I use Asterisk 16.5 and sip trunk.
If known sip channel can i find corresponding trunk channel?
Note: I want do it with Asterisk AMI.
By default channel name is SIP/TRUNK_NAME-uniquepart. So just parse channel and you will get trunk
There are no channels in trunk. Digital trunks have no lines.
Related
We have a SIP trunk in our company. This SIP trunk is connected to Panasonic PBX and the PBX routes the calls to the extensions. Now we need a passive call recording server. The only task that this server should do is recording all the incoming and outgoing calls of the sip trunk and SHOULD NOT answer any call. So can we use asterisk as a recording server? If not what are other solutions?
If you can connect it like panasonic->e1 card->asterisk->incoming/outgoing sip or panasonic ->sip(if your panasonic have it)->asterisk->incoming/outgoing sip - then answer is yes.
If you want it use like panasonic recording card - answer is no.
I want to send all extensions and conference bridge participants of asterisk voice to a analog telephone cable which is connected to a voicelogger ( recorder system) . How can it be done ?. I think this is possible by connecting Analog phone cable to ATA device ( linsys pap2) and sending stream to that ATA extension . But the challange is voicelogger is not an automatic answer machain .
First i have say you that idea is really strange. Asterisk can record all calls and record storage will cost much less then any analog device storage.
If you still insist you need it send to analog, you need multiple line analog device(every call record will require different wire).
Also you need FXS dahdi card and/or sip fxs adapter to connect your recorder.
You can orginize recording by using ChanSpy and/Or Confbridge as "ghost" call to all your calls with other dialling your fxs recording bank.
Complexity of such dialplan will be above average and require significant efforts and asterisk knowledge. You can read this links to get idea.
http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO
http://www.voip-info.org/wiki/view/Asterisk+cmd+ChanSpy
http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out
Other options you can have is record by asterisk and play recorded files one-by-one to your analog recorder or just use usual computer to playback files to recorder.
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
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
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