INSIDE BYE");
$var(loose) = loose_route();
xlog("L_INFO","LOOSE : $var(loose)");
}
Regards,
-Original Message-
From: James Browne
Sent: Thursday, November 2, 2023 12:31 PM
To: Kamailio (SER) - Users Mailing List
Cc: Ali Taher
Subject: Re: [SR-Users] SIP Proxy : BYE next hop
Hello
Did you call the loose_route() function for the BYE? It should just work;
that's one of the basics of how kamailio works, so I'll be very surprised if
there's somehow a bug in kamailio.
If that does not work, perhaps send a list of the listen directives in the
kamailio.cfg, along with the headers of the two SIP BYE messages (incoming and
outgoing) and I'll see what I can figure out for you.
James
On Thu, 2 Nov 2023 at 09:35, Ali Taher via sr-users
wrote:
>
> Hello all,
>
>
>
> I’m using Kamailio as SIP proxy between session manager and SBC.
>
> I came across an issue in BYE request where the SBC sent in the header
> list of Routes where first route is Kamailio IP, and the second is SM
> IP, in addition to other routes (last route is the caller party)
>
> What Kamailio did is to route the BYE to the address set in the Request-URI,
> which is the caller party.
>
> What should be done instead is that Kamailio should remove the Route header
> including its own IP, and route the BYE to the next hop which is in this case
> the SM IP.
>
>
>
> How to accomplish that?
>
>
>
> Following is the Route header in the BYE request coming from SBC:
>
> [truncated]Route:
> , port=udp;lr;av-asset-uid=xx>, p;ibmsid=local._x_x;lr;ibmdrr>,sips:127.0.
> 0.2:15061;ibmsid=local.xx_xx_;lr;ibmdrr,
> sip:SM1@SM_IP;transport=tls;lr;av-asset-uid=xx,
> sip:CALLER_PARTY_IP:5061;transport=tls;lr
>
>
>
> Regards,
>
> __
> Kamailio - Users Mailing List - Non Commercial Discussions To
> unsubscribe send an email to sr-users-le...@lists.kamailio.org
> Important: keep the mailing list in the recipients, do not reply only to the
> sender!
> Edit mailing list options or unsubscribe:
__
Kamailio - Users Mailing List - Non Commercial Discussions
To unsubscribe send an email to sr-users-le...@lists.kamailio.org
Important: keep the mailing list in the recipients, do not reply only to the
sender!
Edit mailing list options or unsubscribe:
[SR-Users] Re: SIP Proxy : BYE next hop
Hi James,
Here are my listen directives:
listen=udp:192.168.1.2:5060
listen=tcp: 192.168.1.2:5060
knowing that:
Kamailio IP: 192.168.1.2
SBC IP: 192.168.1.1
Caller IP: 192.168.1.5
-Original Message-
From: Ali Taher via sr-users
Sent: Thursday, November 2, 2023 1:42 PM
To: James Browne ; Kamailio (SER) - Users Mailing List
Cc: Ali Taher
Subject: [SR-Users] Re: SIP Proxy : BYE next hop
Hi James,
Yes I'm using loose_route() and it's returning 1, but BYE message is not routed
anywhere.
Frame 6213: 673 bytes on wire (5384 bits), 673 bytes captured (5384 bits)
Ethernet II, Src: Cisco_f8:19:ff (00:22:bd:f8:19:ff), Dst: VMware_bf:27:0b
(00:50:56:bf:27:0b) Internet Protocol Version 4, Src: 192.168.1.5, Dst:
192.168.1.2 User Datagram Protocol, Src Port: 16368, Dst Port: 5060 Session
Initiation Protocol (BYE)
Request-Line: BYE sip:192.168.1.1:5060;transport=udp SIP/2.0
Message Header
Via: SIP/2.0/UDP
192.168.1.5:16368;branch=z9hG4bK-d87543-c147891e677a1f19-1--d87543-;rport
Max-Forwards: 70
Route:
Route:
Contact:
To: "121";tag=gtyrzghr-CC-1047-OFC-39
From: "69079514";tag=7d4a7c3c
Call-ID: 085d9048b81abc1eNDUwODVhMjI2NDk5NDM3YmZiZWZiYzI0YWZkOTU1Njc.
