I have it working now!
*I had to install Asterisk 13 with PJSIP support.That's important, even if
you won't use PJSIP.* Then I did this configuration, which is working fine
under NAT:
*sip.conf:*
[6001]
type=friend
secret=REDACTED
host=dynamic
context=interno
disallow=all
I had been using google tts, but it started requiring a captcha for my
browser, and via linux I can't access
http://translate.google.com/translate_tts?q=test (redirects to captcha)
as so, its not reliable
On 27 August 2015 at 17:16, Carlos Chavez cur...@telecomabmex.com wrote:
On 8/26/15 1:15
On Thu, 27 Aug 2015, Tiago Geada wrote:
I had been using google tts, but it started requiring a captcha for my
browser, and via linux I can't
access http://translate.google.com/translate_tts?q=test (redirects to
captcha)
I'm confused. Your subject says 'speech to text' but the URL you
On 8/26/15 1:15 PM, Tech Support wrote:
All;
I have a customer who is looking for a good speech to text
solution, either open source or reasonably priced commercial product,
I’m open to suggestions.
Thanks;
John V
For a commercial option try Lumenvox, had very good results. For free
hi
you can try this link
http://zaf.github.io/asterisk-googletts/
2015-08-26 19:15 GMT+01:00 Tech Support aster...@voipbusiness.us:
All;
I have a customer who is looking for a good speech to text solution,
either open source or reasonably priced commercial product, I’m open to
Thanks Scott.
I’m taking over for someone else’s code, so I must admit I’m still learning the
Agent and Queue concepts. Local channels are something I have not used either.
Would local channels essentially be an internal bridge?
How would I
“Register Local/number@agent in the queue on behalf
I have a polycom phone behind a firewall.
The phone registers - but I only hear half channel audio.
I have tried nat=yes, nat=force_rport,comedia and
nat=autio_force_rport,auto_comedia (reloading asterisk every time).
made no difference.
How might I get full audio path?
Thanks,
Jerry
--
I have a call coming in.
I need to add a SIP Header to the channel.
Then, I need to send the call to the Queue so it is sent to the Agent.
The SIP header I added, I need to have appear in the INVITE sent to the Agent.
It works in chan_sip. I send the call to a macro which does...
To add a header to the call leg that goes to the agent, try using a local
channel to activate dialplan on the outbound call:
Register Local/number@agent in the queue on behalf of the agent (replace
number with the agent's extension number)
In dialplan [agent], wild card match the number, add the
Are you using this method of setting headers on PJSIP?
https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+Function_PJSIP_HEADER
On Thu, Aug 27, 2015 at 4:54 PM, Dan Cropp d...@amtelco.com wrote:
Thanks Scott.
I was able to get the basic concept to run.
However, it seems PJSIP INVITE
Local channels:
http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/DeeperDialplan_id324598.html
This explains adding members to queues, although it doesn't specifically
provide an example of using local channels in a queue:
Thanks Scott.
I was able to get the basic concept to run.
However, it seems PJSIP INVITE for the Dial also does not support added headers.
The Local channel dial plan did have the channel variable values. I added them
as SIP headers, then Dial(PJSIP/Agent).
The INVITE for the Dial on PJSIP
Dne 13.8.2015 v 21:48 Marek Červenka napsal(a):
Dne 13.8.2015 v 17:20 Rusty Newton napsal(a):
On Thu, Aug 13, 2015 at 3:54 AM, Marek Červenka cerv...@fpf.slu.cz
mailto:cerv...@fpf.slu.cz wrote:
hello,
is it possible simultaneously use chan_sip and chan_pjsip?
if yes, can you
On 15-08-27 07:33 AM, Marek Červenka wrote:
Dne 13.8.2015 v 21:48 Marek Červenka napsal(a):
Dne 13.8.2015 v 17:20 Rusty Newton napsal(a):
On Thu, Aug 13, 2015 at 3:54 AM, Marek Červenka
mailto:cerv...@fpf.slu.czcerv...@fpf.slu.cz wrote:
hello,
is it possible simultaneously use
Dne 27.8.2015 v 12:37 Joshua Colp napsal(a):
On 15-08-27 07:33 AM, Marek Červenka wrote:
Dne 13.8.2015 v 21:48 Marek Červenka napsal(a):
Dne 13.8.2015 v 17:20 Rusty Newton napsal(a):
On Thu, Aug 13, 2015 at 3:54 AM, Marek Červenka
mailto:cerv...@fpf.slu.czcerv...@fpf.slu.cz wrote:
hello,
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