[asterisk-users] asterisk not transcoding on a re-invite codec change
I have an old setup based in Asterisk 1.8. The carrier is accepting Ulaw in the initial invite, but immediately the call is established they send a re-invite to change to Alaw. This doesn't get transcoded and the user gets no audio from after the re-invite Is/was this a problem for asterisk changing the codec and transcoding during the call? The user is only offering ulaw. Regards, rik -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can anyone help with a quick app_record.c module improvement and can explain over-riding modules?
Hi Dmitry and Tim (and everyone else with input into this thread)
Just wanted to thank you all; with your guidance, I've managed to bolt
something very clean and efficient together using Dmity and Tim's
templates, piped into the ding-dong npm node package, which calls Google
Speech API node package.
What I particularly like about DingDong is that it's well documented
insofar as it's so simple, it barely needs documentation!
Once I've finished experimenting in a day or two, I'll stick a Gist up with
everything together.
Once again, many, many thanks.
Refs:
https://github.com/antirek/ding-dong
https://github.com/googleapis/nodejs-speech
On 21 January 2018 at 08:30, Dmitriy Serov wrote:
> Hello.
>
> A little sub from my dialplan:
>
> [sub-Read]
> exten => s,1,NoOp(Read)
> same => n,Set(LOCAL(tmp_record_file)=/tmp/asterisk-in/${EPOCH})
> same => n,Monitor(wav16,${tmp_record_file},o)
> same => n,Read(tmp_ext,${ARG2},${ARG3},${ARG4},${ARG5},${ARG6})
> same => n,StopMonitor()
> same => n,NoOp(ReadStatus=${READSTATUS})
> same => n,Gotoif($[ ${LEN(${tmp_ext})} > 0 ]?end)
> same => n,AGI(agi-ruvoip-net.php,speeddial-voice,${ARG1},${tmp_recor
> d_file}-in.wav16)
> same => n,NoOp(Voice recognition result: "${agi_result}")
> same => n,Gotoif($[ "${agi_result}" != "found" ]?end)
> same => n,Return(${agi_call_exten})
> same => n(end),return(${tmp_ext})
>
>
> 21.01.2018 2:57, Jonathan H пишет:
>
> On 20 January 2018 at 23:30, Tim S wrote:
>>
>> I have seen this take over 2 seconds before on a sluggish machine.
>>>
>> Thanks - my host uses SSD and everything seems pretty quick, but I'll
>> give it a 1 second pause.
>>
>> you'd need to pipe that to a Google Speech API tunnel.
>>> That's probably not something you can hack away at with simple
>>> Asterisk dialplan applications.
>>>
>> Funnily enough, I had just found an old reply from last year to
>> another similar question:
>>
>> -- Forwarded message --
>>> From: Matt Riddell
>>> Date: 22 September 2017 at 16:01
>>> Subject: Re: [asterisk-users] Asterisk 15, Jack, streams, speech
>>> recognition… so many questions!
>>> At least in older versions you can use EAGI to get a handle to the audio
>>> stream.
>>>
>> So I had a look and found this:
>>
>> https://stackoverflow.com/questions/34026698/asterisk-write-
>> plugin-to-catch-voice-stream
>>
>> And read this:
>>
>> https://wiki.asterisk.org/wiki/display/AST/Asterisk+15+Application_EAGI
>>
>> There's a few knowledge gaps, but I think with a few days reading and
>> the great help here, we might have a solution :)
>>
>> This is all very helpful - if anyone else feels like wading in, please do.
>>
>> Many thanks!
>>
>>
>
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>
> Check out the new Asterisk community forum at:
> https://community.asterisk.org/
>
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> https://wiki.asterisk.org/wiki/display/AST/Getting+Started
>
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[asterisk-users] Forwarding a call "off pbx"
All; I had someone ask me if they received an incoming phone call and it was forwarded "off pbx" to their cell phone, would the call be strictly between the caller and the cell phone, or would it between the caller, the pbx, and the cell phone where the VoIP minutes continue to be charged. Rather than pull an answer out of my butt, I tested it and found that it's a three-legged call which really surprised me. So my question is, is that the correct behavior? Thanks Again; John V. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Forwarding a call "off pbx"
On Tuesday 23 January 2018 at 14:43:21, Tech Support wrote: > All; > > I had someone ask me if they received an incoming phone call and it was > forwarded "off pbx" to their cell phone, would the call be strictly between > the caller and the cell phone, or would it between the caller, the pbx, and > the cell phone where the VoIP minutes continue to be charged. Rather than > pull an answer out of my butt, I tested it and found that it's a > three-legged call which really surprised me. So my question is, is that the > correct behavior? Yes. Think about the call flow: 1. You have an account with a VoIP provider, and some incoming numbers which, if someone in the world dials them, will result in a SIP call from the provider to your PBX. 2. Your PBX knows that this particular user wants their calls to be forwarded to their mobile number, and they've told you what that number is. 3. Your PBX can make calls to the PSTN over some connection or other - it might be with the same provider as in (1) or it might be with a different provider. If you do Least Cost Routing, you'll have several providers you can make calls over. 4. So, a call comes in to your PBX, which then forwards it out over the PSTN to the mobile phone. There's no way your PBX can tell the inbound VoIP provider to forward that call instead, and anyway, why would they do this - it then costs them money to place the call to the PSTN instead of you. So, the VoIP provider doesn't know what your user's mobile phone number is, and isn't going to pay to place calls to it anyway. They have a contract with you to send calls to your PBX. That's all they do. Hope that helps, Antony. -- "If I've told you once, I've told you a million times - stop exaggerating!" Please reply to the list; please *don't* CC me. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
