Re: [asterisk-users] Voice recognition IVR Is it possible?

Read README, check the requirements and get the google speech api key. Then add a custom destination in FreePBX and edit your extensions_custom.conf. > Am 22.02.2016 um 21:03 schrieb Daniel Chavez : > > Ok. Where I am now is, I copied the speech-recog.agi to the agi-bin

Re: [asterisk-users] Voice recognition IVR Is it possible?

I use FreePBX as well. There is no module for speech recognition. You have too create a custom destination. > Am 22.02.2016 um 20:53 schrieb Daniel Chavez : > > Thanks, this looks promising. I was wondering if there's an easier way to get > this to work inside FreePBX? >

Re: [asterisk-users] Voice recognition IVR Is it possible?

Daniel, try this http://zaf.github.io/asterisk-speech-recog/. I have tested it myself, it works very well. Daniel > Am 22.02.2016 um 19:34 schrieb Daniel Chavez : > > Thanks for the link. > Are there no free alternatives for speech recognition? > > -- >

Re: [asterisk-users] PJSIP Stun/ICE

Bryant, I have the same problem with dynamic public IPs and PJSIP. What is your idea to solve the problem? My suggestion would be to write a script that monitors the change, pjsip.transports.conf updated and Asterisk restarts? Daniel > Am 26.01.2016 um 14:21 schrieb Joshua Colp

Re: [asterisk-users] PJSIP Stun/ICE

ntext. > > Thanks > > Bryant > > From: "Daniel Heckl" <daniel.he...@gmail.com> > Sent: Tuesday, January 26, 2016 10:15 AM > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users@lists.digium.com> > Subject

Re: [asterisk-users] Transfer calls "on demand"

You are searching for „Call Pickup“. It is implemented in Asterisk by default. https://wiki.asterisk.org/wiki/display/AST/Call+Pickup Take a look under section „Configuration Options“. Daniel > Am 29.12.2015 um 07:53 schrieb Luca

Re: [asterisk-users] Transfer calls "on demand"

On top of the page: "Call pickup support added in Asterisk 11“ I think that is the problem. I do not know a solution for 1.8, but maybe someone other. > Am 29.12.2015 um 10:20 schrieb Luca Bertoncello <lucab...@lucabert.de>: > > Daniel Heckl <daniel.he...@gmail.

Re: [asterisk-users] Update peer IP address

Sebastian, If I have understood you correctly, the SIP communication is now via NAT instead forwarded ports. For safety, it is much better. I think it is not because of a UDP timeout, but rather because of a NAT timeout. For this is "qualify" exactly the right thing to let the NAT port

Re: [asterisk-users] Update peer IP address

PM, Daniel Heckl daniel.he...@gmail.com mailto:daniel.he...@gmail.com wrote: Scott, I have changed the configuration as said it and will test it. I’m curious. Can you briefly explain what insecure=invite,port does? ;insecure=port ; Allow matching of peer by IP address without

Re: [asterisk-users] Update peer IP address

(with insecure=port,invite). If the provider is requiring you to accept invites from random IP addresses, get a new provider. On Thu, Apr 2, 2015 at 3:23 PM, Daniel Heckl daniel.he...@gmail.com mailto:daniel.he...@gmail.com wrote: Okay, Scott, I think we are on the wrong path. Maybe I'm

Re: [asterisk-users] Update peer IP address

01.04.2015 um 19:23 schrieb Sebastian Kemper sebastian...@gmx.net: On Wed, Apr 01, 2015 at 11:00:56AM -0400, Andres wrote: On 4/1/15 10:48 AM, Daniel Heckl wrote: John, thank you four your answer. I think you have misunderstood the problem. It’s about a ip address change of the sip trunk

Re: [asterisk-users] Update peer IP address

=invite,port makes any difference either (without alllowguest on). ​ On Thu, Apr 2, 2015 at 9:03 AM, Daniel Heckl daniel.he...@gmail.com mailto:daniel.he...@gmail.com wrote: Ok, I have tested dnsmgr. This is not a solution, the situation has not changed. With dnsmgr I can not place

Re: [asterisk-users] Update peer IP address

has changed via whatever method available, and then issue a sip reload CLI command to asterisk, which will cause it to resend registrations immediately. On Tue, Mar 31, 2015 at 1:36 PM, Daniel Heckl daniel.he...@gmail.com wrote: Maybe someone could elaborate on my first question again

Re: [asterisk-users] Update peer IP address

John, thank you four your answer. I think you have misunderstood the problem. It’s about a ip address change of the sip trunk, not of my asterisk server. Kind regards, Daniel Am 01.04.2015 um 16:40 schrieb Tech Support aster...@voipbusiness.us mailto:aster...@voipbusiness.us: If I

Re: [asterisk-users] Update peer IP address

30.03.2015 um 20:09 schrieb Sebastian Kemper sebastian...@gmx.net: On Mon, Mar 30, 2015 at 06:31:46PM +0200, Daniel Heckl wrote: Hello I use Asterisk 11 with FreePBX 12. Our SIP Provider is Telekom Germany. We have sometimes problems with incoming and outgoing calls. I hope I can explain

Re: [asterisk-users] Update peer IP address

Maybe someone could elaborate on my first question again. If the ip address changes while a REGISTER period, the ip address of the peer isn't been updated. How can asterisk update the ip address of the peer? Am 31.03.2015 um 12:36 schrieb Daniel Heckl daniel.he...@gmail.com: Hello Sebastian

[asterisk-users] Update peer IP address

Hello I use Asterisk 11 with FreePBX 12. Our SIP Provider is Telekom Germany. We have sometimes problems with incoming and outgoing calls. I hope I can explain it understandable. For example, Asterisk sends a REGISTER to 217.0.23.68 (tel.t-online.de http://tel.t-online.de/), the message is