[Generated Call-ID:
085d9048b81abc1eNDUwODVhMjI2NDk5NDM3YmZiZWZiYzI0YWZkOTU1Njc.]
CSeq: 2 BYE
User-Agent: eyeBeam release 1003s stamp 31159
Reason: SIP;description="User Hung Up"
Content-Length: 0
Here is routing block related to BYE:
if (is_method("BYE")) {
xlog("L_INFO","INSIDE BYE");
$var(loose) = loose_route();
xlog("L_INFO","LOOSE : $var(loose)");
}
Regards,
-Original Message-
From: James Browne
Sent: Thursday, November 2, 2023 12:31 PM
To: Kamailio (SER) - Users Mailing List
Cc: Ali Taher
Subject: Re: [SR-Users] SIP Proxy : BYE next hop
Hello
Did you call the loose_route() function for the BYE? It should just work;
that's one of the basics of how kamailio works, so I'll be very surprised if
there's somehow a bug in kamailio.
If that does not work, perhaps send a list of the listen directives in the
kamailio.cfg, along with the headers of the two SIP BYE messages (incoming and
outgoing) and I'll see what I can figure out for you.
James
On Thu, 2 Nov 2023 at 09:35, Ali Taher via sr-users
wrote:
>
> Hello all,
>
>
>
> I’m using Kamailio as SIP proxy between session manager and SBC.
>
> I came across an issue in BYE request where the SBC sent in the header
> list of Routes where first route is Kamailio IP, and the second is SM
> IP, in addition to other routes (last route is the caller party)
>
> What Kamailio did is to route the BYE to the address set in the Request-URI,
> which is the caller party.
>
> What should be done instead is that Kamailio should remove the Route header
> including its own IP, and route the BYE to the next hop which is in this case
> the SM IP.
>
>
>
> How to accomplish that?
>
>
>
> Following is the Route header in the BYE request coming from SBC:
>
> [truncated]Route:
> , port=udp;lr;av-asset-uid=xx>, p;ibmsid=local._x_x;lr;ibmdrr>,sips:127.0.
> 0.2:15061;ibmsid=local.xx_xx_;lr;ibmdrr,
> sip:SM1@SM_IP;transport=tls;lr;av-asset-uid=xx,
> sip:CALLER_PARTY_IP:5061;transport=tls;lr
>
>
>
> Regards,
>
> __
> Kamailio - Users Mailing List - Non Commercial Discussions To
> unsubscribe send an email to sr-users-le...@lists.kamailio.org
> Important: keep the mailing list in the recipients, do not reply only to the
> sender!
> Edit mailing list options or unsubscribe:
__
Kamailio - Users Mailing List - Non Commercial Discussions
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Important: keep the mailing list in the recipients, do not reply only to the
sender!
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__
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[SR-Users] Re: SIP Proxy : BYE next hop
Hi Yuriy,
Thanks, I used what you suggested below, and still getting 1 as result for
loose_route(), but BYE message is not routed anywhere.
Regards,
From: Yuriy G
Sent: Thursday, November 2, 2023 3:05 PM
To: Kamailio (SER) - Users Mailing List
Cc: James Browne ; Ali Taher
Subject: Re: [SR-Users] Re: SIP Proxy : BYE next hop
loose_route returns result in the $rc variable
Return code should be handed like this:
https://www.kamailio.org/wikidocs/cookbooks/5.6.x/pseudovariables/#rc-returned-code
чт, 2 нояб. 2023 г. в 13:27, Ali Taher via sr-users
mailto:sr-users@lists.kamailio.org>>:
Hi James,
Here are my listen directives:
listen=udp:192.168.1.2:5060<http://192.168.1.2:5060>
listen=tcp: 192.168.1.2:5060<http://192.168.1.2:5060>
knowing that:
Kamailio IP: 192.168.1.2
SBC IP: 192.168.1.1
Caller IP: 192.168.1.5
-Original Message-----
From: Ali Taher via sr-users
mailto:sr-users@lists.kamailio.org>>
Sent: Thursday, November 2, 2023 1:42 PM
To: James Browne mailto:ja...@frideo.com>>; Kamailio (SER) -
Users Mailing List
mailto:sr-users@lists.kamailio.org>>
Cc: Ali Taher mailto:ata...@vanrise.com>>
Subject: [SR-Users] Re: SIP Proxy : BYE next hop
Hi James,
Yes I'm using loose_route() and it's returning 1, but BYE message is not routed
anywhere.
