PRACK (100rel) Support

Asterisk Asterisk SIP
21

i am able add below modules in modules.conf and restarted Asterisk and got below error while running call

load => res_pjsip.so
load => res_pjsip_pubsub.so
load => res_pjsip_session.so
load => chan_pjsip.so
load => res_pjsip_exten_state.so
load => res_pjsip_log_forwarder.so

root@svtlabaustin:/etc/asterisk# asterisk -r
Asterisk 15.1.4, Copyright © 1999 - 2016, Digium, Inc. and others.
Created by Mark Spencer markster@digium.com
Asterisk comes with ABSOLUTELY NO WARRANTY; type ‘core show warranty’ for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type ‘core show license’ for details.

Running as user ‘asteriskpbx’
Running under group ‘asteriskpbx’
Connected to Asterisk 15.1.4 currently running on svtlabaustin (pid = 18222)
== Setting global variable ‘SIPDOMAIN’ to ‘144.60.212.77’
– Executing [6166310801@LocalSets:1] Dial(“PJSIP/+525552420103-00000008”, “SIP/vISBC/6166310801”) in new stack
[2018-03-26 16:11:25] WARNING[18637][C-00000009]: channel.c:6051 request_channel: No channel type registered for ‘SIP’
[2018-03-26 16:11:25] WARNING[18637][C-00000009]: app_dial.c:2510 dial_exec_full: Unable to create channel of type ‘SIP’ (cause 66 - Channel not implemented)
== Everyone is busy/congested at this time (1:0/0/1)
– Auto fallthrough, channel ‘PJSIP/+525552420103-00000008’ status is ‘CHANUNAVAIL’
– Executing [h@LocalSets:1] Dial(“PJSIP/+525552420103-00000008”, “SIP/vISBC/h”) in new stack
– Caller hung up before dial.
== Spawn extension (LocalSets, h, 1) exited non-zero on ‘PJSIP/+525552420103-00000008’

do i need to change the extension.conf for pjsip ?

22

Yes, You will have to rewrite your dialplan to switch from SIP to PJSIP.

There are many changes, Not just Dial but hints, channel information, working with header data, etc.

23

if you have any dial plan example please share for PJSIP

current dailplan

exten => _!X.,1,DIAL(${LOCAL1}/${EXTEN})

24

my global variable are

[Globals]
LOCAL1 = SIP/vISBC
LOCAL2 = SIP/vISBC-2

25

https://wiki.asterisk.org/wiki/display/AST/Dialing+PJSIP+Channels

26

John

i tried the same but u it failed ,
exten => _!X.,1,DIAL(PJSIP/${EXTEN})

how to assign ${LOCAL1} variable here

can you help

27

You assign variables with the SET command

https://wiki.asterisk.org/wiki/display/AST/Asterisk+15+Application_Set

exten => s,1,Set(LOCAL1=MyValue)

28

Hi

i was able to convert sip.conf to pjsip.conf and my end clients are also registered successfully .

when i trying to outbound call its failed below error

Setting global variable ‘SIPDOMAIN’ to ‘144.60.212.77’
– Executing [6166310801@LocalSets:1] Dial(“PJSIP/+525552420103-00000000”, “PJSIP/166310801@12.255.145.126”) in new stack
[2018-03-27 13:38:55] ERROR[14400]: chan_pjsip.c:2419 request: Unable to create PJSIP channel - endpoint ‘12.255.145.126’ was not found
[2018-03-27 13:38:55] WARNING[14446][C-00000001]: app_dial.c:2510 dial_exec_full: Unable to create channel of type ‘PJSIP’ (cause 3 - No route to destination)
== Everyone is busy/congested at this time (1:0/0/1)
– Auto fallthrough, channel ‘PJSIP/+525552420103-00000000’ status is ‘CHANUNAVAIL’
– Executing [h@LocalSets:1] Dial(“PJSIP/+525552420103-00000000”, “PJSIP/@12.255.145.126”) in new stack
– Caller hung up before dial.
== Spawn extension (LocalSets, h, 1) exited non-zero on ‘PJSIP/+525552420103-00000000’
svtlabaustin*CLI>

pjsip.conf

[vISBC]
type = aor
contact = sip:12.255.145.126

[vISBC]
type = identify
endpoint = vISBC
match = 12.255.145.126

[vISBC]
type = endpoint
context = unauthenticated
dtmf_mode = rfc2833
disallow = all
allow = ulaw
timers_sess_expires = 3800
direct_media = no
inband_progress = yes
aors = vISBC