I am using Asterisk 1.8.10.1 and a MySQL database connected via ODBC to store CDRs. When my MySQL database isn't available when Asterisk starts or has an outage while Asterisk is running, I would expect Asterisk to retry to connect to the database, but this doesn't happen!
Skype has terminated its partnership with Digium, effectively killing Skype for Asterisk, which integrated Skype's VOIP service with the open source PBX/telephony platform. While some analysts see Microsoft's impending acquisition of Skype as the source of the schism, others argue that Skype has never been an open source supporter, and had already been backing away from the Asterisk product....
I have enabled TLS in asterisk
[general]
tlsenable=yes
tlscertfile=/etc/asterisk/ssl/asterisk.crt
tlsdontverifyserver=no
Everything works, tested on CSipSimple. But there is one problem: not much clients support TLS with client certs (Microsip and blink do not support)
How can I make asterisk do not require client certificates for TLS?
Thanks
If you just install Asterisk gui 2.o from:
svn checkout http://svn.digium.com/svn/asterisk-gui/branches/2.0
And got a 404 URL not found <asterisk server> typing http://ip_machine:8088/static/config/index.html
and already did the “make checkconfig” cmd getting an “everything seems good” output.
and already Checked in Asterisk CLI:
I am trying to figure, if Asterisk can support SIP MESSAGE message outside of established call.
I have to use SIP MESSAGE in order to simulate a near-real-time chat functionality (it's a project specification). Until now my research shows, that this is not possible.
Most people use XMPP or other means to achieve IM functionality, or install OpenSer in front of Asterisk.
It's wrong way I think.During asterisk installation (after "make install" command) run "make config". It creat start/stop script for asterisk and is't listed in ... [by svyat]
Hi all.
I know this is not an asterisk forum, can someone with his/her experience guide me to a wonderful forum like LQ for beginners ?
I am still just explaining my problem. If anyone feels like kindly guide me.
I have installed centos 5.5 on vmware workstation. I successfully installed asterisk 1.4.
I have received sms on my asterisk server via sip on my asterisk version 1.4.11 but not able to route it from agi or send it to some url bellow lines i can see on console.
[Feb 24 23:50:29] WARNING[23972]: chan_sip.c:9496 receive_message: Received message to <sip:13214375437576@16.151.18.14> from <sip:131231233214@164.36.74.30>;tag=sansay1824778355rdb15870, dropped it...
Content-Type:
Hey, i just had to switch from 10 to 11 until 12 comes out, and after setting up asterisk, i noticed VERY VERY BAD sound quality, mostly comprised of severe jittering.