Frame 6213: 673 bytes on wire (5384 bits), 673 bytes captured (5384 bits)
Ethernet II, Src: Cisco_f8:19:ff (00:22:bd:f8:19:ff), Dst: VMware_bf:27:0b
(00:50:56:bf:27:0b) Internet Protocol Version 4, Src: 192.168.1.5, Dst:
192.168.1.2 User Datagram Protocol, Src Port: 16368, Dst Port: 5060 Session
Initiation Protocol (BYE)
Request-Line: BYE
sip:192.168.1.1:5060<http://192.168.1.1:5060>;transport=udp SIP/2.0
Message Header
Via: SIP/2.0/UDP
192.168.1.5:16368;branch=z9hG4bK-d87543-c147891e677a1f19-1--d87543-;rport
Max-Forwards: 70
Route:
Route:
Contact: http://sip:131@192.168.1.5:16368>>
To:
"121"mailto:sip%3A121@192.168.1.1>>;tag=gtyrzghr-CC-1047-OFC-39
From:
"69079514"mailto:sip%3A69079514@192.168.1.2>>;tag=7d4a7c3c
Call-ID: 085d9048b81abc1eNDUwODVhMjI2NDk5NDM3YmZiZWZiYzI0YWZkOTU1Njc.
[Generated Call-ID:
085d9048b81abc1eNDUwODVhMjI2NDk5NDM3YmZiZWZiYzI0YWZkOTU1Njc.]
CSeq: 2 BYE
User-Agent: eyeBeam release 1003s stamp 31159
Reason: SIP;description="User Hung Up"
Content-Length: 0
Here is routing block related to BYE:
if (is_method("BYE")) {
xlog("L_INFO","INSIDE BYE");
$var(loose) = loose_route();
xlog("L_INFO","LOOSE : $var(loose)");
}
Regards,
-Original Message-
From: James Browne mailto:ja...@frideo.com>>
Sent: Thursday, November 2, 2023 12:31 PM
To: Kamailio (SER) - Users Mailing List
mailto:sr-users@lists.kamailio.org>>
Cc: Ali Taher mailto:ata...@vanrise.com>>
Subject: Re: [SR-Users] SIP Proxy : BYE next hop
Hello
Did you call the loose_route() function for the BYE? It should just work;
that's one of the basics of how kamailio works, so I'll be very surprised if
there's somehow a bug in kamailio.
If that does not work, perhaps send a list of the listen directives in the
kamailio.cfg, along with the headers of the two SIP BYE messages (incoming and
outgoing) and I'll see what I can figure out for you.
James
On Thu, 2 Nov 2023 at 09:35, Ali Taher via sr-users
mailto:sr-users@lists.kamailio.org>> wrote:
>
> Hello all,
>
>
>
> I’m using Kamailio as SIP proxy between session manager and SBC.
>
> I came across an issue in BYE request where the SBC sent in the header
> list of Routes where first route is Kamailio IP, and the second is SM
> IP, in addition to other routes (last route is the caller party)
>
> What Kamailio did is to route the BYE to the address set in the Request-URI,
> which is the caller party.
>
> What should be done instead is that Kamailio should remove the Route header
> including its own IP, and route the BYE to the next hop which is in this case
> the SM IP.
>
>
>
> How to accomplish that?
>
>
>
> Following is the Route header in the BYE request coming from SBC:
>
> [truncated]Route:
> ,>
port=udp;lr;av-asset-uid=xx>,http://127.0.0.2>:15060;transport=ud
> p;ibmsid=local._x_x;lr;ibmdrr>,sips:127.0.
