Archives : March-2014
Home » 2014 (paged 36)
The Asterisk Development Team has announced the releases of:DAHDI-Linux-v2.9.1-rc2DAHDI-Tools-v2.9.1-rc2dahdi-linux-complete-2.9.1-rc2+2.9.1-rc2This release is available for immediate download at:http://downloads.asterisk.org/pub/telephony/dahdi-li..
When I get a SIP INVITE as follows:INVITE sip:s@10.1.0.191:5060 SIP/2.0Max-Forwards: 69From: 0475XXXXXX ;tag=as7df9ab18To: Contact: Call-ID: 344d42bd16975a54141d11f635bdfc71@sip.domain.com CSeq: 102 INVITEDate: Wed, 26 Mar 2014 15:06:01 GMTAllow: INVI..
all,I have a user who is reporting dropped calls at his site.We dont have any other users complaining of this.So far, this is what we know:1.The manager bought all new Polycom phones. (POE)2.They replaced the network switch with a POE version.3.Its ..
Everyone.
I am getting this error WARNING[31977][C-00000009]: chan_sip.c:10657 process_sdp: Cant provide secure audio requested in SDP offer
From the sdp can anyone suggest why secure audio cannot be p..
HiIts possible in Asterisk 1.8 enable verbose only in one context or extension?thanksAtt,*Rafael dos Santo..
I see a lot of attempts by hackers to call 00972595301123? or 011972595115207? or variations but that same 972595 is often present.Can someone break down that dial string with an explanation?The 011 look like an overseas call (from Americas), while ..
using asterisk 1.8.12.2 and realtime architecture with mysql.I get the following message on CLI when changing the value in the strategy/[Mar 26 11:02:24] WARNING[10648]: app_queue.c:2030 queue_set_param: Changing to the linear strategy currently requi..
Hi!I have strange requirement: a incoming call should be duplicated to two outgoing calls (to two voice recorders). On the incoming channel we only receive RTP, on the two outgoing channel we only send RTP.I thought of:incoming call-> originate: m..
On a 11.8.1 system, Im trying to configure a queue in which busy agents do no get incoming calls.I dynamically add agents with lines (shamelessly copied from a running Freepbx/Asterisk 1.8 system) such as:AddQueueMember(myqueue,Local/123@agent/n,10,,hint:123@subs)Basical..
I am trying to make PJSIP work with my Cisco SPA504G phone.I have no problems making it work with the chan_sip driver.When I configure my phone, it indicates the contact was added– Added contact sip:7001@192.168.9.142:5063 to AOR 7001 with expirat..