How can i look on cli sip peer status like ringing, busy, in use, etc. Note that sip peer is not member any queue.
Sip peer status is reachable/unreachable.
What you have asked is device state. Asterisk have multiple method for check device state, for example you can do it via AMI or ARI
https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Devicestates+REST+API
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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 have problem Asterisk do not terminate channel when member goes UNREACHABLE or UNREGISTERED.
I have Queue and some member. Member answer incoming call from Queue. If during the conversation member status would beUNREACHABLE or UNREGISTERED Asterisk do not terminate channel. Channel terminated after hangups call.from caller.
Note: i have option
rtptimeout = 10
Peer status have no any relation to channel state
Asterisk should not terminate channel.
rtptimout option have no relation to peer state. It will stop call on no rtp for 10 sec. So likly you have rtp.
I found an interesting documeent about realisation of eCall (Emergency Call) in EU: http://www.heero-pilot.eu/ressource/static/files/heero_wp3_d3-3_final-operational-results_v2.3_final.pdf
Germany somehow did it on Asterisk. Whatever, I don't understand how they process MSD (minimal set of data) using Asterisk. In the call session for the first step caller sends DTMF signals to send MSD packet. As I understood, Asterisk must redirect this call to In-band modem on COM port or to another machine with such modem. After PSAP successfully received MSD for the second step caller switches to voice channel that must be redirected to some sip-client of PSAP operator. How they do it? Is there a way to receive DTMF signals w/o modem by using internal capabilities of Asterisk? How the same call to redirect to another SIP on the same time?
I suspect that you are referring to emergency services, rendered by emergency dialers - eg. for senior citizens. These are fairly common where I live, and I've created in the past a solution to handle the calls from these, based on Asterisk. The solution involved a way to intercept the various DTMF signals that the device generates, then making Asterisk do stuff with it. Back then, I used Asterisk 1.6 and it is pain staking, because I had to do everything from within a MeetMe bridge, and interact with Manager alot. Today, doing the same with Asterisk 12/13 and ARI is a breeze. Just remember one thing, most of these dialers will utilize the A,B,C,D DTMF signals, which are somewhat unknown to most people - they exist and Asterisk is very much capable of handling those.
The only snag is - make sure you are connected via a PRI, as most SIP carriers aren't aware of these signals, and their SIP trunks won't support this type of signalling.
Asterisk can send dtmf(natively,via command SendDTMF in dialplan or D option in Dial command) or any other sound(custom c/c++ app needed)
No, you not need special modem.
However there are no realisation acordinly to that document, you need do that yourself or hire someone
I am new to ZMQ. I find ZMQ socket implementation much simpler than winsock. But my doubt is that "can a client created using ZMQ TCP socket talk to conventional TCP server?" in other words Can my ZMQ client communicate with a TELNET server implemented using winsock?
No it cannot. ZeroMQ is a messaging library and is not just a pure socket. It uses its own protocol called ZMTP and both endpoints are required to understand it.
Yes you can. It's called a ZMQ_STREAM socket. Documentation is here: zmq_socket.txt
A socket of type ZMQ_STREAM is used to send and receive TCP data from a non-ØMQ peer, when using the tcp:// transport. A ZMQ_STREAM socket can act as client and/or server, sending and/or receiving TCP data asynchronously.
When receiving TCP data, a ZMQ_STREAM socket shall prepend a message part containing the identity of the originating peer to the message before passing it to the application. Messages received are fair-queued from among all connected peers.
When sending TCP data, a ZMQ_STREAM socket shall remove the first part of the message and use it to determine the identity of the peer the message shall be routed to, and unroutable messages shall cause an EHOSTUNREACH or EAGAIN error.
When I type "core show channels" I get a channel for the current calls through DAHDI but I want see calls between sip to sip too.
Use http://asternic.org or some other panel project already writed.
btw, core show channels show all calls, including sip. If you not see sip calls, you have set on your peer
canreinvite=no