My system contain:
- Freeswitch server
- Sip Client: Web using sipjs , mobile react-native using https://github.com/datso/react-native-pjsip to receive call.
My problem is when call done i need to know the uuid of CDR recently add to Postgres DB of that call to attach some info to that call
I try many way but can not success ex: write http request to select into postgres DB, but can not find exactly which uuid because one extension can make many call one time.
Can anyone help me solve this case?
Mark each call with custom variable like callid
Send that to your app via any method, as sipheader, as sip message, using http post etc.
But you also can just search cdr by start date.
Related
In my SignalR hub, I use the following method to check whether a user has an active connection:
var receivingClient = Clients.User(receiver);
if (receivingClient != null)
{
But I also track the online users manually over OnConnected \ OnDisconnected (in a ConcurrentDictionary). Now even when I shut down everything and start the server from scratch (e.g. IISExpress from VS), the above code part returns a result for a connection that doesn't exist.
Let's say I send from User A to user B. I start the server, go online with user A, then send a message to B: The above code returns a Microsoft.AspNetCore.SignalR.Internal.UserProxy<mySite.Services.ChatHub>.
I don't get it. Is it wrong to check for existing client connections with a null check? Should I exclusively rely on my manual tracking?
Thanks for some insight!
(PS: This is all on the same server - no load balancing / sharding)
Clients.User(receiver) returns a type that is used to invoke methods for the given user. It doesn't have anything to do with whether the user you pass in exists or not.
Is it wrong to check for existing client connections with a null check? Should I exclusively rely on my manual tracking?
Yes. Use manual tracking.
I am not using real-time asterisk , But still astdb.sqlite3 contains entries of online peers with Reg.Contact information in SIP/registry/peer. key . I would like to store contact information of all peers as they come online in a separate persistent database. I need this for sending push notifications by fetching deviceID etc information in registration contact .
I tried to pull this information from astdb.sqlite3 but the entries are clearing off as soon as devices go offline .Though I am able to fetch the information with "sip show peer XXXX" in asterisk CLI , It is overburdened to fetch every time like this . Instead I want to save only Regcontact information for all the devices in a database ( without realtime) as the devices come online. The other way I tried to pull the information is using AMI event listener. But with AMI I don't see complete information like contact information It displays only below information
Event: PeerStatus
Privilege: system,all
SequenceNumber: 75
File: manager.c
Line: 1856
Func: manager_default_msg_cb
ChannelType: SIP
Peer: SIP/2030
PeerStatus: Reachable
Can someone suggest a better way to push Only Regcontact information to a database as the devices come online .
There are no mechanism like that in asterisk.
You can use kamailio or write patch similar to this one https://reviewboard.asterisk.org/r/4490/
It sounds like you have dynamic IPs for your endpoints, and you want a way to update a separate DB as soon as a device registers with an IP/port pair.
If you enable the security log, you will see all auth events, including the "SuccessfulAuth" event, which includes the RemoteAddress of the endpoint (including port and protocol).
Here is an example line:
[Jul 21 19:53:45] SECURITY[1342] res_security_log.c: SecurityEvent="SuccessfulAuth",EventTV="2020-07-21T19:53:45.182+0000",Severity="Informational",Service="SIP",EventVersion="1",AccountID="102",SessionID="0x7f41040132c0",LocalAddress="IPV4/UDP/10.0.0.200/5060",RemoteAddress="IPV4/UDP/10.0.0.75/5062",UsingPassword="1"
If all you're looking for is AccountID="102" and RemoteAddress="IPV4/UDP/10.0.0.75/5062", a very fast/cheap way to get it is to enable the security log, and use a script to tail it and update your DB as soon as the event occurs. I like to keep the security log on anyways for utilities like fail2ban. Just make sure your script is able to reopen the file each time it is rotated.
Edit:
By default the log is in /var/log/asterisk. To enable it, edit /etc/asterisk/logger.conf and un-comment (or create) the line under [logfiles] that says security => security.
