I'm making an outdial using call files in asterisk and application needs some DTMF input but DTMF not working for all mobile phones, It is not accepting digits from Nokia - 1100 and nokia 6030 where DTMF works if i make an incoming callfrom the same phone but on out dial using DTMF log i can see following messsage on asterisk CLI:
DTMF Begin '1' received on Dahdi-32
DTMF Begin ignored '1' on Dahdi-32
If someone knows the solution for this please let me know.
It works if i set overlapdial=no but in this case it doesn't make two simultaneous outdial on two different numbers.
Have you tried setting relaxdtmf=true in the chan_dahdi.conf file?
Related
New to Asterisk here.
I've setup my phone as a gsm gateway using bluetooth adapter. I then direct calls from the mobile device to a queue that has one member. The member is a soft phone.
extensions.conf looks like:
[incoming-mobile]
exten => s,1,Answer()
same => n,Queue(support)
same => n,Hangup()
This works pretty well. I get the call to the sip soft phone. However, while the call is active, if I receive antother call to my phone device, that call deosn't even reach asterisk. I see the multiple call notification on my phone but it does not go through to asterisk. So my questions:
Is it even possible to queue concurrent calls via chan_mobile or does that kind of thing require more advanced hardware (e.g. gateways with multiple channels)
Even if I had some gsm gateway with 30 sim cards, what happens to the 31st call. How do people handle queuing the calls, once all channels are taken? Seems like it's certainly possible
Any pointers much appreciated
Some facts
1) chan_mobile not support any gsm gateway, it is clearly stated in project docs.
2) there are no gsm gates or channel drivers supporting second call on same gsm sim card.
3) There are no even PHONES which support that. On all phones you have pause/hold first caller to get second call.
Nobody need this feature, so no expect it will be developed in near future.
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 am trying to dial a call to GOIP and want to forward it to another GSM phone. I have added following lines in my extensions.conf (asterisk)
[goip_context]
exten=> 105,1,Set(NUMBER=913579865476)
exten=> 105,2,Dial(SIP/${NUMBER}#${GW2},20)
where 105 is my GOIP extension
NUMBER is the dialled number
and GW2=goip
On dialing a call, following logs appear on asterisk terminal
Call to peer 'goip' rejected due to usage limit of 1
-- Couldn't call SIP/913579865476#goip
Please guide in this regard
Acordinly to message you have call-limit in sip.conf set to 1 for this section
Also you need at least 2 gsm modules to do that.
I am working with an asterisk software pbx.
I have an IP phone which is configured with asterisk.
What i want to do is make call to a special number when the user hangs the phone. I do not want him to have to compose a number.
Do you know if it possible ?
Thanks
You have call, part A(caller) call to party B(called).
So.
You can setup your dialplan if B hangup, asterisk connect A with next number.
You can't setup your dialplan if A hangup, B connected to other number, except variant when you do connect A&B via conference(not via Dial command).
If you want phone call B when A get it, that called HOTLINE and it is feature of phone, not asterisk.
How to recognize a DTMF option as "123"(more than a digit) in IVR bot application.Basically DTMF accepts like press 1,2,3 not as "123".why?
Your question is not clear.
Can you tell us more about your 'IVR bot' application and its limits ?
On standard IVR systems it should be easy to manage the 2 cases (digit by digit or get a number/digits sequence).