Re: [asterisk-users] Multiple readfile oddities, newlines etc

On 18/10/2016, at 10:38 am, Steve Edwards wrote: >> cat /home/test/feature-1.txt | hexdump > > Or just: > > hexdump /home/test/feature-1.txt Heh.. yes, fair call ;) Pete smime.p7s Description: S/MIME cryptographic signature --

Re: [asterisk-users] Streaming for ASR

On Mon, 17 Oct 2016 18:44:39 -0300 Joshua Colp wrote: > >> > >> The UnicastRTP channel driver allows you to send RTP to a specific > >> target address with media. Combined with Chanspy (or Snoop channels in > >> ARI) you can duplicate audio from a channel and send it off to

Re: [asterisk-users] Streaming for ASR

Matt Riddell wrote: On 17/10/2016, at 4:07 PM, Joshua Colp > wrote: Matt Riddell wrote: On 17/10/2016, at 3:43 PM, Luca Pradovera > wrote: I have been

Re: [asterisk-users] Multiple readfile oddities, newlines etc

On Tue, 18 Oct 2016, Pete Mundy wrote: If you want to know what is _really_ in that file (including all invisible characters and anything else that wc etc might not count), pipe it through 'hexdump'. cat /home/test/feature-1.txt | hexdump Or just: hexdump /home/test/feature-1.txt

Re: [asterisk-users] Streaming for ASR

> On 17/10/2016, at 4:07 PM, Joshua Colp wrote: > > Matt Riddell wrote: >> >>> On 17/10/2016, at 3:43 PM, Luca Pradovera >> > wrote: >>> >>> I have been working on designs for two different projects, where both >>>

Re: [asterisk-users] Multiple readfile oddities, newlines etc

> On 18/10/2016, at 2:31 am, Jonathan H wrote: > > I have a plain text file, ASCII, unix line breaks. 1 single line, and all > that is in it is the word "radio". > > Heya Jonathan Interesting problem! Unfortunately I can't help with suitable dialplan code to resolve

Re: [asterisk-users] Streaming for ASR

Matt Riddell wrote: On 17/10/2016, at 3:43 PM, Luca Pradovera > wrote: I have been working on designs for two different projects, where both of them would need to use the IBM Watson streaming ASR service. Would it be possible to

Re: [asterisk-users] SIP on multiple ports

> On 18/10/2016, at 12:30 am, Jerry Geis wrote: > > I am running iptables on the 10.201 machine. I have not control over the > other machine. It is a microsoft lync product. > > my definition... > [MyTrunk] > type=friend > dtmfmode=rfc2833 > disallow=all > allow=ulaw >

Re: [asterisk-users] Streaming for ASR

> On 17/10/2016, at 3:43 PM, Luca Pradovera wrote: > > I have been working on designs for two different projects, where both of them > would need to use the IBM Watson streaming ASR service. > > Would it be possible to write out the audio frames as they get recorded?

[asterisk-users] Streaming for ASR

Hello, I have been working on designs for two different projects, where both of them would need to use the IBM Watson streaming ASR service. Based on our discussion at AstriDevCon, I know there is currently no support for that. However, there may be some workarounds I am not aware of. Would it

Re: [asterisk-users] Registered successfully, but after a minute or so no SIP messages anymore

Hi, I made some interesting observations regarding this. Remind the following scenario: asterisk registers number A towards provider A (sipgate.de) asterisk also registers another number towards provider B (tel.t-online.de) I make a test call from a remote location, which is registered as well

Re: [asterisk-users] PJSIP how to change the generated SIP CALL ID

Saint Michael wrote: ​I need to change the automatically generated SIP call ID, from (example) 64f61c7a-c68f-498b-8661-b42e5c771363 to 64f61c7a-c68f-498b-8661-b42e5c771...@my.ip.add.ress since that is the only way to make sure the calls came from my box. How do configure this in the system? It

[asterisk-users] PJSIP how to change the generated SIP CALL ID

​I need to change the automatically generated SIP call ID, from (example) 64f61c7a-c68f-498b-8661-b42e5c771363 to 64f61c7a-c68f-498b-8661-b42e5c771...@my.ip.add.ress since that is the only way to make sure the calls came from my box. How do configure this in the system? It should be

Re: [asterisk-users] Surfing the web via Asterisk.

