When I have call request in AGI script, is it possible to get time since call was created? For example, if it was routed sequentially through several softswitches can I get whole time sinse first softswitch or since generation in remote softphone ?
SIP only is used.
I also use AMI, and in CDR event I see that (Duration - BillableSeconds) is sometimes longer (for 10-20 seconds) then time from AGI request to Up, so duration may be already includes some time before call passed to my Asterisk.
No, there are no way determine any history of call before it go your switch unless other switch inform you about by special headers.
No, CDR(duration) is counted from time when call arived to asterisk. CDR(billsec) always less or equal to duration.
Related
Is there a way to get the value of how long the extension on hold state when call is happen ? I know there is rtp_holdtimeout in pjsip context option endpoints type, but i need only the value without having hangup the extension. Thank
Yes, each hold produce event on start/stop.
You have read events via AMI Events routine, record for this call ID all start/stop and calculate.
No, it is not doable without external program.
I am using Putty to simulate my phone's modem connected via serial. When my phone receives a call it outputs 'RING' into putty but when the caller cancel the call Putty doesn't out put any response or result.
How would the modem know that the caller disconnect/cancelled the call, but not output it in putty?
Thanks
To detect missed calls you can try three things.
Check if there is a suitable AT+CIND indicator you can turn on. I do not think call will do since I assume it only goes to 1 when the call is answered. If your phone supports callsetup or something similar that should be what you need (you will have to implement logic to detect when a call does not go to state active).
For an example of enabling AT+CIND indicators, see chapter "8.57 Informative examples" in 27.007 for more explanation, and pay close attention to The subparameter order in the command is defined by the query command order, e.g.
if AT+CIND=? returns
+CIND: ("abc",(0-1)),("xyz",(0,1)),("call",(0,1))
then call is index 3, and for
+CIND: ("abc",(0-1)),("call",(0,1)),("xyz",(0,1))
call is index 2. Do not hard code any assumptions here, this should be parsed and checked run-time (one check at the beginning is enough).
Alternatively you can upon RING start polling call status with AT+CLCC until the call is no longer listed.
Or you could poll the MC phonebook storage and detect changes.
Most modems show the incoming phone number and a RING when a call is received and an END when the call is cancelled. To view the missed calls, you may use the following AT Commands.
AT+CPBS="MC"
AT+CPBR=1,99
First command tells the modem to look in the missed call phone book and the second command loads entries from 1 to 99. Note that this behavior is not standard. I was able to replicate this on a GSM module but not on my 3G modem. Try it on your modem and check if this works. All the best.
I forward incoming calls to external numbers.
I do this with Follow me module for each number.
Sometimes I use one more external number in follow-me list, to call the second ona if the first one is busy. Bu it is calling the first one and ringing. First one is busy and it is still ringing. I does not call the second number.
What i am doing wrong?
Your provider or ATA not recognize busy signal. Unfortanly no way say what exactly happens, info provided not enought
I have my own database to log the calls in astreisk. I need to insert call duration of every call into a table. How can I do this? Can I do this in my dialplan?
You are not giving much information about what db backend you would like to use, and also if you are asking about how to write yourself the call duration or how to configure asterisk to write the cdr in question.
So, generally speaking, you have 3 possible options for this (see below). For options 2 and 3 you would have to open the connection to the database yourself, write the queries needed to insert/update whatever row(s) needed, handle errors, etc. While for option 1 you just need to configure asterisk to do the job.
1) Asterisk can do this by default on its own, by writing the CDR (Call Detail Record) of every call to a backend. This backend can be csv, mysql, pgsql, sqlite and other databases through the cdr_odbc module. You have to configure your cdr.conf (and depending on the backend you chose, cdr_mysql.conf, cdr_odbc.conf, cdr_pgsql.conf with your backend information, like credentials, table names, etc).
The CDR will be written by default with some contents, which are the CDR Variables (taken from the predefined asterisk variable list)
If the channel has a cdr, that cdr record has it's own set of
variables which can be accessed just like channel variables. The
following builtin variables are available and, unless specified,
read-only.
The ones interesting for you at this point would be:
${CDR(duration)} Duration of the call.
${CDR(billsec)} Duration of the call once it was answered.
${CDR(disposition)} ANSWERED, NO ANSWER, BUSY
When disposition is ANSWER, billsec will contain the number of seconds to bill (the total "answered time" of the call), and duration will hold the total time of the call, including non-billed time.
2) If, on the other hand, if you are not asking about the cdr, but want to write yourself the call duration, you can have an AGI script that after issuing a dial(), reads the CDR(billsec) variable, or the ANSWEREDTIME (set by the Dial() command):
${DIALEDTIME} * Time for the call (seconds)
${ANSWEREDTIME} * Time from dial to answer (seconds)
3) You can also achieve the same result by having an AMI client listening for the event VarSet for the variable ANSWEREDTIME. The event in question will contain the channel for which this variable has been set.
So, options 2 and 3 are clearly more useful if you already have an AGI script or an AMI client of your own controlling/handling calls, and option 1 is more generic but maybe slightly less flexible.
Hope it helps!
When a workflow has a receive activity that occurs after another receive activity and the second receive activity is called first the workflow holds the caller by blocking for 1 minute before timing out.
I want the workflow to return immediately when there are no matching workflow instances.
I do not want to change the timeout on the client as some calls may take a while.
This is a known issue in WF4, at least I am not aware of it being fixed yet.