asterisk extension calls between res_pjsip and chan_sip

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2017-09-08 11:55

As you know, asterisk has integrated pjsip for sip signalling since asterisk-11. pjsip has advantages of faster and flexiblility than chan_sip.
If you are used to use chan_sip, you' d better to migrate your projects from chan_sip to res_pjsip. But at the beginning, sometimes, you also need to use both of them.
This article tries to illustate how to handle this job.

1. compile and install asterisk-13.5.0 with pjsip enabled, you can refer to the following steps
"How to Install Asterisk 13 and PJSIP on CentOS 6"
http://blogs.digium.com/2015/02/24/install-asterisk-13-pjsip-centos-6/

2. configure chan_sip and pjsip extensions
the following conf file contents is not full file that i displayed here, it is the only must!
chan_sip to use port 5060 as sip signalling as default:

1) /etc/asterisk/sip.conf
[general]
udpbindaddr=0.0.0.0

2) res_pjsip to use port 5070 as sip signalling:
/etc/asterisk/pjsip.conf
[transport-udp]
type=transport
protocol=udp ;udp,tcp,tls,ws,wss
bind=0.0.0.0:5070

3) create two pjsip extension 6001, and 6002 (also in pjsip.conf)
[6001]
type=endpoint
transport=transport-udp
context=from-internal
disallow=all
allow=ulaw
allow=alaw
allow=gsm
auth=auth6001
aors=6001

[auth6001]
type=auth
auth_type=userpass
password=6001
username=6001

[6001]
type=aor
max_contacts=1
;contact=sip:6001@192.168.66:5060

[6002]
type=endpoint
transport=transport-udp
context=from-internal
disallow=all
allow=ulaw
allow=alaw
auth=auth6002
aors=6002

[auth6002]
type=auth
auth_type=userpass
password=6002
username=6002

[6002]
type=aor
max_contacts=2

4) create two chan_sip extension 805, 806
extensions_additional.conf

[ext-local]
include => ext-local-custom
exten => 805,1,Macro(exten-vm,novm,805)
exten => 805,hint,SIP/805
exten => 806,1,Macro(exten-vm,novm,806)
exten => 806,hint,SIP/806

sip_additional.conf

[805]
username=805
type=friend
TermType=0
secret=freepbx321ext
record_out=Adhoc
record_in=Adhoc
qualify=yes
port=5060
pickupgroup=1
nat=yes
mailbox=805@device
host=dynamic
dtmfmode=rfc2833
DispReg=0
context=from-internal
canreinvite=no
calllimit=2
callgroup=1
callerid=device <805>

[806]
username=806
type=friend
TermType=0
secret=freepbx321ext
record_out=Adhoc
record_in=Adhoc
qualify=yes
port=5060
pickupgroup=1
nat=yes
mailbox=806@device
host=dynamic
dtmfmode=rfc2833
DispReg=0
context=from-internal
canreinvite=no
calllimit=2
callgroup=1
callerid=device <806>

5) destination for pjsip extensions
extensions_additional.conf

[ext-pjsip]
exten => _6XXX,1,Dial(PJSIP/${EXTEN})

6) dialplan for 'from-internal' of extensions.conf
[from-internal]
;allow phones to use applications
include => app-userlogonoff
include => app-directory
include => app-dnd
include => app-callforward
include => app-callwaiting
include => app-messagecenter
include => app-calltrace
include => parkedcalls
include => from-internal-custom
;allow phones to dial other extensions
include => ext-fax
include => ext-local
include => ext-group
include => ext-queues
include => ext-zapbarge
include => ext-meetme
include => ext-record
include => ext-test
include => ext-pjsip
;allow phones to access trunks
include => outbound-allroutes
exten => s,1,Macro(hangupcall)
exten => h,1,Macro(hangupcall)

[default]
include => ext-local
include => ext-pjsip
exten => s,1,Playback(vm-goodbye)
exten => s,2,Macro(hangupcall)

sip_additonal.conf should be included in sip.conf
extensions_additional.conf should be included in extensions.conf

