SIP - no audio when calling within LAN but is in external

Asterisk Asterisk SIP
1

Hello.

I have problem with audio in my PBX. When I calling within LAN connection between SIPs I don’t have audio of one SIP (reciever). Excatly situation is when I calling from LAN to external SIP or PSTN phone number. Only audio is when I calling from external SIP to another external SIP.

I don’t change anything in dial plan, all is default. On my SIP server options I set NAT - server is behind NAT. Maybe it is misconfiguration about NAT.

My version of Asterisk is 11.23.1 and FrePBX 13.0.192.9.

When I calling I have logs:

[2017-07-10 15:11:44] ERROR[558][C-0000001d]: pbx.c:4360 ast_func_read: Function PJSIP_HEADER not registered
[2017-07-10 15:11:44] ERROR[558][C-0000001d]: pbx.c:4399 ast_func_read2: Function PJSIP_HEADER not registered
[2017-07-10 15:11:44] ERROR[558][C-0000001d]: pbx.c:4360 ast_func_read: Function PJSIP_HEADER not registered
[2017-07-10 15:11:44] ERROR[558][C-0000001d]: pbx.c:4399 ast_func_read2: Function PJSIP_HEADER not registered
[2017-07-10 15:11:44] ERROR[558][C-0000001d]: pbx.c:4360 ast_func_read: Function PJSIP_HEADER not registered
[2017-07-10 15:11:44] ERROR[558][C-0000001d]: pbx.c:4399 ast_func_read2: Function PJSIP_HEADER not registered
[2017-07-10 15:11:44] ERROR[558][C-0000001d]: pbx.c:4360 ast_func_read: Function PJSIP_HEADER not registered
[2017-07-10 15:11:44] ERROR[558][C-0000001d]: pbx.c:4399 ast_func_read2: Function PJSIP_HEADER not registered
[2017-07-10 15:11:44] ERROR[558][C-0000001d]: pbx.c:4360 ast_func_read: Function PJSIP_HEADER not registered
[2017-07-10 15:11:44] WARNING[558][C-0000001d]: pbx.c:4227 func_args: Can't find trailing parenthesis for function 'DB(DEVICE/110/dial'?
[2017-07-10 15:11:44] ERROR[558][C-0000001d]: pbx.c:4399 ast_func_read2: Function PJSIP_HEADER not registered
[2017-07-10 15:11:44] WARNING[558][C-0000001d]: pbx.c:4227 func_args: Can't find trailing parenthesis for function 'DB(DEVICE/110/dial'?
[2017-07-10 15:11:44] ERROR[558][C-0000001d]: pbx.c:4360 ast_func_read: Function PJSIP_HEADER not registered
[2017-07-10 15:11:44] ERROR[558][C-0000001d]: pbx.c:4399 ast_func_read2: Function PJSIP_HEADER not registered
[2017-07-10 15:11:46] NOTICE[558][C-0000001d]: res_rtp_asterisk.c:4476 ast_rtp_read: Unknown RTP codec 95 received from 'x.x.x.x:10288'

Whats mean “Can’t find trailing parenthesis for function ‘DB(DEVICE/110/dial’?”?

Does anybody known whats going on with that? I will be greatfull. If Youneed any other infomration abaout config, please let me know.

2

All function calls must end in “)”. Either there is an error in your provided dialplan, or there is a bug in FreePBX. We cannot deal with the latter.

However, that is not why you would get one way audio. That sounds like a NAT or firewall issue.

3

Hello again. Thank you, david551 for your answer.

About error “Can’t find trailing parenthesis for function ‘DB(DEVICE/110/dial’?”" is strength, I don’t change anything in config files, all is default.

What about “Unknown RTP codec 95 received from ‘x.x.x.x:10288’” error? I known it doesnt has to do with no audio, but will be nice when I get of rid of that message to.

In SIP server options and extensions I enabled NAT. Im going to read about Asterisk and NAT, but I will be greatfull for enty tips.

4

Unknown codec is generally a bug in the peer, and could be related to lack of audio.

However, some earlier versions of Asterisk had problems with peers that didn’t use the codec number it was expecting for RFC 2833, even though it has be “negotiated” correctly. One would need to see the sip set debug on output for the INVITE transaction to see whether codec 95 was definitely a bug in the peer.