Oh... I looked at that before, but I don't see how to play a warning
before the caller is disconnected with TIMEOUT?
On Sat, 28 Jul 2018 at 23:05, Social Boh wrote:
>
> TIMEOUT function:
>
> example
>
> same => n,Set(TIMEOUT(absolute)=600)
>
> after 600 seconds Asterisk Hankup the call
>
>
TIMEOUT function:
example
same => n,Set(TIMEOUT(absolute)=600)
after 600 seconds Asterisk Hankup the call
Regards
---
I'm SoCIaL, MayBe
On 7/28/18 16:08, Jonathan H wrote:
Last question for today, I promise!
The problem: In order to disconnect calls after x minutes, I need to do this:
On Sat, Jul 28, 2018, at 6:28 PM, Jonathan H wrote:
> OK, thanks. Shall I file a ticket to get that example file updated?
Sure!
--
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
--
On Sat, Jul 28, 2018, at 6:27 PM, Jonathan H wrote:
> Thanks, but... whoah! I think I just found a bug!
>
> As soon as I changed
> accepts_registrations = yes
> to
> sends_registrations = yes
>
> and did a pjsip reload, Asterisk crashed. I tried starting asterisk.
> Nothing. In the syslog:
>
>
OK, thanks. Shall I file a ticket to get that example file updated?
On Sat, 28 Jul 2018 at 21:50, Joshua Colp wrote:
>
> On Sat, Jul 28, 2018, at 5:42 PM, Jonathan H wrote:
> > I'm trying to configure sip2sip, which says:
> > http://wiki.sip2sip.info/projects/sip2sip/wiki/SipDevicesAsterisk
> >
Thanks, but... whoah! I think I just found a bug!
As soon as I changed
accepts_registrations = yes
to
sends_registrations = yes
and did a pjsip reload, Asterisk crashed. I tried starting asterisk.
Nothing. In the syslog:
Jul 28 22:20:41 televox kernel: [ 50.728769] asterisk[1504]:
segfault at
Last question for today, I promise!
The problem: In order to disconnect calls after x minutes, I need to do this:
[setup]
exten => setup,1,Answer()
same => n,Set(LIMIT_PLAYAUDIO_CALLER=yes)
same =>
n,Set(LIMIT_WARNING_FILE=/var/lib/asterisk/sounds/en_GB_TNS/time_limit_reached)
same
On Sat, Jul 28, 2018, at 3:06 PM, Jonathan H wrote:
> Using pjsip 2.7.2 on Asterisk 15.5
> Really struggling to make sense of translating these old 1.8 SIP
> instructions into a neat pjsip_wizard conf suitable for 2018
>
On Sat, Jul 28, 2018, at 5:42 PM, Jonathan H wrote:
> I'm trying to configure sip2sip, which says:
> http://wiki.sip2sip.info/projects/sip2sip/wiki/SipDevicesAsterisk
> "Asterisk, is currently unable to handle more that one result for a
> DNS SRV lookup, and the Asterisk configuration needed for
I'm trying to configure sip2sip, which says:
http://wiki.sip2sip.info/projects/sip2sip/wiki/SipDevicesAsterisk
"Asterisk, is currently unable to handle more that one result for a
DNS SRV lookup, and the Asterisk configuration needed for getting it
work with the SIP2SIP service is not trivial"
It
I've not needed to do a dialplan reload for a while, so I don't know
exactly which version is stopped working, but on 15.5, I'm not seeing
ANY debug info at any debug level.
So I'm not really sure how to find mistakes in the dialplan. This is
all I get... how do I enable this debug mode to see
Using pjsip 2.7.2 on Asterisk 15.5
Really struggling to make sense of translating these old 1.8 SIP
instructions into a neat pjsip_wizard conf suitable for 2018
http://wiki.sip2sip.info/projects/sip2sip/wiki/SipDevicesAsterisk#Version-18
In pjsip_wizard.conf, I have the following, which seems to
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