freepbx : failed to open stream: Permission denied

file(/etc/asterisk/asterisk.conf): failed to open stream: Permission denied

Make sure Selinux  is disbaled


[root@vultr ~]# sestatus
SELinux status:                 disabled

asterisk record

[surf_report]
exten=>_+x.,1,Answer()
same=>n,Set(path=/var/www/html/surf/)
same =>n(2),read(edit,${path}${EXTEN:1},1,,)
same=>n,GotoIf($["${edit}"="5"]?auth)
same=>n,hangup()
same=>n(auth),authenticate(1973)
same=>n(rec),playback(pls-enter-num-message-after-tone)
same=>n,Record(${path}${EXTEN:1}.wav,,,k)
same=>n,playback(${path}${EXTEN:1})
same => n(confirm),read(confirm,press-star-cancel,1,,)
same=>n,GotoIf($["${confirm}"="*"]?rec)
same=>n,Playback(auth-thankyou)
same=>n,hangup()


recording with confirmation

asterisk pjsip easy installation

sterisk 13.8.0 will come with a new option for enabling PJSIP functionality. This functionality is called bundling and comes courtesy of a community member, George Joseph, who you can also thank for such PJSIP additions as wizards for configuration and the PJSIP_HEADER dialplan function. Before we talk about bundling let’s take a look at the history of PJSIP usage in Asterisk and how we got to where we are today.

PJSIP itself is part of a set of libraries and tools which forms PJPROJECT. Another part is referred to as PJNATH and has been used since Asterisk 11 to provide ICE, STUN, and TURN support in the res_rtp_asterisk module. When Asterisk 11 was released PJPROJECT was embedded in Asterisk itself to make it easier for users to have the new functionality available. When the decision was made to work on a new PJSIP channel driver one of the desires expressed by the community was to remove PJPROJECT from Asterisk and have it be distributed using other mechanisms, such as by the distributions as packages or from a manual install process. Before Asterisk 12 was released this was completed and contributed upstream to Teluu who created PJPROJECT. This has meant that to enable PJSIP support in Asterisk you have needed to install PJPROJECT yourself using some method. As of Asterisk 13.8.0 another simpler option will be available instead: bundling.

Bundling allows a self-contained PJSIP to exist within Asterisk and be used by all functionality within it. It is enabled by passing an option to the configure script:

1	./configure --with-pjproject-bundled ./configure --libdir=/usr/lib64 --with-pjproject-bundled

When you run ./configure, you can set the PJPROJECT_URL environment variable to point to an alternate site.
Example:

# ./configure --with-pjproject-bundled PJPROJECT_URL=http://pjsip.org/release/2.6

When the option is enabled the build process will download a version of PJPROJECT as specified in the bundling configuration, patch it with any changes that may not have yet been published in a PJPROJECT release, build it using the best options suitable for Asterisk, make it available to all the Asterisk PJSIP modules, and make the Asterisk PJSIP modules available for building. This reduces the barrier to entry for using PJSIP to a minimal amount allowing more people to use the new SIP functionality. There is no additional work required except for enabling the option.

While the lower barrier to entry may seem like the only immediate benefit there’s also some internal things that are improved by using bundling. By controlling the version of PJSIP that Asterisk is used against we can ensure that Asterisk will build and work properly against it. We can also ensure that both Asterisk and PJSIP have been built using the same configuration and that the configuration matches the usage required by Asterisk. This can lead to improved performance and reduce crashes.

There’s also no harm in using this new bundled support if you already have PJSIP installed. The PJSIP downloaded and built for Asterisk will only be bundled in Asterisk and not exposed to the rest of your system.

fop2 installation easy on freepbx

wget -O - http://download.fop2.com/install_fop2.sh | bash

Asterisk Len & Execif

1. exten=>001,1,Answer()
   same=>n,Noop(${CALLERID(num)})
   same=>n,Set(ARG1=${CALLERID(num)})
   same=>n,Noop( len of ${LEN(${ARG1})})
   same=>n,ExecIf($[${LEN(${ARG1})}= 11]?Set(CALLERID(num)=${ARG1:0:7}${RAND(1000,9999)}))
   same=>n,ExecIf($[${LEN(${ARG1})}> 11]?Set(CALLERID(num)=${ARG1:0:7}))

Asterisk func_odbc

http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/getting_funky.html

Difference Between 180 Ringing and 183 Session Progress

Difference Between 180 Ringing and 183 Session Progress

In general, ringing is controlled via two Informational Responses in SIP: the 180 Ringing and the 183 Session Progress.

The main difference between them, is the 180 Ringing message instructs the UA to create the dial-tone locally, whereas the 183 Session Progress contains an SDP, which allows for regional ring-back and carrier announcements as well.

Typically 183 contains SDP and is used to play media before the call is connected

-----------------------------------------------------

a) 180 Ringing

   The UA receiving the INVITE is trying to alert the user. This

   response MAY be used to initiate local ringback.

b) 183 Session Progress

   The 183 (Session Progress) response is used to convey information

   about the progress of the call that is not otherwise classified.

https://supportforums.cisco.com/t5/ip-telephony/difference-between-180-ringing-and-183-session-progress/td-p/2413632