Unable to hear sound supposedly being played

Hi all, I just installed Asterisk 13.13.1 on CentOS 7 and everything is working fine, I can make a test call from one sip phone to another. When I tried to use the Background and Playback commands as a test it says the sound is “playing” but you cannot hear anything from the caller’s end. So I was wondering is this a problem to do with firewall, port forwarding, or something else?

Permissions are set to read/write/allow executions so that can’t be a problem.

This is from the CLI

– Executing [6123@from-internal:1] Playback(“SIP/demo-bob-00000001”, “you-entered”) in new stack
– <SIP/demo-bob-00000001> Playing ‘you-entered.slin’ (language ‘en’)
– Auto fallthrough, channel ‘SIP/demo-bob-00000001’ status is ‘UNKNOWN’

So it appears to “play” with no errors, the file is found, I made sure to select MP3s and WAVS during installation so I don’t understand what else it could be. Any help would be appreciated.

BTW, the sip.conf is here:

friends_internal
type=friend
host=dynamic
context=from-internal
;disallow=all
allow=ulaw
allow=gsm

demo-alice
secret=password

demo-bob
secret=password; put a strong, unique password here instead

Extensions.conf is here

[from-internal]
exten=>6001,1,Dial(SIP/demo-alice,20)
exten=>6002,1,Dial(SIP/demo-bob,20)
exten => 6123,1,Playback(you-entered)

Hello

I can suggest you 2 things:

  • try to use another codec
  • in the sip.conf add the following line: nat=yes and then press a sip reload from CLI
1 Like

Hi, I appreciate the reply but I tried both and it did not work. To try a different codec I replaced allow=ulaw with allow=alaw, is there another codec to try? I also tried with both mp3 and wav, no luck.

Also, I put nat=yes in sip.conf, actually it came up with an error about deprecation and told me to put nat=force_rport,comedia
so I put that instead (I wasn’t sure where so I put it in both contexts):

friends_internal
type=friend
host=dynamic
context=from-internal
;disallow=all
;allow=ulaw
allow=alaw
allow=gsm
nat=force_rport,comedia

demo-alice
secret=blagobung
allow=gsm
nat=force_rport,comedia

I still get the same message that it “plays” but there is no audio. This system already works to make calls from one sip phone to another. I don’t know, is there something to the error message about “auto fallthrough”?

– <SIP/demo-alice-0000001c> Playing ‘you-entered.slin’ (language ‘en’)
– Auto fallthrough, channel ‘SIP/demo-alice-0000001c’ status is ‘UNKNOWN’

Also: if it works for phone calls between two different sip phones then the codec is working ok as is right? Or can the codec work for voice but not for playing sounds?

None of the audio codecs should, themselves, have any problems playing back audio prompts.

I just realized when making a call from sip phone to sip phone the call ends after only 15 seconds with this error message:

[Jan 27 06:18:53] WARNING[31206]: chan_sip.c:4061 retrans_pkt: Retransmission timeout reached on transmission 636f41e653e34cc6a127a1cd9730afcb for seqno 1602 (Critical Response) – See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions

So I am guessing this has to do with why the sound does not play either. Something with NAT? I even disabled firewalld on CentOS but still no luck. I already have tried nat=force_rport and nat=force_rport,comedia in sip.conf but this did not help either. If it is some NAT, firewall, issue can anyone think of the best solution?

You never got an ACK, which is typically a NAT problem; you are sending an unrouttable address in a Contact: header.

1 Like

Asterisk is running on CentOS 7 on a VM (Virtualbox) on Windows 10. I am
just experimenting like this until I see its working and then I’ll put it
on a VS somewhere. So, where do I need to forward all the ports? On
Windows 10, CentOS, or both? I disabled firewall on Windows 10 and CentOS
and it still does not work. Unfortunately on Virtualbox you have to
forward each port one by one. So is that where the ports need to be
forwarded?