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
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 am trying to get a solution to ask one my connected phone (SIP) to call an another one.
I can easily do this with an originate via one AMI request, but I have one problem :
The display on my sip phone always show "incomming call".
There is one solution to get the display "outgoing call" on my sip phone ? Either with asterisk or in SIP protocol.
"Incoming call" is writed by your phone's firmware.
So you can only change to "outgoing call"(actualy it IS incoming) by rewriting call firmware or use phone which have that feature in firmware.
SIP standart have no any info about such feature.
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 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 have two phones one using IAX2 second SIP. When I press Hold button on IAX2 phone I get Asterisk event "Hold", When I press Hold button on SIP Phone I get Unlink then Bridge event. Can I change this behavior for SIP phone to get "Hold" event from Asterisk? Why SIP phone not send one "Hold" event?
Thank you!
What version of Asterisk are you using?
A re-INVITE coming from a SIP UA notifying Asterisk to put the channel on hold should not unlink a bridge with another channel, unless the hold from the re-INVITE is asking it to do something other the just simply restrict the flow of the RTP.
You may want to post this on issues.asterisk.org/jira. If you do so, please include a DEBUG log, with sip set debug on, illustrating the problem.