I want to put a call me now button on the web site that will place the request into an asterisk call Queue and then when an agent picks up the call in the queue, place the outbound call to the customer.The following AMI command works, but it calls ..
everybody, i have a problem with Asterisk 1.8 and Call Hold My problem is that Asterisk dont send re-invite when i pick up the call from hold. I already insert canreinvite=no in all my sip channels, set dtmfmode=info in sip.conf and my Dial() comm..
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everyone,I am your friendly neighborhood developer here with a question that may impact some of you.Right now there is a small discussion occurring on the Asterisk development mailing list about the expected behavior of music on hold and AGIs str..
!ConfBridge DTMF_passthrough=no doesnt seem to have any effect. DTMF gets transmitted throughout the conference. Ive tried Asterisk 10.7.1 from the official RPMs and 10.8.0 compiled from source.Ive confirmed that its disabled via the CLI confbridge s..
According to http://www.voip-info.org/wiki/view/Asterisk+RealTime+Sip:Quote:If you place ipaddr before host (in the case of dynamic), you will never load the public IP address of your sip device, as it will be overwritten when host is encountered. UnQuote.F..
are there any known reasons why Asterisk would disconnect random calls ?My server uses 1,5 GB out of 8 GB RAMMy server uses up to 35% CPU at peak There are about 40 concurrent calls. I have 300 RTP-ports available.I just see the call ending, as if ..
Is there a way to move 100 .call files in to/var/spool/asterisk/outgoing/ at once and have Asterisk call at maximum 10 at a time?The maxcalls setting in asterisk.conf will limit the number of calls but not gracefully. The calls over 10 just fail. I n..
this might seem a stupid question but I really dont see the solution to the problem.Using Asterisk 220.127.116.11In extconfig.conf I have :voicemail => mysql,AsteriskHosted,voicemail_users sipusers => mysql,AsteriskHosted,sip_buddies sippeers => mysql,AsteriskHosted,sip_budd..
Has anyone had experience using a SIP trunk provided by Paetec over MPLS?With or without FreePBX Regards,Ja..