Call audio cutting in and out on grandstream fxo gateway

1

Hello everything seems to be working for almost a year now. The freepbx system is using a 4 channel fxo gateway with 4 hardware lines from the phone company. Just recently the audio keeps cutting in and out for unknown reasons. I don’t know how to check it and see what is wrong if it’s the freepbx server or the Grand stream box. all phones are Cisco sip ip phones. Doesn’t matter if it’s just one call or four calls it just keeps cutting in and out when talking and listening. Can someone please help me.

I’m not sure what other information to give please forgive me on that.

2

From the Asterisk CLI,

RTP set debug on

you can confirm if the gateway is sending packets.

3

Hello how can i do that, To set it?

4

use your keyboard :wink:

2 Likes
5

Hello no what i mean is how to turn on the debug to see the packets.

6

You just type in that string of characters at the Asterisk CLI. Do you know how to get to the Asterisk CLI?

7

No sorry i do not. I never did that before.

8

Do you know what a linux “shell” is?, maybe you heard of “putty” ? If none of the above, you probably need to spend some time in the Wiki linked to at the top of this page.

9

I can putty in. I’m not at where the pbx server is at. I’m 12 hours drive from it. I have remote access to it.

10

Do you mean this asterisk -rvvvv?

11

Yes, i do

12

What am i looking for?

13

bi directional packets, increasing sequenced and properly interlaced, just do it , you will see, try google perhaps?

14

when i tried to call out I’m getting this error i can see.

Connected to Asterisk 11.21.2 currently running on localhost (pid = 2499)
[2018-10-17 18:41:36] WARNING[2642]: chan_sip.c:4099 retrans_pkt: Timeout on 1283969925-1406338886-574133022 on non-critical invite transaction.
localhostCLI> Connected to Asterisk 11.21.2 currently running on localhost (pid = 2499)
localhostCLI> [2018-10-17 18:41:36] WARNING[2642]: chan_sip.c:4099 retrans_pkt: Timeout on 1283969925-1406338886-574133022 on non-critical invite transaction.
localhostCLI> localhostCLI>
No such command ‘Connected to Asterisk 11.21.2 currently running on localhost (pid = 2499)’ (type ‘core show help Connected to’ for other possible commands)
No such command ‘[2018-10-17 18:41:36] WARNING[2642]: chan_sip.c:4099 retrans_pkt: Timeout on 1283969925-1406338886-574133022 on non-critical invite transaction.’ (type ‘core show help [2018-10-17 18:41:36]’ for other possible commands)
No such command ‘localhostCLI>’ (type 'core show help localhostCLI>’ for other possible commands)

15

Don’t type anything that you don’t understand when in the CLI

16

Okay i got something else in there.

[2018-10-17 19:01:54] NOTICE[14505][C-0000044a]: res_rtp_asterisk.c:4444 ast_rtp_read: Unknown RTP codec 126 received from

17

It’s a notice that means the gateway is offering a codec that asterisk doesn’t know about, usually not a biggie, it will settle on a mutually agreeable one.

18

Maybe the codec on the grand streams are off.

In there grand stream user account says for audio settings i can select
GSM
G723.1
G729A/b
PCMU
PCMA

In another section it says

Voice Frames per TX: 64 (up to 10/20/32/64 for G711/G726/G723/other codecs respectively)

but there is nothing for G711 in that other section.

What should i set both of these to?

19

PCMU = ulaw = g711u
PCMA = alaw = g711a

20

So set it for PCMU? what about the other voice frames per tx?