Currently i am running 15.X release of Asterisk , Does this version support PRACK
There are two SIP channel drivers in Asterisk 15. The chan_sip channel driver does not support PRACK. The chan_pjsip one does.
how to install chan_pjsip version here
currently these modules installed
root@svtlabaustin:/usr/src/asterisk-15.1.4/channels# locate chan_pjsip
Asterisk 15 makes PJSIP available automatically, unless you explicitly disable it using “make menuselect”. There are also instructions on the wiki for how to set it up and other things.
Asterisk Module and Build Option Selection ************************************************** Press 'h' for help. --- Core --- [*] chan_bridge_media XXX chan_dahdi [*] chan_iax2 XXX chan_motif [*] chan_pjsip [*] chan_rtp --- Extended --- XXX chan_alsa XXX chan_console [*] chan_mgcp XXX chan_misdn XXX chan_nbs [*] chan_oss [*] chan_phone [*] chan_sip [*] chan_skinny [*] chan_unistim XXX chan_vpb
chan_pjsip Module already enabled
do we need to disable chan_sip ? enable only chan_pjsip module only
kindly please help
They both can’t use the same port, but chan_sip can be configured to use a different port (such as 5070).
should we need to build with chan_pjsip again and disable chan_sip and build again
No, chan_pjsip is already built. You just have to configure things.
can you please help us to do the configuration
we are new to this . Really Appreciated your support
I have already provided a link to the wiki for configuring PJSIP. It includes examples and other things. For configuring chan_sip you need to look at the sample file and see where the port can be set.
i have below parameters configured in sip.conf can you please let me know which one i need to configure
context=unauthenticated ; default context for incoming calls
allowguest=yes ; enable unauthenticated calls
srvlookup=yes ; enabled DNS SRV record lookup on outbound calls
udpbindaddr=0.0.0.0 ; listen for UDP requests on all interfaces
tcpenable=no ; disable TCP support
progressinband=yes ; If we should generate in-band ringing always
directmedia=no ; Asterisk provides bridging of RTP packets
session-expires=3800 ; Session expiry timer
usereqphone=yes ; If yes, “;user=phone” is added to uri that contains
; a valid phone number
office-phone ; create a template for our devices
type=friend ; the channel driver will match on username first, IP second
context=LocalSets ; this is where calls from the device will enter the dialplan
host=dynamic ; the device will register with asterisk
;nat=force_rport,comedia ; assume device is behind NAT
secret=288Labs ; a secure password for this device – DON’T USE THIS PASSWORD!
dtmfmode=auto ; accept touch-tones from the devices, negotiated automatically
disallow=all ; reset which voice codecs this device will accept or offer
allow=ulaw ; which audio codecs to accept from, and request to, the device
Have you looked at the documentation or tried?
this is what we have in our system i am looking the parameter to enable it
I understand that, but have you tried at all or are you just relying on me to tell you and not even try to figure it out yourself?
i Migrated file from sip.conf to pjsip.conf
Please, report any issue at:
Converting to PJSIP…
svtlabaustinCLI> sip reload
[2018-03-26 14:52:21] NOTICE: chan_sip.c:32170 reload_config: Unable to load config sip.conf
can you please help us to add modules on modules.conf
Why are you reloading chan_sip when you are attempting to use chan_pjsip?
where to load this module chan_pjsip . could you please help me
Asterisk configuration file
; Module Loader configuration file
; Any modules that need to be loaded before the Asterisk core has been
; initialized (just after the logger has been initialized) can be loaded
; using ‘preload’. This will frequently be needed if you wish to map all
; module configuration files into Realtime storage, since the Realtime
; driver will need to be loaded before the modules using those configuration
; files are initialized.
; An example of loading ODBC support would be:
;preload => res_odbc.so
;preload => res_config_odbc.so
; Uncomment the following if you wish to use the Speech Recognition API
;preload => res_speech.so
; If you want Asterisk to fail if a module does not load, then use
; the “require” keyword. Asterisk will exit with a status code of 2
; if a required module does not load.
; require = chan_sip.so
; If you want you can combine with preload
; preload-require = res_odbc.so
; If you want, load the GTK console right away.
noload => pbx_gtkconsole.so
;load => pbx_gtkconsole.so
load => res_musiconhold.so
; Load one of: chan_oss, alsa, or console (portaudio).
; By default, load chan_oss only (automatically).
noload => chan_alsa.so
;noload => chan_oss.so
noload => chan_console.so