Is it possible to make a condition of outgoing VoIP calls on Asterisk?
if(the receiver is a fax){
don't start calling;
}
If not, can we log(save) outgoing fax calls(with call info like time, duration, channel, location) into a file somehow?
Thank you in advance!
You need to detect when a fax machine answeres an outgoing call. NV_FaxDetect in calls. For more details use this link.
Using Asterisk to Detect and Redirect Fax Calls
Related
I want to program an esp8266 doorbell to call me when someone presses the bell. I have a STARFACE telephone system (Asterisk) and would like to tell STARFACE to make a broadcast call. I have searched the Internet but I find only FritzBox examples.
I do not want to do this with a call file.
Sorry for my English. I am not a native Englishman.
Call file is simplest way do that.
Some other ways
asterisk AMI protocol Originate command
asterisk ARI
perl,sipp(testing tool) or other script which send sip invite with auth.
https://gist.github.com/maximevalette/802764
http://sipp.sourceforge.net/
click2call script on asterisk (using call file or other)+ curl request on your device.
I want to ask how the mechanism when broadcast communication occurs (many endpoints) where each endpoint enters the bridge through the ARI originate (outgoing call) function.
Suppose there are 3 participants in a communication bridge. When one endpoint is talking, how does the mechanism occur? Does the endpoint make 2 packets with the same payload, then sent separately by the endpoint or only send one packet, then the asterisk duplicates the packet as much as n endpoint then sends it to each endpoint registered on the bridge?
Thank you
You can check app_conference.c code if you want.
Asterisk make independend packet for each party. How that packet created will depend on confbridge settings.
There is experimental res_rtp_multicast.c, but it not used in most scenarios.
I am using csipsimple as sip client and asterisk server to set up call. Calls are made between 2 sip clients but voice is not getting transferred.
Calls are made between 2 sip clients using AMI.
I can give my asterisk cli log.
Can anybody please give me some idea to solve this issue?
Thanks
More info would be useful. First, make sure both clients are registered, and can use at least one common codec. In most cases, these aren't the problem. It's usually a NAT/Firewall issue. Are the two clients on the same subnet? Is there any firewall rules blocking the communication?
SIP signaling usually goes on udp:5060. But that seems working. Media is tricky. In each call, the ports for RTP audio changes, in the range specified in rtp.conf. This RTP traffic goes over UDP as well. By default it't 10000-20000.
If there is only routing done between the two endpoints, it should still be fine. NAT (Network Address Translation) is your main concern. Take a look at iptables, sip_nat_conntrack. To debug, use asterisk's sip set debug on command and look for the SIP headers and verify the correct IP addresses.
I'm doing an application that uses the Modbus ASCII protocol (RS485).
A client will communicate with a PLC service that will talk to the PLC through the serial port.
I'm not sure whether there is any limitation about sending one command each time or if I need to wait for the response to come back from the PLC before sending the next command.
That will affect the architecture of the PLC service because if only one instruction can be processed at a time the design of my class will support one async method call and will expose IsBusy property. Otherwise I'll have to support concurrent method calls and pass in the stateId for each call (I'll be using EAP: Event-based Asynchronous pattern)
Thanks in advance.
The protocol is synchr. like comli so yes you can only send one command and must wait for a reply - I also doubled checked some old code just to be sure, been ages...
You can also read more about the protocol here
The Modbus protocol handles one command at a time, so wait until you have a response and then send the next command.
I am new to Asterisk. I am not clear about my concept and don't know is it possible one.
Mobile number divert all incoming call to Virtual Telephone Number. Virtual number divert to Asterisk PBX using SIP. If Asterisk receive a call, is it any possibility to get the phone number from and original destination number in Asterisk
Thanks
Check the sip headers and see if the itsp is passing it