Archives : April-2016
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Id like to record the barged call… but whichever leg of the call I try to barge, my speaking is never recorded using MixMonitor. Any idea about the reas..
Im using the following Dial command syntax:Dial*(SIP/peer/exten!sip:xyz@xyz.com *), the SIP URIafter the ! mark should be set as To-URI in outgoing INVITEfrom Asterisk. It works, but problem is that To-URI formatting is a bit messed up, It looks as follows:*sip:sip:xyz@xyz…
Im giving PJSIP a try on an Asterisk 13 box. More specifically Im studying in a lab, how to configure T.38.My setup is:SendFAX –>– asterisk 11asterisk 13 –>– ReceiveFAXI can observe fax are successfully sent end received but Im failling to see ..
!On an Asterisk-Server I have some users. Just two of them have a Mailbox. I want to write a little Web interface to manage many things and Id like to have a menu point for the voicemail, but just if the user has a Mailbox.I found the AMI-Command MailboxStat..
I recently had to reinstall Asterisk and FreePBX. asteirsk 11.20 and FreePBX 12. This is running on CentOS 6.7 32 bit. When I use amportal start It comes up with the errors below Error in argument 1, char 2: option not found r/usr/local/sbin/amport..
Hi.Im here to invite you all to test another PoC that I developed and that uses Asterisk and XMPP, called XyBot.XyBot is a XMPP bot written in python and its main goal is to enable users to interact with asterisk directly from their XMPP client.Xy..
to everyoneI have a Automatic Call Distribution for I receive calls, and it is normalBut how can I make for outbound calls using a E1 links with 30 channels?Is there a specific code for t..
The Asterisk Development Team has announced the release of Certified Asterisk 13.1-cert6. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/certified-asteriskThe release of Certified Asterisk 13.1-cert6 resol..
The Asterisk Development Team has announced the release of Asterisk 13.8.2. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asteriskThe release of Asterisk 13.8.2 resolves several issues reported by ..
all,Checking on the asterisk source code Ive seen that SIP will always use the IP address in the c= field of SDP to send media. Is that correct?Is there a case where asterisk would send media to the received source IPaddress instead of the one he ..