Archives : October-2016
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Hi!I need to make a dialplan by DID.where it gets the asterisk values did? from sip headers or ….
HelloIm a bit confused on how to group agents (give agents a group number) when using realtime queues.I read on the wiki :* If you include groups in your queue definition the calls get routedin the order of the group regardless of the specified strate..
can you recommend open source helpdesk solution with working Asterisk integra..
My sip provider gave me 2 numbers for the incoming call via pstn.nro1 = 12341234nro2 = 45674567I have a dialplan for each. if i put this on my dialplan:exten => s,1,Dial(SIP/1001)exten => Hangup()Works!But if i put one of them:exten => 12341234,1,Dial(SIP/1001)ex..
I am making SIP calls using SIP.js and configuring Asterisk 11.x for websockets calls under CentOS 7. On 11.23.1 and earlier, I had to patch the code to disable auto negociation due to ASTERISK-25659. Now that the bug is supposedly fixed in commit 8653da4fa228e1e289e09e5d024e11d24da87d..
HelloI keep getting the following error when trying to connect to the Asterisk server using AMI :$socket = fsockopen(tls://11.22.33.44,5039, $errno, $errstr, 5);Erorr on CLI :[Oct 26 14:38:19] ERROR[2992]: tcptls.c:609 handle_tcptls_connection: Prob..
Hey All,This is a friendly notice that as of today Asterisk 11 has entered security fix only mode.From this point onward additional releases of Asterisk 11 will no longer be made unless there is a security fix being applied to the branch.Users of Aster..
The Asterisk Development Team has announced the release of Asterisk 11.24.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asteriskThe release of Asterisk 11.24.0 resolves several issues reported by ..
On September 8, the Asterisk development team released the AST-2016-007security advisory. The security advisory involved an RTP resource exhaustion that could be targeted due to a flaw in the allowoverlapoption of chan_sip. Due to new information presen..
I am trying to configure new opus codec in asterisk 14, but unable to find any examples of codecs.conf settings for this codec.All I am trying to do – setup peer with using opus in narrow band mode(8kHz sampling rate). Does anybody know how to config..