X-lite tidak keluar suara

Asterisk
1

Saat telpon sesama client masuk tetapi tidak keluar suara, masalahnya kira" dimananya pak?

2

Bisanya issue nat.

Bisa ceritakan kondisi networknya pak ?

3

Begini pak, saat kami tes calling menggunakan jaringan local x-lite nya lancar, tapi setelah kami menggunakan IP public x-lite tidak keluar suara, tetapi terhubung.

4

Sepertinya kasusnya mirip2 ini :

Asterisk ID , Issue : Suara tidak terdengar dari PSTN ke SIP

Coba baca2 postingan disitu dulu pak. Lalu bapak bisa tarik kesimpulan dan bisa ceritakan disini kondisi bapak dengan jelas seperti konfigurasi network dan konfigurasi SIP nya.

5

saya menggunakan jaringan LAN namun tetap tidak keluar suara di X-lite,bagaimana solusinya pak ?

6

topologinya kurang lebih seperti ini pak.

IP public
|
Router Mikrotik
|
IP PBX Asterisk
|
Client Softphone

7

revisi pak mengenai topologinya

routernya langsung dengan internet tidak dibelakang nat/firewall,

8

IP Address Asterisk bapak tidak langsung di set IP Public , IP Public ada di router Mikrotik?

Jadi bapak perlu melakukan NAT dari Mikrotik, jadi yang mengarah ke IP Public untuk port-port VoIP itu diteruskan ke IP PBX Asterisk.

Caranya kalo di Mikrotik seperti ini :

ex :
IP Public : 219.0.0.2
IP Asterisk : 192.168.2.2

Pada Mikrotik tambahkan aturan firewall :

ip firewall nat add chain=dstnat dst-address=219.0.0.2 action=dst-nat to-addresses=192.168.2.2
ip firewall nat add chain=srcnat src-address=192.168.2.2 action=src-nat to-addresses=219.0.0.2

Contoh kasus :

9

Asterisk Sy langsung set IP Public

10

coba paste disini konfigurasi sip dan log debug SIP nya pada saat call tidak ada suara tersebut

11

[general]

externip=111.XXX.XXX.XX
context=default
bindport=5060
bindaddr=0.0.0.0
tcpbindaddr=0.0.0.0
udpbindaddr=0.0.0.0
nat=yes
srvlookup=yes

disallow=all
allow=alaw
allow=gsm
allow=ulaw
allow=ilbc
allow=g722

[111]

nat=yes
qualify=yes
dtmfmode=rfc2833
callerid=usER1
type=friend
context=local
host=dynamic
secret=111
username=user1
allow=all

[222]

nat=yes
qualify=yes
dtmfmode=rfc2833
callerid=user2
type=friend
context=local
host=dynamic
secret=222
username=user2
allow=all

12

[Jan 4 16:29:43] VERBOSE[12107] config.c: == Parsing ‘/etc/asterisk/logger.conf’: [Jan 4 16:29:43] VERBOSE[12107] config.c: == Found
[Jan 4 16:29:43] VERBOSE[12107] logger.c: Asterisk Queue Logger restarted
[Jan 4 16:30:00] VERBOSE[12107] asterisk.c: – Remote UNIX connection disconnected
[Jan 4 16:30:19] VERBOSE[12084] netsock2.c: == Using SIP RTP CoS mark 5
[Jan 4 16:30:19] VERBOSE[12108] pbx.c: – Executing [111@local:1] Dial(“SIP/222-00000004”, “SIP/111”) in new stack
[Jan 4 16:30:19] VERBOSE[12108] netsock2.c: == Using SIP RTP CoS mark 5
[Jan 4 16:30:19] VERBOSE[12108] app_dial.c: – Called SIP/111
[Jan 4 16:30:19] VERBOSE[12108] app_dial.c: – SIP/111-00000005 is ringing
[Jan 4 16:30:33] VERBOSE[12108] app_dial.c: – SIP/111-00000005 answered SIP/222-00000004
[Jan 4 16:30:33] VERBOSE[12108] rtp_engine.c: – Remotely bridging SIP/222-00000004 and SIP/111-00000005
[Jan 4 16:30:34] NOTICE[12108] res_rtp_asterisk.c: Unknown RTP codec 126 received from ‘114.4.143.98:52246’
[Jan 4 16:31:50] VERBOSE[12108] pbx.c: == Spawn extension (local, 111, 1) exited non-zero on ‘SIP/222-00000004’
[Jan 4 16:33:03] VERBOSE[12056] asterisk.c: Executing last minute cleanups
[Jan 4 16:33:04] WARNING[12116] ccss.c: Could not find valid ccss.conf file. Using cc_max_requests default
[Jan 4 16:33:04] NOTICE[12116] loader.c: 32 modules will be loaded.
[Jan 4 16:33:04] WARNING[12116] loader.c: Error loading module ‘res_musiconhold.so’: File not found
[Jan 4 16:33:04] WARNING[12116] loader.c: Error loading module ‘res_musiconhold.so’: File not found
[Jan 4 16:33:04] WARNING[12116] loader.c: Module ‘res_musiconhold.so’ could not be loaded
[Jan 4 16:33:04] WARNING[12116] loader.c: Error loading module ‘res_smdi’: File not found
[Jan 4 16:33:04] VERBOSE[12116] chan_sip.c: SIP channel loading…
[Jan 4 16:33:04] WARNING[12116] chan_sip.c: No valid transports available, falling back to ‘udp’
[Jan 4 16:33:04] ERROR[12116] netsock2.c: getaddrinfo(“metarouter”, “(null)”, …): Name or service not known
[Jan 4 16:33:04] WARNING[12116] acl.c: Unable to lookup ‘metarouter’

4 16:33:04] ERROR[12116] acl.c: Cannot create socket
4 16:33:04] NOTICE[12116] chan_sip.c: The ‘username’ field for sip peers has been deprecated in favor of the term 'defaultuser’
4 16:33:05] NOTICE[12144] chan_sip.c: Peer ‘222’ is now Reachable. (118ms / 2000ms)
4 16:33:05] WARNING[12116] chan_dahdi.c: Ignoring any changes to ‘userbase’ (on reload) at line 23.
4 16:33:05] WARNING[12116] chan_dahdi.c: Ignoring any changes to ‘vmsecret’ (on reload) at line 31
4 16:33:05] WARNING[12116] chan_dahdi.c: Ignoring any changes to ‘hassip’ (on reload) at line 35.
4 16:33:05] WARNING[12116] chan_dahdi.c: Ignoring any changes to ‘hasiax’ (on reload) at line 39
4 16:33:05] WARNING[12116] chan_dahdi.c: Ignoring any changes to ‘hasmanager’ (on reload) at line 47
4 16:33:05] NOTICE[12144] chan_sip.c: Peer ‘111’ is now Reachable. (115ms / 2000ms)

13

Halo,

Untuk konfigurasi SIP, berikut beberapa masukan dari saya :

  • externip tidak perlu diset apabila asterisk langsung menggunakan IP Public
  • hilangkan nat=yes pada [general] section
  • Untuk konfigurasi SIP Account, coba ikutin panduan disini : SIP Account
  • Lakukan debug pada saat call dari 111 ke 222 bukan pada saat reload, ikuti panduan disini : SIP Debug