Re: [asterisk-users] Asterisk only registering at one provider
Hey guys, thanks for your effort. I just replicated the typical problem: the problem sits in front of the monitor. (And I am so glad that my systems are not in production). The issue was that register = is *only* allowed in the [general]-section. But since I like a precise and clean configuration I put it like this: [general] register = SIP1 [SIP1] ... register = SIP2 [SIP2] ... And then the second register is ignored as it is not in [general]. However, no error messages are thrown... Best regards and a happy weekend! --- Dennis Guse --- Dennis Guse On Wed, Mar 18, 2015 at 10:19 PM, Joshua Colp jc...@digium.com wrote: Dennis Guse wrote: Hey, I am running default Asterisk 11.16.0 on a FreeBSD-Machine. I need to register to several other SIP-Services (actually 3): short sip.conf register = XX@a register = XX@b register = XX@c If I remember correctly this worked quite well, but I now checked the system again and it is only obeying the first register statement. sip show registry only reports the first entry and if I reorder them, this effect stays the same. Did something changed recently in the parsing code for sip.conf or so? Nope, and I'd expect we'd be seeing many bug reports if something like this was occurring. I just did this in my general section in 11: register = meh@tacos register = hola@bob register = yolo@dave And confirmed they appeared as expected: Hostdnsmgr Username Refresh StateReg.Time dave:5060 N yolo 120 Request Sent bob:5060N hola 120 Request Sent tacos:5060 N meh120 Request Sent Does anything show up on the console when chan_sip is loaded? -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk only registering at one provider
Hey, I am running default Asterisk 11.16.0 on a FreeBSD-Machine. I need to register to several other SIP-Services (actually 3): short sip.conf register = XX@a register = XX@b register = XX@c If I remember correctly this worked quite well, but I now checked the system again and it is only obeying the first register statement. sip show registry only reports the first entry and if I reorder them, this effect stays the same. Did something changed recently in the parsing code for sip.conf or so? --- Dennis Guse -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Echo Cancellation on VoIP networks
On VoIP echo cancellation is basically: hope that the client is doing AND is doing it well. In the best case each client uses a knowledge about his hardware (microphone, speaker, distance etc.). --- Dennis Guse On Tue, Aug 26, 2014 at 1:30 PM, Emiliano Vazquez emilianovazq...@gmail.com wrote: El 26/08/14 a las 05:33, Grant Bagdasarian escibió: I’m new to Echo Cancellation and I was wondering how it is handled/works on pure VoIP networks using Asterisk? there is no echo problems on pure VoIP networks. echo is a common problem when you have changes from analog to digital. The only echo problem you will have is when you call another network who has analog circuits with wrong configuration or poor hardware. But you can't solve it. Best regards. -- Emiliano Vazquez | PcCentro Informatica CCTV Office: +54 (11) 4635-3218 y Rotativas Movil: 011-15-6253-7165 Mail: emilianovazq...@gmail.com Web: http://www.pccentro.com.ar -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Strange Error
Sound like chan_sip was not build. Just a guess: check that openssl-dev is available --- Dennis Guse On Thu, Jul 3, 2014 at 12:01 PM, Andrew Colin and...@vsave.co.za wrote: Hi Guys, Does anyone know what this error means and how to fix it? [Jul 3 11:57:27] WARNING[17040] pbx.c: Don't know what to do with 'SIP/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to execute an AGI script for each call.
You could try either the predial-handler or the dial-macro M. --- Dennis Guse On Wed, Jul 2, 2014 at 3:06 PM, Joshua Colp jc...@digium.com wrote: Anurag Rana wrote: Hi All, Kia ora, I am trying to execute some AGI script no matter what extension is called. There is 'h' extension to call AGI script when any call hangs up no matter what extension hangup. for example - [some-context] /// something here which call AGI script no matter what extension receive call. exten = 111,1,Dial(SIP/111) exten = 112,1,Dial(SIP/112) exten = h,1,AGI(pt.py) ;; executes no matter what extension hang up Have your first priority be a pattern match of something like _X. which executes your AGI. Then have your second priority be the specialized logic (such as the Dial above). That should do what you want. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using macros in extensions.lua?
Got it:
extensions = {
[macro-test] = {
[s] = function(c, e)
app.verbose(This is my macro)
end;
};
default = {
[_X] = function(c, e)
app.dial(SIP/00, nil, mM(test))
end;
};
};
---
Dennis Guse
On Fri, Jun 6, 2014 at 6:49 PM, George Joseph george.jos...@fairview5.com
wrote:
On Fri, Jun 6, 2014 at 1:48 AM, Dennis Guse
dennis.g...@alumni.tu-berlin.de wrote:
Hi,
I have defined a dialplan in lua and now would like to use dial with
the macro M to implement some logic, when the callee-channel gets created.
Working old style would be (extensions.conf)
[default]
exten = _X,1,dial(SIP/1,,M(mymacro^parameter))
[macro-mymacro]
exten = s,1,verbose(${ARG1})
How to implement the same functionality using pbx_lua?
Details: Asterisk 11.7 on Ubuntu 14.04
Kind regards
Dennis Guse
Here's how I do it for pre-dial handlers...
extensions.handlers = {
[addheader] = function(c,e)
channel.PJSIP_HEADER('add', Alert-Info):set(;info=custom1)
end;
}
extensions.local_default = {
[] = function(c,e)
app.dial('PJSIP/'..e,nil,'b(handlers^addheader^1)')
end;
}
--
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[asterisk-users] Using macros in extensions.lua?
Hi,
I have defined a dialplan in lua and now would like to use dial with the
macro M to implement some logic, when the callee-channel gets created.
Working old style would be (extensions.conf)
[default]
exten = _X,1,dial(SIP/1,,M(mymacro^parameter))
[macro-mymacro]
exten = s,1,verbose(${ARG1})
How to implement the same functionality using pbx_lua?
Details: Asterisk 11.7 on Ubuntu 14.04
Kind regards
Dennis Guse
Quality and Usability Lab
Telekom Innovation Laboratories
TU Berlin
Ernst-Reuter-Platz 7
D-10587 Berlin, Germany
Tel: +49 30 8353 58874
Fax: +49 30 8353 58409
E-mail: dennis.g...@telekom.de
Web: www.qu.tlabs.tu-berlin.de
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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[asterisk-users] Indications.conf: change volume
Hi, I use Asterisk to create the dial tone (indications.conf), which works quite well. However the generated signal is quite loud at the client side (in comparison to the following speech ). Is there an option to modify the volume? --- Dennis Guse -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Indications.conf: change volume
I let Asterisk generate the ringtone: DIAL(SIP/XX, 'r')... --- Dennis Guse On Tue, Apr 22, 2014 at 4:47 PM, jg webaccounts...@jgoettgens.de wrote: The call invitation is only signaled in most cases. You need to check the settings of your phones. Hi, I use Asterisk to create the dial tone (indications.conf), which works quite well. However the generated signal is quite loud at the client side (in comparison to the following speech ). Is there an option to modify the volume? --- Dennis Guse -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
