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
Related
I am quit new in asterisk.
My problem is: I have to implement call hold and pick up in asterisk.
Let's say we have an incoming call called incomingCallA.
As an operator I want to hold incomingCallA for a while to deal with other works, then come back soon, If i am not come back, others operator will pick up my held call.
I have done some researches that told me:
Call hold is sip client function, dont need to changes at asterisk server.
I can use call parking to dial to others operator if i am not come back.
Please help me to about the solution to deal with above problem.
Many thanks.
Simplest for begginer is create 100 conference rooms using Meetme after that just send to first availible room and record room number.
You can forward call using Transfer AMI function.
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.
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.
I am trying to implement a data transfer from an embedded board to a PC. For this, I need to use low latency communication and I am bound to use Ethernet with TCP/IP.
Furthermore, I'm using the lwip stack.
First of all, I disabled nagle algorithm, because I have to send small packets of data (10 KB) and I want them to be sent as soon as possible, without waiting for intermediate ACKS.
The Wireshark Log shows me that this is working quite fine (the whole data is being sent to the PC in about 1msec).
After that, the PC takes about 200msec to send the last ACK (because the last Segment is not maximum size).
The problem is now, that on the embedded processor, it takes a very long time, until the lwip gives my application the message, that all of the data has been ACKED.
When I decrease the TCP_INTERVAL (to let's say 5), it speeds up greatly.
I am wondering, why lwip behaves like this? I would think that the Periodic-TCP-Tasks (which are being called according to the TCP_INTERVAL) have nothing to do with the Handling of the received frames (which is really another call in the main).
I hope I could state my problem somehow understandable, if not I would appreciate feedback, so I can improve my question!
Thanks!
EDIT:
After more debugging, I found out that the process of sending data results in the following function calls:
My main calls tcp_write(...)
tcp_tmr() is called multiple times (through the LwIP_Periodic_Handle() function). This happens seven times. During the eigth call:
tcp_output() is called. During this call, all segments which were added during the last tcp_write() call are sent by calling tcp_output_segment().
So now it is clear that if I reduce the TCP_INTERVAL, of course the data gets sent sooner, because the tcp_tmr() function is called more quickly.
but my question is still: Is this the normal behaviour? It seems a bit odd, that lwIP is waiting such a long time before actually sending the data.
Since Youre doing this My main calls tcp_write(...)
use tcp_output() immediately after tcp_write
or else use tcp_write() in tcp_recv callback
This is not a programming question per se. I am trying to build a system which consists of the following:
User calls system using regular land line
Some processing is done by asterisk
Call is forwarded to an external number (another landline/mobile phone)
Now I would like atleast 2 simultaneous lines on which the user can call. I would like to know the following:
Will the TDM410 work for what I am trying to achieve?
Since I want call forwarding, do I need an extra line for that? Or can I do it on the same line? e.g. for one user do I need one incoming line and one outgoing line or can I do both receiving and forwarding on the same line?
I have both asterisk books but am still unclear as to which card to purchase. Is the TDM410 with 4 FXO the right one? I am thinking of TDM410 because it has 4 lines so that I can use two for incoming and two for outgoing. Am I right? Can someone please point me to a link/online store?
Thank you very much for your time.
PS- I do not wish to use SIP. I want to use POTS for all my calls (incoming and outgoing)
The TDM400 with 4 FXO modules will work for what you need.
When you receive a call on one line, then that line is busy and you must use a different line to forward the call (Asterisk will bridge the calls but it needs 2 lines for that). You can buy them directly from Digium.
You need at least 3 landlines if you want to be able to receive calls from 2 lines; the third line will be the one you use to forward the calls, but only one call can be forwarded at at time and the other person will have to wait. If you don't want that then you need 4 landlines.