I am not able to detect hangup event when I am using Indian numbers but for USA numbers the the hangup is detected. If an indian number is connected and he is in a conference (confbridge) then if he hangup his phone then the hangup event is not fired.
Please help.
Assuming this is a SIP channel... You could try to detect the hangup based on the RTP timeout.
The setting in your [device] or globally, is:
rtptimeout=60
Which is the default, 60 seconds. You could crank this down a lot lower if it helps you out.
That depend of provider used. Change sip provider/or choose more costly plan. Hangup detection in this case have be handled by provider's equipment.
Related
Asterisk 14 (from Packages # tucny.com)
Connected to Twilio SIP trunk
Asterisk does not detect inband DTMF tones.
Other ways(INFO, rfc2833) to send DTMF works correctly.
I've played with tone duration and volume without success.
In DTMF debug I can see the asterisk reaction to incoming INFO or rfc2833 events, but nothing happens when inband tone is coming.
Seems like asterisk does not "hear" the line at all.
I couldn't find any information about modules requeried to detect inband DTMF.
Installed modules:
- asterisk
- asterisk-odbc
- asterisk-pjsip
- asterisk-hep
- asterisk-sounds-core-en-alaw
- asterisk-sounds-core-en-ulaw
Will be very thankfull for any information.
I do not speak English very well, but I can always try to learn. Sorry for bad interpretations.
DTMF tones need more dependencies as a codec.
Check the codec used on the channel and try changing the dtmfmode to inband. Another item can help is the tones / frequencies in cases of FXO.
In case of codec, check to use alaw.
I hope I have helped or at least give a light.
Hugs!
The cause of the problem was found.
Pjsip does not support Inband DTMF detection.
There is information that to detect inband DTMF with pjsip you need to write your own pjsip plugin:
https://trac.pjsip.org/repos/wiki/FAQ#dtmf
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
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.
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