# PRACK (100rel) Support

**URL:** <https://community.asterisk.org/t/prack-100rel-support/74063>\
**Category:** Asterisk SIP\
**Created:** [March 26, 2018, 6:52pm UTC](https://community.asterisk.org/t/prack-100rel-support/74063 "2018-03-26T18:52:53Z")\
**Posts on this page:** 20\
**Page:** 1

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**Author:** ![veluvarti](https://avatars.discourse-cdn.com/v4/letter/v/ecc23a/32.png) [@veluvarti](https://community.asterisk.org/u/veluvarti)\
**Post date:** [March 26, 2018, 6:52pm UTC](https://community.asterisk.org/t/prack-100rel-support/74063/1 "2018-03-26T18:52:53Z")

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Currently i am running 15.X release of Asterisk , Does this version support PRACK

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<div class="post-metadata">

**Author:** ![jcolp](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/jcolp/32/8848_2.png) [@jcolp](https://community.asterisk.org/u/jcolp)\
**Post date:** [March 26, 2018, 6:56pm UTC](https://community.asterisk.org/t/prack-100rel-support/74063/2 "2018-03-26T18:56:46Z")

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There are two SIP channel drivers in Asterisk 15. The chan\_sip channel driver does not support PRACK. The chan\_pjsip one does.

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<div class="post-metadata">

**Author:** ![veluvarti](https://avatars.discourse-cdn.com/v4/letter/v/ecc23a/32.png) [@veluvarti](https://community.asterisk.org/u/veluvarti)\
**Post date:** [March 26, 2018, 6:58pm UTC](https://community.asterisk.org/t/prack-100rel-support/74063/3 "2018-03-26T18:58:38Z")

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how to install chan\_pjsip version here

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<div class="post-metadata">

**Author:** ![veluvarti](https://avatars.discourse-cdn.com/v4/letter/v/ecc23a/32.png) [@veluvarti](https://community.asterisk.org/u/veluvarti)\
**Post date:** [March 26, 2018, 6:59pm UTC](https://community.asterisk.org/t/prack-100rel-support/74063/4 "2018-03-26T18:59:14Z")

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currently these modules installed

root@svtlabaustin:/usr/src/asterisk-15.1.4/channels# locate chan\_pjsip  
/usr/lib/asterisk/modules/chan\_pjsip.so  
/usr/src/asterisk-15.1.4/channels/.chan\_pjsip.makeopts  
/usr/src/asterisk-15.1.4/channels/.chan\_pjsip.moduleinfo  
/usr/src/asterisk-15.1.4/channels/.chan\_pjsip.o.d  
/usr/src/asterisk-15.1.4/channels/chan\_pjsip.c  
/usr/src/asterisk-15.1.4/channels/chan\_pjsip.exports  
/usr/src/asterisk-15.1.4/channels/chan\_pjsip.o  
/usr/src/asterisk-15.1.4/channels/chan\_pjsip.so  
/usr/src/asterisk-15.1.4/channels/pjsip/include/chan\_pjsip.h  
root@svtlabaustin:/usr/src/asterisk-15.1.4/channels#

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<div class="post-metadata">

**Author:** ![jcolp](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/jcolp/32/8848_2.png) [@jcolp](https://community.asterisk.org/u/jcolp)\
**Post date:** [March 26, 2018, 6:59pm UTC](https://community.asterisk.org/t/prack-100rel-support/74063/5 "2018-03-26T18:59:40Z")

</div>

Asterisk 15 makes PJSIP available automatically, unless you explicitly disable it using “make menuselect”. There are also instructions on the wiki[1] for how to set it up and other things.

[1] [https://wiki.asterisk.org/wiki/display/AST/Configuring+res\_pjsip](https://wiki.asterisk.org/wiki/display/AST/Configuring+res_pjsip)

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<div class="post-metadata">

**Author:** ![veluvarti](https://avatars.discourse-cdn.com/v4/letter/v/ecc23a/32.png) [@veluvarti](https://community.asterisk.org/u/veluvarti)\
**Post date:** [March 26, 2018, 7:10pm UTC](https://community.asterisk.org/t/prack-100rel-support/74063/6 "2018-03-26T19:10:49Z")