3. examine extensions in the dialplan
you can reload asterisk
[root@zhenwen ~]# asterisk -vvvr
zhenwen*CLI> core reload
zhenwen*CLI> module show like res_pjsip.so
Module Description Use Count Status Support Level
res_pjsip.so Basic SIP resource 24 Running core
1 modules loaded
zhenwen*CLI> module show like chan_sip.so
Module Description Use Count Status Support Level
chan_sip.so Session Initiation Protocol (SIP) 0 Running core
1 modules loaded

now register 4 ip phones to asterisk server (which ip addredd is to 192.168.1.66) with exetension 805, 806, 6001, 6002 .
zhenwen*CLI> sip show peers
Name/username Host Dyn Forcerport Comedia ACL Port Status Description
805/805 192.168.1.124 D Yes Yes 2158 OK (100 ms)
806/806 192.168.1.124 D Yes Yes 7458 OK (100 ms)
2 sip peers [Monitored: 2 online, 0 offline Unmonitored: 0 online, 0 offline]

zhenwen*CLI> pjsip show endpoints
Endpoint: <Endpoint/CID.....................................> <State.....> <Channels.>
I/OAuth: <AuthId/UserName...........................................................>
Aor: <Aor............................................> <MaxContact>
Contact: <Aor/ContactUri..........................> <Hash....> <Status> <RTT(ms)..>
Transport: <TransportId........> <Type> <cos> <tos> <BindAddress..................>
Identify: <Identify/Endpoint.........................................................>
Match: <criteria.........................>
Channel: <ChannelId......................................> <State.....> <Time.....>
Exten: <DialedExten...........> CLCID: <ConnectedLineCID.......>
==========================================================================================

Endpoint: 6001 Not in use 0 of inf
InAuth: auth6001/6001
Aor: 6001 1
Contact: 6001/sip:6001@192.168.1.30:10860;rinstance 184588825f Unknown nan
Transport: transport-udp udp 0 0 0.0.0.0:5070

Endpoint: 6002 Not in use 0 of inf
InAuth: auth6002/6002
Aor: 6002 2
Contact: 6002/sip:6002@192.168.1.30:65515;rinstance 2096239e2c Unknown nan
Transport: transport-udp udp 0 0 0.0.0.0:5070

Objects found: 2

4. make calls between extensions.
1) chan_sip extensions: 805 --> 806
dial 806 from 805, then 805 answers the call.

2) res_pjsip extensions: 6001-->6002
dial 6002 from 6001, then 6002 answers the call.

3) see channels in the system
zhenwen*CLI> core show channels
Channel Location State Application(Data)
SIP/805-00000024 s@macro-dial:11 Up Dial(SIP/806,45,Wwtm)
SIP/806-00000025 (None) Up AppDial((Outgoing Line))
PJSIP/6002-00000022 (None) Up AppDial((Outgoing Line))
PJSIP/6001-00000021 6002@from-internal:1 Up Dial(PJSIP/6002)
4 active channels
2 active calls
52 calls processed
zhenwen*CLI>
the two calls are setup.

4) call between res_pjsip extension and chan_sip extension
hangup the above calls.
dial 805 from 6001. then 6001 answers the call;
dial 6002 from 806, then 806 answers the call.
now we see the channels:

zhenwen*CLI> core show channels
Channel Location State Application(Data)
SIP/805-00000028 6001@from-internal:1 Up Dial(PJSIP/6001)
SIP/806-00000029 (None) Up AppDial((Outgoing Line))
PJSIP/6002-00000026 s@macro-dial:11 Up Dial(SIP/806,45,Wwtm)
PJSIP/6001-00000025 (None) Up AppDial((Outgoing Line))
4 active channels
2 active calls
56 calls processed
zhenwen*CLI>

5. uses pjsip extension as queue member

1) configure queue 200, login 6001 to the queue.

zhenwen*CLI> queue add member PJSIP/6001 to 200
Added interface 'PJSIP/6002' to queue '200'

zhenwen*CLI> queue show
default has 0 calls (max unlimited) in 'ringall' strategy (0s holdtime, 0s talktime), W:0, C:0, A:0, SL:0.0% within 0s
No Members
No Callers

200 has 0 calls (max 30) in 'leastrecent' strategy (4s holdtime, 42s talktime), W:0, C:1, A:0, SL:0.0% within 0s
Members:
PJSIP/6001 (ringinuse enabled) (dynamic) (In use) has taken 1 calls (last was 150806 secs ago)
No Callers

2) make call to 200
dial 200 from 805, then 6001 rings, answer 6001.

zhenwen*CLI> queue show
default has 0 calls (max unlimited) in 'ringall' strategy (0s holdtime, 0s talktime), W:0, C:0, A:0, SL:0.0% within 0s
No Members
No Callers