> 0.2:15061;ibmsid=local.xx_xx_;lr;ibmdrr,
> sip:SM1@SM_IP;transport=tls;lr;av-asset-uid=xx,
> sip:CALLER_PARTY_IP:5061;transport=tls;lr
>
>
>
> Regards,
>
> __
> Kamailio - Users Mailing List - Non Commercial Discussions To
> unsubscribe send an email to
> sr-users-le...@lists.kamailio.org<mailto:sr-users-le...@lists.kamailio.org&
[SR-Users] Redis issue when getting results in Kamailio
Hello all,
I have redis database where the data is inserted in JSON format:
#Redis Command
JSON.GET Index1J:12345678
#Redis Result
[
{
"SupplierID": "7",
"BCode": "987654",
"Rate": 0.159
},
{
"SupplierID": "7",
"BCode": "765432",
"Rate": 0.159
},
{
"SupplierID": "7",
"BCode": "5432159",
"Rate": 0.159
}
]
When I try to get the result using Lua script in Kamailio like the following:
#kamailio command
if KSR.ndb_redis.redis_cmd("srvN", "JSON.GET pricelistJ:441246890", "ra") then
KSR.xlog.xinfo("array size: $redis(ra=>size)\n");
KSR.xlog.xinfo("Redis Response: " .. tostring(ra) .. "\n")
KSR.xlog.xinfo("first values: $redis(ra=>value[0]) ,
$redis(ra=>value[1])\n");
end
Nov 7 07:58:15 868d08ee318d /usr/sbin/kamailio[72]: INFO: LUA {INVITE}:
[SR-Users] Re: Redis issue when getting results in Kamailio
I'm still stuck... I appreciate any assistance here.
From: Ali Taher via sr-users
Sent: Tuesday, November 7, 2023 12:47 PM
To: Kamailio (SER) - Users Mailing List
Cc: Ali Taher
Subject: [SR-Users] Redis issue when getting results in Kamailio
Hello all,
I have redis database where the data is inserted in JSON format:
#Redis Command
JSON.GET Index1J:12345678
#Redis Result
[
{
"SupplierID": "7",
"BCode": "987654",
"Rate": 0.159
},
{
"SupplierID": "7",
"BCode": "765432",
"Rate": 0.159
},
{
"SupplierID": "7",
"BCode": "5432159",
"Rate": 0.159
}
]
When I try to get the result using Lua script in Kamailio like the following:
#kamailio command
if KSR.ndb_redis.redis_cmd("srvN", "JSON.GET pricelistJ:441246890", "ra") then
KSR.xlog.xinfo("array size: $redis(ra=>size)\n");
KSR.xlog.xinfo("Redis Response: " .. tostring(ra) .. "\n")
KSR.xlog.xinfo("first values: $redis(ra=>value[0]) ,
$redis(ra=>value[1])\n");
end
Nov 7 07:58:15 868d08ee318d /usr/sbin/kamailio[72]: INFO: LUA {INVITE}:
[SR-Users] Re: Redis issue when getting results in Kamailio
Hi Yurig,
Yes, GET/SET is working normally from kamailio if not using JSON.
Honestly, I didn’t get what do you mean by the below. Can you please elaborate
how to fix the issue?
Regards,
From: Yuriy G
Sent: Thursday, November 9, 2023 10:55 AM
To: Kamailio (SER) - Users Mailing List
Cc: Ali Taher
Subject: Re: [SR-Users] Re: Redis issue when getting results in Kamailio
Did you try to use not JSON but usual GET/SET? Do you have the same issue?
AFAIK - ndb_redis is a wrapper oven hiredis wrapper and doesn't implement
serialisation protocol for redis, so, whatever you send delivered to redis
being only wrapped via hirer's client library.
Btw if uou use lua < 5.2 you may try redis-lua library. It is easier to use
with lua.
On Thu, 9 Nov 2023, 09:34 Ali Taher via sr-users,
mailto:sr-users@lists.kamailio.org>> wrote:
I’m still stuck… I appreciate any assistance here.
