I need to make a conference voice call on GSM network.
The maximum I have seen in the datasheet is that the command AT+CLCC can report a list of current calls of ME automatically when the current call status changes.
How can I make conference calls with a SIM800L? I have 2 phone numbers for call.
The key command for the feature you are asking for is AT+CHLD (Call Holding Services). It is important to say that this is well known GSM Supplementary Service, and since AT+CHLD is a standard command it is a feature that is likely to be supported by all cellular modems, not only SIM800.
The main constraints that any user must know are:
It is a service strictly related to VOICE calls
The network operator must support this service as well
ETSI Specifications about multi-party calls
Though it may appear as a boring introduction, we need to build our procedure on solid basis. Feel free to skip this paragraph if you are just interested in the AT commands sequence.
ETSI specification TS 127.007 v15.3.0 describes its behaviour at chapter 7.13: "Call related supplementary services +CHLD":
This command allows the control of the following call related
services:
a call can be temporarily disconnected from the MT but the connection is retained by the network;
multiparty conversation
(conference calls);
the served subscriber who has two calls (one
held and the other either active or alerting) can connect the other
parties and release the served subscriber's own connection.
A further document related to MPTY calls is then referenced : 3GPP TS 22.084 (that can be found here. Another interesting source is ETSI 300 954 which states
The served mobile subscriber A may initiate an active MultiParty call
from an active call C and a held call B.
This means that we can obtain a conference call just by adding held calls to active calls.
AT Commands procedure
From the previous documents we can deduce that the following steps will setup a multi-party call:
Start a voice call with one of the parties by issuing ATD<number>;, or answer to an incoming call with ATA
Put on hold the first call by issuing AT+CHLD=2 (well supported by your SIM800, that for +CHLD=2 states "Place all active calls on hold (if any) and accept the other (held or waiting) call.").
Start a call with the third party
Start the multiparty by issuing AT+CHLD=3 (well supported by your SIM800, that for +CHLD=3 states "adds an held call to the conversation.").
About AT+CLCC
The command you mentioned in the question is not directly responsible for starting a multiparty conversation, but it is someway related to it. In fact, it is able to list the status of all the active calls.
Execution command AT+CLCC provides the following answer:
[+CLCC: <id1>,<dir>,<stat>,<mode>,<mpty>[,<number>,<type >,<alphaID>]
[<CR><LF>+CLCC: <id2>,<dir>,<stat>,<mode>,<mpty>[,<number>,<type>,<alphaID>]
[...]]]
OK
We will linger on just two relevant parameters:
<id N> is the ID of the Nth call. It is relevant because many options of +CHLD command allow to selective hold/release an X call, and this ID it is required in order to specify X in the command. All these options, not mentioned in the this answer, can be useful in order to select correctly the call to be added in the multiparty conversation.
<mpty> is the multiparty call flag, and if it is set to 1 it means that the call is one of multiparty (conference) call parties.
Related
I do have an Asterisk 11 PBX and I'm developing an Windows Service application using the github AsterNET.AMI Library to connect my PBX. Till here everything is working fine, I can send commands and read incoming event messages.
But now I need to develop a feature on my software based over one information that I thought it could be easy to retrieve. The information I'm looking for is - who hanged up?
I googled for it a lot and I could find a few answers, most of them talking about setup the G option on CDR but also some considerations about this approach. Still like this I couldn't grab any valuable information for my scenario.
Maybe if I tell you about my working scenario you could help me. Lets go, I'm going to bullet split this:
I do have a caller calling from a cellphone and this calling are incoming to my internal PBX extension
My PSTN trunk is a E1/R2 directly to my PBX
No matter if caller or callee hangs up always I do have "normal clearing" message for hangup_cause
I know I'm receiving from my service provider the information about the releasing device, because if I use my Siemens 3800 Hipath over CSTA I can retrieve this information.
So the gold question is: How can I retrieve who is the releasing device on this situation?
You can try a combination of g and F options in the Dial application. The g option allows dialplan execution when the called party hangs up, while the F option allows you to continue execution to a context,extension,priority of your choice if the caller hangs up.
So, you can understand which party is hanging up by the dialplan being executed after the call ended.
Find here more info on these options: https://www.voip-info.org/asterisk-cmd-dial/
The only way I could find after read Asterisk doc almost entirely was reading HangupRequest event messages.
As I'm using AsterNet.AMI library to connect and manage my Asterisk, so I change the source code a little bit to have an event handler do read HangupRequest event.
HangupRequest event writes the messages like the following one:
Event: HangupRequest
Privilege: call,all
Channel: SIP/8103-000001be
ChannelState: 6
ChannelStateDesc: Up
CallerIDNum: 8103
CallerIDName: Agent 8103
ConnectedLineNum: 51999887766
ConnectedLineName: 51999887766
Language: en
AccountCode:
Context: from-internal
Exten: 8100
Priority: 1
Uniqueid: 1569618521.446
Linkedid: 1569618519.445
So accordly to HangupRequest Event Asterisk documentation I could notice the channel in the message is the channel related to the releasing device, also CallerIDNum and CallerIDName are related to.
This feature is not implemented right now on the github library, but I'm going to push over there and ask them to include on next release.
Yet I don't know where to read this information on FreePBX Admin.
I was wondering, how can one get the list of ongoing calls in Asterisk when using Asterisk-Java?
It is needed when e.g. restarting ManagerAPI and I would like to display all the calls, which are currently running on Asterisk.
Previously it was done by observing properties on Channels created in Asterisk (I think it was 'linkedChannel' property), but this is now (Asterisk 13) not possible - linkedChannel is not updated when call is created.
You should issue via AMI COMMAND action with "core show channels concise" and collect output.