200 has 0 calls (max 30) in 'leastrecent' strategy (3s holdtime, 42s talktime), W:0, C:1, A:0, SL:0.0% within 0s
Members:
PJSIP/6001 (ringinuse enabled) (dynamic) (in call) (In use) has taken no calls yet
No Callers

zhenwen*CLI>

zhenwen*CLI> core show channels
Channel Location State Application(Data)
SIP/805-0000002a 200@q200:4 Up Queue(200,t,,,60)
PJSIP/6001-00000027 200@from-internal:1 Up AppQueue((Outgoing Line))
2 active channels
1 active call
57 calls processed
zhenwen*CLI>

6. uses asterisk cli to make outbound call
use cli command 'channel originate' to dial extensions.

zhenwen*CLI> channel originate PJSIP/6001 extension 805@ext-local
zhenwen*CLI> channel originate SIP/806 extension 6002@ext-pjsip
== Using SIP RTP CoS mark 5
-- Called 806
-- SIP/806-0000002c is ringing
-- SIP/806-0000002c is ringing
-- SIP/806-0000002c answered
-- Executing [6002@ext-pjsip:1] Dial("SIP/806-0000002c", "PJSIP/6002") in new stack
-- Called PJSIP/6002
-- PJSIP/6002-00000029 is ringing
-- PJSIP/6002-00000029 answered SIP/806-0000002c
-- Channel PJSIP/6002-00000029 joined 'simple_bridge' basic-bridge <035b50ce-a644-443a-b1ca-07161c3477a3>
-- Channel SIP/806-0000002c joined 'simple_bridge' basic-bridge <035b50ce-a644-443a-b1ca-07161c3477a3>
zhenwen*CLI>
zhenwen*CLI> core show channels
Channel Location State Application(Data)
SIP/805-0000002b (None) Up AppDial((Outgoing Line))
SIP/806-0000002c 6002@ext-pjsip:1 Up Dial(PJSIP/6002)
PJSIP/6002-00000029 (None) Up AppDial((Outgoing Line))
PJSIP/6001-00000028 s@macro-dial:11 Up Dial(SIP/805,45,Wwtm)
4 active channels
2 active calls
59 calls processed

7. uses asterisk manager to make calls
login to asterisk manager,
then send the following command:
Action: Originate
Channel: SIP/805
Context: default
Exten: 6001
Priority: 1
Callerid: 805
Timeout: 30000

asterisk cli displays:
== Using SIP RTP CoS mark 5
-- Called 805
-- SIP/805-0000002d is ringing
-- SIP/805-0000002d is ringing
-- SIP/805-0000002d answered
-- Executing [6001@default:1] Dial("SIP/805-0000002d", "PJSIP/6001") in new stack
-- Called PJSIP/6001
-- PJSIP/6001-0000002a is ringing
-- PJSIP/6001-0000002a answered SIP/805-0000002d
-- Channel PJSIP/6001-0000002a joined 'simple_bridge' basic-bridge <381573a4-c31e-40e4-b657-309a06cec842>
-- Channel SIP/805-0000002d joined 'simple_bridge' basic-bridge <381573a4-c31e-40e4-b657-309a06cec842>

zhenwen*CLI> core show channels
Channel Location State Application(Data)
SIP/805-0000002d 6001@default:1 Up Dial(PJSIP/6001)
PJSIP/6001-0000002a (None) Up AppDial((Outgoing Line))
2 active channels
1 active call
60 calls processed
zhenwen*CLI>

8. dial into meetme room
we configure a meetme room in extensions.conf
[ext-meetme]
; general confrooms with no pin requered , 2017-08-03 Yin Zhenwen
exten => _2XXX!,1,Answer
exten => _2XXX!,n,MeetMe(${EXTEN},qdsMF)

now, we use 4 phones to dial 2806 into the same room.
zhenwen*CLI> meetme list 2806
User #: 01 6001 6001 Channel: PJSIP/6001-0000002b (unmonitored) 00:00:44
User #: 02 6002 6002 Channel: PJSIP/6002-0000002c (unmonitored) 00:00:30
User #: 03 805 device Channel: SIP/805-0000002e (unmonitored) 00:00:19
User #: 04 806 device Channel: SIP/806-0000002f (unmonitored) 00:00:09
4 users in that conference.
zhenwen*CLI>

ok, the room has 4 talkers.

---------
summerize: we have configured and tested calls between res_pjsip and chan_sip extensions at the following scenario:
1) dial from extensions
2) dial to queue
3) dial from cli command
4) dial from manager command
5) dial into meetme room

if you need any help concern about the above test, please contact me: QQ: 1124844608.

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