I'm guessing you're going to be wanting something closer to this: https://www.npmjs.com/package/speech-rule-engine -- Cheers, Matt Riddell ___ http://www.venturevoip.com/news.php (Daily Asterisk

Re: [asterisk-users] Surfing the web via Asterisk.

On 17 October 2016 at 16:12, Matt Riddell wrote: > https://www.npmjs.com/package/speech-rule-engine Thanks. That and the tip about jackaudio look interesting, although that thing above is just a parser, not a renderer. I think, at this stage, it's an idea to go back in

Re: [asterisk-users] Surfing the web via Asterisk.

> On 17/10/2016, at 9:51 AM, Jonathan H wrote: > > Ah, no, you misunderstand. Asterisk wouldn't care one little bit what > is on the page - Chromevox would do all that. > A screenreader usually tabs or arrows their way about, selecting > headings to read content. > >

[asterisk-users] Tips, tools and a question about debug level....

Lots of little bits in one email to save polluting the list too much today, time for me to try and give a little back, too! Someone posted about sngrep a couple of days ago. What a great tool! Is there a list of useful stuff like this that isn't hopelessly out of date? Talking about hopelessly

Re: [asterisk-users] Surfing the web via Asterisk.

Ah, no, you misunderstand. Asterisk wouldn't care one little bit what is on the page - Chromevox would do all that. A screenreader usually tabs or arrows their way about, selecting headings to read content. Thus, Asterisk ONLY needs to be able to hear content FROM the browser and pipe it to the

Re: [asterisk-users] Multiple readfile oddities, newlines etc

Just by chance, I was browsing the mailing list and it looks like Sebastian's reply made it to the list, but not to me because it's apparently blacklisted by UCEPROTECTL3. This is what he wrote: > Theres always garbage in the end of the files. > I do this when I want to read a file: > same

Re: [asterisk-users] Surfing the web via Asterisk.

This is a really interesting project but I think it's going to be seriously hard. You're going to need to parse meaning from a site, and that's not an easy thing to do. If you're focused on a few of the bigger sites then it might be easier. You almost want a middle layer that can parse

[asterisk-users] Surfing the web via Asterisk.

Has anyone attempted making the web phone accessible? I can only find one company which operated between 1996 and 2000. I was thinking, install Chrome with Chromevox, headless, on a server, and use something like an AGI to send basic keyboard commands to navigate a page, as a screenreader user

Re: [asterisk-users] Multiple readfile oddities, newlines etc

Theres always garbage in the end of the files. I do this when I want to read a file: same => n,Set(featurefile=/home/test/feature-1.txt) same => n,Set(unfilteredfeat2=${FILE(${featurefile},0,1,l,u)}) same => n,Set(feature2=${SHIFT(unfilteredfeat2)}) After that, add a , inside

[asterisk-users] Multiple readfile oddities, newlines etc

I have a plain text file, ASCII, unix line breaks. 1 single line, and all that is in it is the word "radio". Here's some test dialplan: exten => 5,1,Verbose(Context: ${CONTEXT} Exten:${EXTEN}) same => n,Set(feature=${FILE(/home/test/feature-1.txt,0,1,l,u)}) same => n,Verbose(${feature})

Re: [asterisk-users] SIP on multiple ports

>Jerry has already clarified in a previous reply that he is running SIP over TCP, not UDP. >But he hasn't clarified on which machine he is applying the iptables header rewrite rules (10.201, or 1.3?). >Either way though, it seems like a kludgy work-around. IMO, it'd be better to focus on creating