search for: greybeam

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2019 Apr 10
7
Forking AGI or GoSub
I have an AGI that can sometimes take time complete. I don't want the dialplan to be held up by the agi. Is there any way to call it and have Asterisk continue with the dialplan? -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20190410/4c704231/attachment.html>
2019 Apr 19
2
Forking AGI or GoSub
...e: > On 4/19/2019 1:49 PM, Dovid Bender wrote: >> Mark, >> >> I am using PHP agi and when forking the call does not continue util >> the forked process is done. Am I doing it wrong? >> >> >> On Wed, Apr 10, 2019 at 4:27 PM Mark Wiater <mark.wiater at greybeam.com >> <mailto:mark.wiater at greybeam.com>> wrote: >> >> On 4/10/2019 3:54 PM, Dovid Bender wrote: >>> I have an AGI that can sometimes take time complete. I don't want >>> the dialplan to be held up by the agi. Is there any way to call...
2008 Oct 23
1
Returning to Voicemail after returning call
Hello all, I've got dialout= and callback= set in my voicemail.conf so that I can have users return calls to folks who have left messages. They really like this feature. But when the callback is over, a normal hangup occurs instead of the caller being put back into voicemail at the next message. Is it possible that the users be returned into the voicemail system where they left off? thanks
2017 Jul 12
2
Copying received and sent RTP packets due legal obligations
Hi, I am facing a problem where for legal obligations (LI) I have to copy/mirror/forward the RTP streams for some selected call to an external address/port and I have not found a way to do it with built-in functionality. Do I miss something? The basic requirements are: * Raw RTP (no transcoding, header and payload as is) * Direction (did it arrive at asterisk or was it sent) * End
2016 Feb 22
4
Windstream SIP Trunk settings
Does anyone on this list use Windstream as a SIP trunk provider? If so, would you mind sharing your peer settings? I'm using asterisk 13.7.2 and can't seem to get the inbound working correctly (using registration). Outbound is fine, but they are seeing an authentication error on their end. Here are my inbound peer settings: username=<accountnumber> secret=<secret>
2017 Jan 16
4
How to send SIP_NOTIFY messages with variable content ?
Hello, One common mean to remotely configure a phone is to send it some XML data using HTTP. Of course, this XML data is vendor specific but thanks to Asterisk multiple tools, it is quite easy to customize your dialplan to both build and send this specific XML data. I have just discovered one interesting capability from one phone vendor: getting XML data from incoming SIP NOTIFY messages instead
2017 Jul 19
4
Integration of Google Speech API V2
Hi, I'm trying to integrate Google cloud speech recognition v2 in it. I can get the audio recorded, have created Service key and API key but whenever I try to access it, I just get 403 access denied. I am at my wits end here. Has anybody tried it ? were you successful ? Could you please guide me how to do it ? I'll be grateful to you if this works ! -- Warm Regds. MathuRahul
2017 Apr 18
3
SIP connections over OpenVPN connection get one-way voice.
You need to ensure that traffic to the SIP box is sent to the correct IP. Also if you use split-tunnel (eg: not redirect-gateway def1) you must make sure NAT and traffic redirection works as is so the Asus router knows it should send the traffic through tunnel and not via WAN. IMPORTANT: Then you must, in the ASUS RT-N66U make a port forward inwards from TUN to the phone client. I would suggest
2016 Aug 30
12
Multiple phones when one is unregistered
I have an extension that looks like this: exten => 5555551111,1,Verbose(Door buzzer calling) same => n,Dial(SIP/user1&SIP/user2&SIP/user3) The idea is that any of the three users can answer the phone to let someone in. The problem is that if, say, user2 unplugs his phone then the call immediately goes to his voice mail and the other two do not have the ability to open the door.
2015 Jun 18
0
setting outbound caller ID
On 6/18/2015 1:27 PM, Greg Woods wrote: > My provider claims that I am somehow sending an old number that > doesn't appear anywhere <snip> > (I just moved from a POTS provider Century Link to a VOIP provider). Set(CALLERID(number)=${var}) works fine for me. Perhaps some debugging on the channel could help? Set sip debug if it's a sip trunk? You'll at least get to