From: Ali Taher via sr-users
mailto:sr-users@lists.kamailio.org>>
Sent: Tuesday, November 7, 2023 12:47 PM
To: Kamailio (SER) - Users Mailing List
mailto:sr-users@lists.kamailio.org>>
Cc: Ali Taher mailto:ata...@vanrise.com>>
Subject: [SR-Users] Redis issue when getting results in Kamailio
Hello all,
I have redis database where the data is inserted in JSON format:
#Redis Command
JSON.GET Index1J:12345678
#Redis Result
[
{
"SupplierID": "7",
"BCode": "987654",
"Rate": 0.159
},
{
"SupplierID": "7",
"BCode": "765432",
"Rate": 0.159
},
{
"SupplierID": "7",
"BCode": "5432159",
"Rate": 0.159
}
]
When I try to get the result using Lua script in Kamailio like the following:
#kamailio command
if KSR.ndb_redis.redis_cmd("srvN", "JSON.GET pricelistJ:441246890", "ra") then
KSR.xlog.xinfo("array size: $redis(ra=>size)\n");
KSR.xlog.xinfo("Redis Response: " .. tostring(ra) .. "\n")
KSR.xlog.xinfo("first values: $redis(ra=>value[0]) ,
$redis(ra=>value[1])\n");
end
Nov 7 07:58:15 868d08ee318d /usr/sbin/kamailio[72]: INFO: LUA {INVITE}:
[SR-Users] pv cache limit is going to be exceeded
Hello all,
I'm using the following Lua script to fetch data from Redis, which is in form
of list:
function split(str, delimiter)
for token in string.gmatch(str, "[^" .. delimiter .. "]+") do
return token
end
end
function contains(mainString, subString)
return string.find(mainString, subString, 1, true) ~= nil
end
function longest_match_acode()
local currentSeconds = os.time()
local milliseconds = os.clock() * 1000
local ddate = os.date("%Y-%m-%d %H:%M:%S", currentSeconds) ..
string.format(".%03d", milliseconds)
KSR.xlog.xinfo("starting date:" ..ddate)
local size = 0
local result
local acode = "29192"
local inputbcode = "4474173"
local delimiter = ":"
while size == 0 do
local currentSeconds = os.time()
acode = string.sub(acode, 1, string.len(acode)
- 1)
KSR.ndb_redis.redis_cmd("srvN", "LRANGE price:"
.. acode .." 0 -1", "r")
size = tonumber(KSR.pv.get("$redis(r=>size)"))
end
local left = 0
local right = size - 1
while left <= right do
local currentSeconds = os.time()
local milliseconds = os.clock() * 1000
local ddate = os.date("%Y-%m-%d %H:%M:%S", currentSeconds) ..
string.format(".%03d", milliseconds)
resultRight =
tostring(KSR.pv.get("$redis(r=>value[" .. right .. "])"))
resultLeft =
tostring(KSR.pv.get("$redis(r=>value[" .. left .. "])"))
local bcodeLeft = split(resultLeft, delimiter)
local bcodeRight = split(resultRight, delimiter)
if contains(inputbcode, bcodeRight) then
KSR.xlog.xinfo("Input bcode
contains bcode value from Redis: " .. resultRight .. " at index: " .. right ..
" with date: " .. ddate)
end
if contains(inputbcode, bcodeLeft) then
KSR.xlog.xinfo("Input bcode contains bcode
value from Redis: " .. resultLeft .. " at index: " .. left .. " with date: " ..
ddate)
end
left = left + 1
right = right - 1
end
local currentSeconds = os.time()
local milliseconds = os.clock() * 1000
local ddate = os.date("%Y-%m-%d %H:%M:%S", currentSeconds) ..
string.format(".%03d", milliseconds)
KSR.xlog.xinfo("ending date:" ..ddate)
end
function ksr_request_route()
if KSR.is_INVITE() then
longest_match_acode()
KSR.sl.send_reply("404", "Not found");
end
end
I'm getting the following warning in the log : WARNING: LUA {INVITE}:
[core/pvapi.c:340]: pv_cache_add(): pv cache limit is going to be exceeded -
pkg memory may get filled with pv declarations
I think the issue is in the following lines:
resultRight = tostring(KSR.pv.get("$redis(r=>value[" .. right .. "])"))
resultLeft = tostring(KSR.pv.get("$redis(r=>value[" .. left .. "])"))
the list coming from Redis has around 30k items, which I'm iterating through
using value[x] which is exhausting the pv declaration as per my understanding.
How to overcome this issue? Is it possible to save the list as an array
variable and then iterate through that array?
Regards,
__
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[SR-Users] Re: pv cache limit is going to be exceeded
Hello all,
If the data returned from redis is of type array, and if I want to iterate
through all the elements in the array, then I have to get $redis(r=>value[x])
for each element, and a pseudo variable for each element is allocated in the
memory, is that correct?