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```
  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
```

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<div class="post-metadata">

**Author:** ![veluvarti](https://avatars.discourse-cdn.com/v4/letter/v/ecc23a/32.png) [@veluvarti](https://community.asterisk.org/u/veluvarti)\
**Post date:** [March 26, 2018, 7:11pm UTC](https://community.asterisk.org/t/prack-100rel-support/74063/7 "2018-03-26T19:11:13Z")

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Hi

chan\_pjsip Module already enabled

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<div class="post-metadata">

**Author:** ![veluvarti](https://avatars.discourse-cdn.com/v4/letter/v/ecc23a/32.png) [@veluvarti](https://community.asterisk.org/u/veluvarti)\
**Post date:** [March 26, 2018, 7:16pm UTC](https://community.asterisk.org/t/prack-100rel-support/74063/8 "2018-03-26T19:16:35Z")

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Hi  
do we need to disable chan\_sip ? enable only chan\_pjsip module only

kindly please help

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<div class="post-metadata">

**Author:** ![jcolp](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/jcolp/32/8848_2.png) [@jcolp](https://community.asterisk.org/u/jcolp)\
**Post date:** [March 26, 2018, 7:17pm UTC](https://community.asterisk.org/t/prack-100rel-support/74063/9 "2018-03-26T19:17:13Z")

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They both can’t use the same port, but chan\_sip can be configured to use a different port (such as 5070).

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<div class="post-metadata">

**Author:** ![veluvarti](https://avatars.discourse-cdn.com/v4/letter/v/ecc23a/32.png) [@veluvarti](https://community.asterisk.org/u/veluvarti)\
**Post date:** [March 26, 2018, 7:20pm UTC](https://community.asterisk.org/t/prack-100rel-support/74063/10 "2018-03-26T19:20:10Z")

</div>

should we need to build with chan\_pjsip again and disable chan\_sip and build again

please help

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<div class="post-metadata">

**Author:** ![jcolp](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/jcolp/32/8848_2.png) [@jcolp](https://community.asterisk.org/u/jcolp)\
**Post date:** [March 26, 2018, 7:21pm UTC](https://community.asterisk.org/t/prack-100rel-support/74063/11 "2018-03-26T19:21:11Z")

</div>

No, chan\_pjsip is already built. You just have to configure things.

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<div class="post-metadata">

**Author:** ![veluvarti](https://avatars.discourse-cdn.com/v4/letter/v/ecc23a/32.png) [@veluvarti](https://community.asterisk.org/u/veluvarti)\
**Post date:** [March 26, 2018, 7:22pm UTC](https://community.asterisk.org/t/prack-100rel-support/74063/12 "2018-03-26T19:22:14Z")

</div>

can you please help us to do the configuration

we are new to this . Really Appreciated your support

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<div class="post-metadata">

**Author:** ![jcolp](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/jcolp/32/8848_2.png) [@jcolp](https://community.asterisk.org/u/jcolp)\
**Post date:** [March 26, 2018, 7:23pm UTC](https://community.asterisk.org/t/prack-100rel-support/74063/13 "2018-03-26T19:23:27Z")

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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.

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<div class="post-metadata">

**Author:** ![veluvarti](https://avatars.discourse-cdn.com/v4/letter/v/ecc23a/32.png) [@veluvarti](https://community.asterisk.org/u/veluvarti)\
**Post date:** [March 26, 2018, 7:27pm UTC](https://community.asterisk.org/t/prack-100rel-support/74063/14 "2018-03-26T19:27:15Z")

</div>

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

transport=udp  
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  
;transport=tcp  
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

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<div class="post-metadata">

**Author:** ![jcolp](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/jcolp/32/8848_2.png) [@jcolp](https://community.asterisk.org/u/jcolp)\
**Post date:** [March 26, 2018, 7:29pm UTC](https://community.asterisk.org/t/prack-100rel-support/74063/15 "2018-03-26T19:29:16Z")

</div>

Have you looked at the documentation or tried?

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<div class="post-metadata">

**Author:** ![veluvarti](https://avatars.discourse-cdn.com/v4/letter/v/ecc23a/32.png) [@veluvarti](https://community.asterisk.org/u/veluvarti)\
**Post date:** [March 26, 2018, 7:30pm UTC](https://community.asterisk.org/t/prack-100rel-support/74063/16 "2018-03-26T19:30:34Z")

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this is what we have in our system i am looking the parameter to enable it

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<div class="post-metadata">

**Author:** ![jcolp](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/jcolp/32/8848_2.png) [@jcolp](https://community.asterisk.org/u/jcolp)\
**Post date:** [March 26, 2018, 7:30pm UTC](https://community.asterisk.org/t/prack-100rel-support/74063/17 "2018-03-26T19:30:55Z")

</div>

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?

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<div class="post-metadata">

**Author:** ![veluvarti](https://avatars.discourse-cdn.com/v4/letter/v/ecc23a/32.png) [@veluvarti](https://community.asterisk.org/u/veluvarti)\
**Post date:** [March 26, 2018, 8:15pm UTC](https://community.asterisk.org/t/prack-100rel-support/74063/18 "2018-03-26T20:15:30Z")

</div>

Hi

i Migrated file from sip.conf to pjsip.conf

root@svtlabaustin:/etc/asterisk# /usr/src/asterisk-15.1.4/contrib/scripts/sip\_to\_pjsip/sip\_to\_pjsip.py  
Please, report any issue at:  
[https://issues.asterisk.org/](https://issues.asterisk.org/)  
Reading sip.conf  
Converting to PJSIP…  
Writing pjsip.conf

svtlabaustin_CLI\> sip reload  
Reloading SIP  
[2018-03-26 14:52:21] NOTICE[11939]: chan\_sip.c:32170 reload\_config: Unable to load config sip.conf  
svtlabaustin_CLI\> exit

can you please help us to add modules on modules.conf

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<div class="post-metadata">

**Author:** ![johnkiniston](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/johnkiniston/32/28_2.png) [@johnkiniston](https://community.asterisk.org/u/johnkiniston)\
**Post date:** [March 26, 2018, 8:21pm UTC](https://community.asterisk.org/t/prack-100rel-support/74063/19 "2018-03-26T20:21:48Z")

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> [@veluvarti](#):
>
> svtlabaustinCLI\> sip reload
> 
> Reloading SIP
> 
> [2018-03-26 14:52:21] NOTICE[11939]: chan\_sip.c:32170 reload\_config: Unable to load config sip.conf
> 
> svtlabaustinCLI\> exit

Why are you reloading chan\_sip when you are attempting to use chan\_pjsip?

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<div class="post-metadata">

**Author:** ![veluvarti](https://avatars.discourse-cdn.com/v4/letter/v/ecc23a/32.png) [@veluvarti](https://community.asterisk.org/u/veluvarti)\
**Post date:** [March 26, 2018, 8:28pm UTC](https://community.asterisk.org/t/prack-100rel-support/74063/20 "2018-03-26T20:28:10Z")

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hi John

where to load this module chan\_pjsip . could you please help me

**module.conf** :

Asterisk configuration file  
;  
; Module Loader configuration file  
;

[modules]  
autoload=yes  
;  
; 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  
;

[Next page](https://community.asterisk.org/t/prack-100rel-support/74063.md?page=2)