I logged in using agc (agent) panel in vicidial and just to test manual dial, I just executed following MySql query fron cmd.
UPDATE vicidial_live_agents set external_dial='12122351880!!YES!NO!YES!!1478530720!!!!!!' where user='1001';
After this , i got a a call on my softphone (already configured).
I am unable to understand one thing, How can an explicit entry to the vicidial_live_agents table of mysql, make a call ?
You shouldn't be messing with the database for calls.
If you configured everything correctly, you should recieve a call as soon as you log-in which will send you to a conference, you should hear a voice confirming it (you are the only one in this conference), all future calls will be connected to that conference, you should keep that call at all times and not control anything in the softphone.
If you need to make a manual calls you can either set the "Manual Dial Override" option in the campaign to "ALLOW ALL", which will enable a link in the agent screen labeled "Manual Dial" or you can use the agent_api with the external_dial function.
What you have found is the raw method that the Agent API uses to implement a manual dial call. For more information on how the Agent API works, and for all of the options you can use with the "external_dial" function, take a look at the Agent API documentation:
http://vicidial.org/docs/AGENT_API.txt
Within my BizTalk 2010 solution I have the following orchestration that’s is started by the receipt of a courier update message. It them makes a couple of call to AX's WCF AIF via two solicit-response ports, a Find request and an Update request.
For this application we are meeting audit requirements through use of the tracking database to store the message body. We are able to link to this from references provided in BAM when we use the TPE. The result for the customer is nice, they get a web portal from which they can view BAM data of message timings etc. but they can also click a link to pull up a copy of the message payloads from the tracking db. Although this works well and makes use of out-of-box functionality for payload storage it has led to relatively complex jobs for the archiving of the tracking db (but that's another story!).
My problem relates to continuation. I have created the following Tracking Profile:
I have associated the first of the orchestration's two solicit response ports with the continuation RcvToOdx based on the interchange Id and this works, I get the following single record written to the Completed activity table:
So, in this case we can assume that an entry was first written on receipt in the inbound message, with the TimeReceivedIntoBts column populated by the physical file receive port. The FindRequestToAx column was then populated by the physical WCF send port. Because this was bound to the RcvToOdx continuation Id and used the same interchange Id and the previously mentioned file receive message, the update was made to the same activity. Notification of the resulting response was also updated to the same activity record - into the FindResponseFromAx column.
My problem is that I would also like BAM to record a timestamp for the subsequent UpdateRequestToAx. Because this request will have the same interchange Id as the previous messages I would expect to be able to solve this problem by simply binding the AxUpdate send port (both send and receive parts of it) to the same continuation id, as seen in the following screen grab:
I also map the UpdateRequestToAx milestone to the physical Ax_TrackAndTraceUpdate_SendPort (Send) and the OrchestrationCompleted milestone to Ax_TrackAndTraceUpdate_SendPort (Receive)
Unfortunately, when I try this I get the following result:
Two problems can be seen from the above db screen grab:
1. Date for the update send port was inserted into a new activity record
2. The record was never completed
I was surprised by this because I'd thought since they update port was enlisted to use the same continuation, and the single InterchangeId was being used by all ports for the continuation Id then all the data milestones would be applied to a single activity.
In looking for a solution to this problem I came across the following post on Stack Overflow suggesting that the continuation must be closed from the BAM API: BAM Continuation issue with TPE. So, I tried this by calling the following method from an expression shape in my orchestration:
public static void EndBAMContinuation(string continuationId)
{
OrchestrationEventStream.EndActivity(CARRIER_ORDER_ACTIVITY_NAME, continuationId);
}
I can be sure the method was called ok because I wrapped the call with a log entry from the CAT framework which I could see in debug view:
I checked the RcvToOdx{867… continuation Id against the non-closed activity and confirmed they matched:
This would suggest that perhaps the request to end the continuation is being processed before the milestone of the received message from the UpdateAx call?
When I query the Relationsips tables I get the following results:
Could anyone please advise why the UpdateToAx activity is not being completed?
I realise that I may be able to solve the problem using only the BAM API but I really want to exhaust any possibility of the TPE being fit for purpose first since the TPE is widely used in other BizTalk solutions of the organisation.
To solve this problem I created a 2nd continuation in the TPE.
"RcvToOdx" continuation for the Find and "OdxToUpdate" continuation for the update - source is InterchangeId on the initial receive port - UPS_TrackAndTrace (same as for other "RcvToOdx" continuation), dest Id is the InterchangeId mapped to update send port.
I have a database where one entry is structured like so:
number_to_call date file_to_play
Using asterisk, I need to do the following:
1. Check the database daily.
2. If date matches that of today's, then initiate call on number.
3. Once phone has been picked up, play file_to_play.
Some general pointers on how I even begin to do this would be great.
Most of the applications that I have written so far have worked on incoming calls. I have the following questions:
1. How do I write a "daemon" that will check the database? Asterisk should have both user and group privileges for it to execute properly. How do I do this?
2. Can I initiate an outgoing call from outside of the asterisk dialplan?
The calls are being made to a PSTN/mobile number.
You can Write any script Which can check DB on daily basis and once it maches the date range you can initiate a call using .call files.
Please study asterisk auto-dial out from voip-info.org - I think you can understand better then.
You can run your script for same user as asterisk runs there is also one more method to initiate call from linux which we can call Originate CLI command which can also refer to http://voip-info.org/.
In general it is not a great idea to write your own dialer, unless your volume is very very low. Where I work, we started rolling our own but at the end went with a commercial solution that handled most of the logic. There are a number of commercial and free solutions out there, so don't reinvent the wheel.
See my answer to https://stackoverflow.com/questions/11666755/outbound-dialer-using-asterisk/14589901#14589901 for why it is not a good idea to roll your own.