Regards,
From: Ali Taher via sr-users
Sent: Wednesday, November 15, 2023 4:51 PM
To: Kamailio (SER) - Users Mailing List
Cc: Ali Taher
Subject: [SR-Users] pv cache limit is going to be exceeded
Hello all,
I'm using the following Lua script to fetch data from Redis, which is in form
of list:
function split(str, delimiter)
for token in string.gmatch(str, "[^" .. delimiter .. "]+") do
return token
end
end
function contains(mainString, subString)
return string.find(mainString, subString, 1, true) ~= nil
end
function longest_match_acode()
local currentSeconds = os.time()
local milliseconds = os.clock() * 1000
local ddate = os.date("%Y-%m-%d %H:%M:%S", currentSeconds) ..
string.format(".%03d", milliseconds)
KSR.xlog.xinfo("starting date:" ..ddate)
local size = 0
local result
local acode = "29192"
local inputbcode = "4474173"
local delimiter = ":"
while size == 0 do
local currentSeconds = os.time()
acode = string.sub(acode, 1, string.len(acode)
- 1)
KSR.ndb_redis.redis_cmd("srvN", "LRANGE price:"
.. acode .." 0 -1", "r")
size = tonumber(KSR.pv.get("$redis(r=>size)"))
end
local left = 0
local right = size - 1
while left <= right do
local currentSeconds = os.time()
local milliseconds = os.clock() * 1000
local ddate = os.date("%Y-%m-%d %H:%M:%S", currentSeconds) ..
string.format(".%03d", milliseconds)
resultRight =
tostring(KSR.pv.get("$redis(r=>value[" .. right .. "])"))
resultLeft =
tostring(KSR.pv.get("$redis(r=>value[" .. left .. "])"))
local bcodeLeft = split(resultLeft, delimiter)
local bcodeRight = split(resultRight, delimiter)
if contains(inputbcode, bcodeRight) then
KSR.xlog.xinfo("Input bcode
contains bcode value from Redis: " .. resultRight .. " at index: " .. right ..
" with date: " .. ddate)
end
if contains(inputbcode, bcodeLeft) then
KSR.xlog.xinfo("Input bcode contains bcode
value from Redis: " .. resultLeft .. " at index: " .. left .. " with date: " ..
ddate)
end
left = left + 1
right = right - 1
end
local currentSeconds = os.time()
local milliseconds = os.clock() * 1000
local ddate = os.date("%Y-%m-%d %H:%M:%S", currentSeconds) ..
string.format(".%03d", milliseconds)
KSR.xlog.xinfo("ending date:" ..ddate)
end
function ksr_request_route()
if KSR.is_INVITE() then
longest_match_acode()
KSR.sl.send_reply("404", "Not found");
end
end
I'm getting the following warning in the log : WARNING: LUA {INVITE}:
[core/pvapi.c:340]: pv_cache_add(): pv cache limit is going to be exceeded -
pkg memory may get filled with pv declarations
I think the issue is in the following lines:
resultRight = tostring(KSR.pv.get("$redis(r=>value[" .. right .. "])"))
resultLeft = tostring(KSR.pv.get("$redis(r=>value[" .. left .. "])"))
the list coming from Redis has around 30k items, which I'm iterating through
using value[x] which is exhausting the pv declaration as per my understanding.
How to overcome this issue? Is it possible to save the list as an array
variable and then iterate through that array?
Regards,
__
Kamailio - Users Mailing List - Non Commercial Discussions
To unsubscribe send an email to sr-users-le...@lists.kamailio.org
Important: keep the mailing list in the recipients, do not reply only to the
sender!
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[SR-Users] Re: pv cache limit is going to be exceeded
Thanks Daniel,
Works like a charm!
Regards,
From: Daniel-Constantin Mierla
Sent: Friday, November 17, 2023 10:17 AM
To: Kamailio (SER) - Users Mailing List
Cc: Ali Taher
Subject: Re: [SR-Users] Re: pv cache limit is going to be exceeded
Hello,
the way the pseudo-variables were designed makes different definitions for
$redis(r=>value[i] and $redis(r=>value[j], practically, if you have the index
going on a large range, then you define a lot of variables. The solution is to
leverage another pseduo-variable that you set with the index, like:
KSR.pvx.var_seti("i", right);
resultRight = tostring(KSR.pv.get("$redis(r=>value[$var(i)])"));
KSR.pvx.var_seti("i", left);
resultLeft = tostring(KSR.pv.get("$redis(r=>value[$var(i)])"));
Cheers,
Daniel
On 16.11.23 10:46, Ali Taher via sr-users wrote:
Hello all,
If the data returned from redis is of type array, and if I want to iterate
through all the elements in the array, then I have to get $redis(r=>value[x])
for each element, and a pseudo variable for each element is allocated in the
memory, is that correct?
