On Thu, Aug 20, 2015 at 10:52 AM, Bastian Schern m...@reventix.de wrote:
Hello Everybody,
Howdy
in past times I used macros but since a while they are deprecated.
So I replaced my macros with subroutines. In most cases this is really no
problem.
But in some rare cases I miss the macro
You need to:
1. Have a systemname = whatever in asterisk.conf under [options]
2. Set rtsavesysname=yes in sip.conf under [general]
3. Have a column called “regserver” in the SIP peer database
Source:
Hi,
Curious why didn’t you try AQuA http://sevana.biz/products/aqua/ to score
the quality? Using voice files for tests has more representation to my
opinion.
Thanks,
vallu
On Thu, Aug 20, 2015 at 4:11 AM, Pete Mundy p...@fiberphone.co.nz wrote:
Markus
That's a fascinating concept!
Can
Greetings everyone,
I am attempting to adjust the volume of a call using Set(VOLUME) in my
extensions.conf file. I am finding that Set(VOLUME(TX)=x) and Set(VOLUME(RX)=y)
have no discernable effect on my endpoints (Snom 300 IP phones). I have tried
setting x and y to -30, -10, -3, -2, -1, 0, 1,
Am 20.08.2015 um 03:16 schrieb Pete Mundy:
Ah cr@p, sorry Steve, didn't mean to top-post there.
On 20/08/2015, at 5:23 AM, Markus Weiler markus_wei...@mailworks.org
mailto:markus_wei...@mailworks.org wrote:
We started the 500 calls and used milliwatt app on the first and
record on the
On 20 Aug 2015, at 11:12, Sevana Oy sa...@sevana.fi wrote:
Curious why didn’t you try AQuA http://sevana.biz/products/aqua/ to score
the quality? Using voice files for tests has more representation to my
opinion.
Spot the salesman? ;)
Steve--
Hello,
I have what I would think would be a common situation: I run asterisk at
home simply as a land line. I started a new job working remotely and
they gave me a SIP account with user name, domain, and proxy. I've never
had to deal with sip domains before. My user '1...@4354766787.com' is
Hi Barry Flanagan,
Dominique Haeber dominique.hae...@xig.ch schrieb am Mit, 19. Aug 15:13:
Barry Flanagan barryf-li...@flanagan.ie schrieb am Mit, 19. Aug 11:06:
SIPP is probably what you seek. http://sipp.sourceforge.net/
Hope this helps.
That looks pretty like what I'm looking for!
Hello Everybody,
in past times I used macros but since a while they are deprecated.
So I replaced my macros with subroutines. In most cases this is really
no problem.
But in some rare cases I miss the macro channel variables (e.g.
${MACRO_CONTEXT}).
I am running Asterisk 13.5.0.
I have the Transfer working when using the chan_sip support.
However, when I try to perform a Transfer using pjsip, it is failing.
The one difference I am seeing in the SIP trace is chan_sip automatically sends
the Referred-By. PJSIP does not.
The switch provider
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