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
---------------------
作者：far_away_12
来源：CSDN
原文：https://blog.csdn.net/Yin_Zhenwen/article/details/77891617
版权声明：本文为博主原创文章，转载请附上博文链接！
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