Regards,
From: Ali Taher via sr-users
<mailto:sr-users@lists.kamailio.org>
Sent: Wednesday, November 15, 2023 4:51 PM
To: Kamailio (SER) - Users Mailing List
<mailto:sr-users@lists.kamailio.org>
Cc: Ali Taher <mailto:ata...@vanrise.com>
Subject: [SR-Users] pv cache limit is going to be exceeded
Hello all,
I’m using the following Lua script to fetch data from Redis, which is in form
of list:
function split(str, delimiter)
for token in string.gmatch(str, "[^" .. delimiter .. "]+") do
return token
end
end
function contains(mainString, subString)
return string.find(mainString, subString, 1, true) ~= nil
end
function longest_match_acode()
local currentSeconds = os.time()
local milliseconds = os.clock() * 1000
local ddate = os.date("%Y-%m-%d %H:%M:%S", currentSeconds) ..
string.format(".%03d", milliseconds)
KSR.xlog.xinfo("starting date:" ..ddate)
local size = 0
local result
local acode = "29192"
local inputbcode = "4474173"
local delimiter = ":"
while size == 0 do
local currentSeconds = os.time()
acode = string.sub(acode, 1, string.len(acode)
- 1)
KSR.ndb_redis.redis_cmd("srvN", "LRANGE price:"
.. acode .." 0 -1", "r")
size = tonumber(KSR.pv.get("$redis(r=>size)"))
end
local left = 0
local right = size - 1
while left <= right do
local currentSeconds = os.time()
local milliseconds = os.clock() * 1000
local ddate = os.date("%Y-%m-%d %H:%M:%S", currentSeconds) ..
string.format(".%03d", milliseconds)
resultRight =
tostring(KSR.pv.get("$redis(r=>value[" .. right .. "])"))
resultLeft =
tostring(KSR.pv.get("$redis(r=>value[" .. left .. "])"))
local bcodeLeft = split(resultLeft, delimiter)
local bcodeRight = split(resultRight, delimiter)
if contains(inputbcode, bcodeRight) then
KSR.xlog.xinfo("Input bcode
contains bcode value from Redis: " .. resultRight .. " at index: " .. right ..
" with date: " .. ddate)
end
if contains(inputbcode, bcodeLeft) then
KSR.xlog.xinfo("Input bcode contains bcode
value from Redis: " .. resultLeft .. " at index: " .. left .. " with date: " ..
ddate)
end
left = left + 1
right = right - 1
end
local currentSeconds = os.time()
local milliseconds = os.clock() * 1000
local ddate = os.date("%Y-%m-%d %H:%M:%S", currentSeconds) ..
string.format(".%03d", milliseconds)
KSR.xlog.xinfo("ending date:" ..ddate)
end
function ksr_request_route()
if KSR.is_INVITE() then
longest_match_acode()
KSR.sl.send_reply("404", "Not found");
end
end
I’m getting the following warning in the log : WARNING: LUA {INVITE}:
[core/pvapi.c:340]: pv_cache_add(): pv cache limit is going to be exceeded -
pkg memory may get fil
[SR-Users] print_dset(): no r-uri or branches
Hello all,
I'm using Kamailio as SIP redirect where I'm appending the contact header then
send reply back using sl_send_reply("300","Multiple Choices").
But I'm getting "print_dset(): no r-uri or branches" as warning in the log and
no packet is sent back from Kamailio to the original sender.
I'm using same Kamailio configuration in another setup and it's working
correctly without facing this error.
I'm not able to detect the issue. Can you please assist?
Regards,
__
Kamailio - Users Mailing List - Non Commercial Discussions
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Important: keep the mailing list in the recipients, do not reply only to the
sender!
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[SR-Users] Kamailio cluster options
Hello all, I'm using Kamailio as SIP redirect (act as routing decision for SBC) depending mainly on sqlops and db_postgres modules, and it's working perfectly. What I need now is to have a kind of cluster of Kamailio servers for high availability purpose. What is the best option for me ? BR __ Kamailio - Users Mailing List - Non Commercial Discussions To unsubscribe send an email to sr-users-le...@lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe:
