[asterisk-users] Don't Miss Out - AstriCon is NEXT MONTH!

If you, like me, have a tendency to leave things until the last minute, you may be in danger of missing out on AstriCon this year - as it is a little earlier than usual, September 27-29 in Glendale, AZ. www.astricon.net Three days of nothing but Asterisk-themed talks, demos and Expo exhibits are

Re: [asterisk-users] Dial and start music on hold after timeout

Thank you for the idea. The problem with RetryDial, is that it will cancel the first call, play the announce and then dial the SIP peer once again, so the telephone will display a missed call. I would prefer to do everything in a single call. Le 22/08/2016 à 17:57, John Kiniston a écrit :

Re: [asterisk-users] Dial and start music on hold after timeout

You could try using RetryDial() instead of Dial, It supports playing an announcement. On Mon, Aug 22, 2016 at 8:45 AM, Jean Aunis wrote: > Sorry, I forgot to write that the SIP peer must keep ringing while the > announcement is being played. > > Le 22/08/2016 à 17:42,

Re: [asterisk-users] Dial and start music on hold after timeout

Sorry, I forgot to write that the SIP peer must keep ringing while the announcement is being played. Le 22/08/2016 à 17:42, John Kiniston a écrit : This seems like the obvious answer but maybe I'm misunderstanding the question. exten => s,1,Dial(SIP/alice,20) same =>

Re: [asterisk-users] Dial and start music on hold after timeout

This seems like the obvious answer but maybe I'm misunderstanding the question. exten => s,1,Dial(SIP/alice,20) same => n,Playback(myannouncement) same => n,NoOP(Whatever else you want to do goes here) On Mon, Aug 22, 2016 at 8:36 AM, Jean Aunis wrote: > Hello, > >

[asterisk-users] Dial and start music on hold after timeout

Hello, I am searching a way to dial a SIP peer, and if it does not answer within 20 seconds, play an announcement to the caller. This means that the caller would hear a ring tone for 20 seconds, and only then hear the announcement if the callee did not answer. I know it is possible to do

[asterisk-users] Asterisk Realtime RTUPDATE issue

Hi, I'm currently using Asterisk 11.7.0.The issue currently I'm facing in Asterisk realtime sip_buddies table i.e. if I try to unregister the extension, ipaddr, port, regseconds, fullcontact, useragent and lastms remain still populated with data unless do the sip reload. This issue also obser In

Re: [asterisk-users] http.conf - haproxy

On Mon, Aug 22, 2016 at 11:53 AM, Marek Červenka wrote: > hello, > > is it possible move asterisk http server behind haproxy (haproxy as SSL > endpoint, asterisk http only) > > any examples? > > my current http.conf > > [general] > enabled=yes > bindaddr=0.0.0.0 >

[asterisk-users] http.conf - haproxy

hello, is it possible move asterisk http server behind haproxy (haproxy as SSL endpoint, asterisk http only) any examples? my current http.conf [general] enabled=yes bindaddr=0.0.0.0 tlsenable=yes tlsbindaddr=0.0.0.0:8089 tlscertfile=/etc/pki/tls/certs/some.crt