Hi,
I am also trying to integrate sipml5 demo.For that i made some
configuration.
Call works fine using chrome browser but facing "One way audio issue".
And firefox browser not able to originate call.
Here is the my configuration: http://pastebin.com/EtVzK2T2
let me know if i miss something.
Hi,
I'm trying to install voicemail-imap support but there seems to be a
missing module:
imap_tk
checking for mandatory modules: IMAP_TK... fail
configure: ***
configure: *** The IMAP_TK installation appears to be missing or broken.
configure: *** Either correct the installation, or run conf
Please check rtp.conf
Look for stunaddr setting. You can try with google STUN server
stunaddr = stun.l.google.com:19302
*Thanks & Regards,*
Amit Patkar
On 5/21/2014 9:13 PM, Gary Shergill wrote:
Hi again,
Just noticed this is being sent to the wrong thread... first time using a
mailing list
Hi again,
Just noticed this is being sent to the wrong thread... first time using a
mailing list and I just replied to the mail sent by the mailing list for Amit's
reply. Hope this time it works...
Anyway, I have audio from 1000 to 6901 working, that was a mistake on my side
(I tested using th
Hi Amit,
ICE/STUN is configured correctly. The extension for the webrtc user is defined
in sip.conf on the asteriskrtc.local server. The other user is defined in
Elastix.
I have "directmedia=no" set for the user on asteriskrtc.local.
My exact setup/scenario is below:
- asteriskgary.local has a
> Here are links to the Asterisk Wiki for CDR and SIP tables. I
> didn't find extensions listed, but it's pretty simple and I can
> provide the structure for that if needed, but it would be without a
> definitive source beyond me having used it for years. :-)
I think the problem with those li
Here are links to the Asterisk Wiki for CDR and SIP tables. I didn't find
extensions listed, but it's pretty simple and I can provide the structure
for that if needed, but it would be without a definitive source beyond me
having used it for years. :-)
https://wiki.asterisk.org/wiki/display/AST/M
Hi Gary
You need to check if ICE / STUN is configured.
How are these extensions configured? If you are in private network, you
might have to disable DirectMedia / reInvite for calls going between 2
asterisk boxes.
I hope this helps to resolve your issue.
*Thanks & Regards,*
Amit Patkar
On 5
Hi,
we implemented ispeech for voice recognition. I works fine. But you have
to develop an app of your own to do it.
Take a look at http://www.ispeech.org/api (Section 3 Automated Speech
Recognition).
ispeech let you upload a recorded speex file via http-upload and will
return the result a
Hi,
I've run into a slight issue when using WebRTC and two Asterisk boxes.
I am using SIPml as the test WebRTC client.
My two asterisk boxes, one of them is configured for WebRTC with websockets,
etc (asteriskrtc.local) and the other is just a standard asterisk server
(asteriskgary.local).
De
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