No Sound From Asterisk

Asterisk Asterisk Support
1

I have installed version
Asterisk 1.8.7.1-1digium2~oneiric built by pbuilder @ nighthawk on a x86_64 running Linux on 2011-10-18 17:33:22 UTC
on Ubuntu
Distributor ID: Ubuntu
Description: Ubuntu 11.10
Release: 11.10
Codename: oneiric

I have configured extensions for SIP channel.
I can see gsm codec in /usr/lib/asterisk/modules. SayDigits() function does play gsm file (I checked verbose output). I do not hear anything.
Operating system’s sound recorder works fine so it cannot be hardware issue.

I am new to this.

extension.conf entry:

exten => 6391,1,Answer()
exten => 6391,n,Wait(5)
exten => 6391,n,SayDigits(${EXTEN})
exten => 6391,n,Wait(2)
exten => 6391,n,Hangup()

Verbose log:

– Executing [6391@default:1] Answer(“SIP/10.99.108.101-00000010”, “”) in new stack
– Executing [6391@default:2] Wait(“SIP/10.99.108.101-00000010”, “5”) in new stack
– Executing [6391@default:3] SayDigits(“SIP/10.99.108.101-00000010”, “6391”) in new stack
– <SIP/10.99.108.101-00000010> Playing ‘digits/6.gsm’ (language ‘en’)
– <SIP/10.99.108.101-00000010> Playing ‘digits/3.gsm’ (language ‘en’)
– <SIP/10.99.108.101-00000010> Playing ‘digits/9.gsm’ (language ‘en’)
– <SIP/10.99.108.101-00000010> Playing ‘digits/1.gsm’ (language ‘en’)
– Executing [6391@default:4] Wait(“SIP/10.99.108.101-00000010”, “2”) in new stack
– Executing [6391@default:5] Hangup(“SIP/10.99.108.101-00000010”, “”) in new stack
== Spawn extension (default, 6391, 5) exited non-zero on ‘SIP/10.99.108.101-00000010’

2

Verbose console logs please. Also this is the wrong forum for support requests.

3

sorry for posting this here. I have added console and extension.conf info to the post. Please let me know where should I post these questions.

4

viewforum.php?f=1 (Asterisk Support)

I don’t see anything wrong, so I suspect you have a firewall problem.

5

I have a similar issue.

Here is a setup which works.

Local Intel Machine running windows 7 (IP: 192.168.1.2)
Virtual Machine running Ubuntu on top of Windows 7 (IP:192.168.139.183 - NAT). This machine has Asterisk running
Virtual Machine running Windows XP in top of Widows 7 (IP: 192.168.139.180). This machine is running X-Lite client

I have google voice integration with Asterisk.

X-Lite client registers with the Asterisk server on the other VM. I am able to make an outbound call from the X-Lite client to a landline (or any phone). Call quality is great.

Now, here is a setup which does not work

Buoyed by the success on local VMs, I created an Ubuntu instance on EC3 Amazon cloud and installed Asterisk. Did the same thing as I did on my desktop’s VM machine (as far as installing Asterisk and the .conf files).

I used the X-Lite on my home’s Virtual machine to connect to the Asterisk instance on Amazon’s server. It connects fine, also calls out fine, but there is no audio.

I have a feeling that I have to tweak the firewall or ports on the Amazon server. What could it be?

I have tried various settings with sip.conf and sip_nat.conf etc. Nothing seems to work.

6

[quote=“csahai”]I have installed version
extension.conf entry:

exten => 6391,1,Answer()
exten => 6391,n,Wait(5)
exten => 6391,n,SayDigits(${EXTEN})
exten => 6391,n,Wait(2)
exten => 6391,n,Hangup()
[/quote]

I have the same problem. Also I noticed that when I put Playback(hello) between the wait and saydigits, then all of the sudden all audio works. So maybe it could be fixed by playing back silence.

But I have the same problem with the call parking module. When I park a call, the parking lot is being played back, probably using the same app, so I don’t hear to what extension the call went to.

7

In my case the error was that I didn’t start with Answer().

This solves my case in the dialplan, but still not with the parked calls feature.

Anyways, I thought I should let ppl know in this thread.

8

I’m having a similar problem. Running Asterisk on CentOS, trying to get basic PBX working. Everything seems to run ok, the logs don’t seem to show any foul play, but when I call the number no sound comes through.

Here’s what I have in extensions.conf:
exten => 7144090267,1,Answer()
exten => 7144090267,n,Wait(5)
exten => 7144090267,n,SayDigits(${EXTEN})
exten => 7144090267,n,Playback(hello-world)
exten => 7144090267,n,Wait(3)
exten => 7144090267,n,Hangup()

Here’s what I see in the CLI when I dial in:
== Using SIP RTP CoS mark 5
– Executing [7144090267@inbound:1] Answer(“SIP/vitel-inbound-0000001e”, “”) in new stack
– Executing [7144090267@inbound:2] Wait(“SIP/vitel-inbound-0000001e”, “5”) in new stack
– Executing [7144090267@inbound:3] SayDigits(“SIP/vitel-inbound-0000001e”, “7144090267”) in new stack
– <SIP/vitel-inbound-0000001e> Playing ‘digits/7.gsm’ (language ‘en’)
– <SIP/vitel-inbound-0000001e> Playing ‘digits/1.gsm’ (language ‘en’)
– <SIP/vitel-inbound-0000001e> Playing ‘digits/4.gsm’ (language ‘en’)
– <SIP/vitel-inbound-0000001e> Playing ‘digits/4.gsm’ (language ‘en’)
– <SIP/vitel-inbound-0000001e> Playing ‘digits/0.gsm’ (language ‘en’)
– <SIP/vitel-inbound-0000001e> Playing ‘digits/9.gsm’ (language ‘en’)
– <SIP/vitel-inbound-0000001e> Playing ‘digits/0.gsm’ (language ‘en’)
– <SIP/vitel-inbound-0000001e> Playing ‘digits/2.gsm’ (language ‘en’)
– <SIP/vitel-inbound-0000001e> Playing ‘digits/6.gsm’ (language ‘en’)
– <SIP/vitel-inbound-0000001e> Playing ‘digits/7.gsm’ (language ‘en’)
– Executing [7144090267@inbound:4] Playback(“SIP/vitel-inbound-0000001e”, “hello-world”) in new stack
– <SIP/vitel-inbound-0000001e> Playing ‘hello-world.gsm’ (language ‘en’)
– Executing [7144090267@inbound:5] Wait(“SIP/vitel-inbound-0000001e”, “3”) in new stack
– Executing [7144090267@inbound:6] Hangup(“SIP/vitel-inbound-0000001e”, “”) in new stack
== Spawn extension (inbound, 7144090267, 6) exited non-zero on ‘SIP/vitel-inbound-0000001e’

Any ideas?

9

As before, firwall.

10

You’re right david55, it was a firewall issue! I hadn’t opened ports 10000-20000 yet, just did it. Now i’m getting sound. Thanks.