A while back I set up BizTalk to pick up a file via FTP and drop it into a network directory. It's all passsthru so I didn't use an orchestration.
Now I've been asked to execute a stored procedure once the file is picked up. The procedure contains no parameters and I do not need the contents of the file.
It seems like such a simple request but I can't figure it out. Is there any way to do this without over complicating things?
This can be accomplished through the use of either the WCF-SQL adapter or the WCF_Custom adapter with a SQL binding. You can do this using messaging only with just a SendPort with a filter/map on it thus no orchestration needed.
For the SOAP action header use TypedProcedure/dbo/name_of_your_stored_procedure and in the messages tab you can specify the paramters to the stored procuders as well as add a payload in the following manner:
<name_of_your_stored_procedure xmlns="http://schemas.microsoft.com/Sql/2008/05/TypedProcedures/dbo">
<parameter1>XXXX</parameter1>
<xml_parameter>
<bts-msg-body xmlns="http://www.microsoft.com/schemas/bts2007" encoding="string"/>
</xml_parameter>
</name_of_your_stored_procedure>
In the above case xml_parameter will have the contents of the message payload passed to it.
The stored procedure should look something like :
CREATE PROCEDURE [dbo].[name_of_your_stored_procedure]
#parameter1 int,
#xml_parameter nvarchar(max)
AS
BEGIN
-- your code goes here
END
More details can be found here
Regards Hasse
This MSDN page describes the process and has this to say: "You must create a BizTalk orchestration to use BizTalk Server for performing an operation on SQL Server."
However if you're really desperate not to use an orchestration I believe you have the option of setting the operation context property in a custom pipeline component. Then you can initialise the message in a map on a port. In theory this should work but I can't guarantee it.
I'am working on a Symfony app that provides a rest web service (simple HTTP Request with JSON).
That service check some rules and inserts few lines in two MySQL table (write only).
For optimize reason, even if Doctrine bundle is available, i use native MySQL Query (with bind params) to insert this lines.
My need is : If for any reason, the database is not available, write "runnables" queries into a log file.
The final purpose is that when database is back, i want to be able to execute directly the file's content on the database.
Note that there is no unique constraint (pk is a generated uuid) and no lock or transaction to handle (simple insert statements).
I write a custom SQLLogger, but when $connection->insert(...) is called, the connect fail before logger is called.
So, my question is : There is a way to get the final query (with binded parameters) without database connection ?
Or should i rewrite the mecanism that bind params into query and log it myself when database is not available ?
Best regards,
Julien
As the final query with parameters is build by the database, there is just no way to build the query with PHP and to be garanteed that the query will be the same as the database.
The only way si to build query without binded parameters, but this is clearly not a good practice.
So, i finally decided to store all the JSON (API request body) in a file if the database is not available.
So when the database is back, instead of replay SQL queries, i can replay the original HTTP query.
Hope this late self-anwser will help someone.
Best regards.
How can I achieve the following with FreePBX 12 (and 6): I need our system to check on the fly the destination for that incoming phone call to be transferred too.
When a call comes in, the system needs to check a database table to see if there’s a record with that Caller ID, that record will also have the destination extension where that call needs to be routed too.
The database is a MySQL Table and it will consist of the following fields: id, callerid, destination_extension, created_at, updated_at
Call flow
1- answer incoming call
2- get call caller id: 876-718-7137
3- connect to mysql database
4- check if theres a record with that caller id and get the
destination extension where to route it (SELECT
destination_extension FROM callers_table WHERE caller_id =
876-718-7137) - (Returns: 1001)
5- transfer incoming from to extension 1001
Any suggestions on how to accomplish this? Thank you!
I use the Dynamic Route module to accomplish this. See:
http://www.voipsupport.it/pmwiki/pmwiki.php?n=Freepbx.DynamicRouting
There are no features like that in freepbx.
Only posibility - add all combination in inbound route in format DID/CID
You also can write custom dialplan using
http://www.voip-info.org/wiki/view/Asterisk+func+func_odbc
or
http://www.voip-info.org/wiki/view/Asterisk+RealTime