[Q] No sound from messages, but no probs with talking

Asterisk Asterisk Support
1

Hi all,

I have an asterisk setup within LinuxMCE, which uses a wrapper originally written for Pluto@Home. Everything set up fine, including the use of a linksys 3102 and cisco 7970 phone. I can call, I can accept calls, and I can access voice menus for voicemail etc.
The problem is that none of the asterisk generated messages are actually audible. In the log, I see that a message is being played, but I cannot hear anything.

I did not find anything useful googling around or asking at the LinuxMCE forums/IRC.

Could it have to do with codecs? If this is a recognizable issue, how would I fix it? If it is not, which logs and config files are useful for getting to the bottom of this?

Any thoughts greatly appreciated,

Mark

2

[quote=“domodude”]Hi all,
Could it have to do with codecs? If this is a recognizable issue, how would I fix it? If it is not, which logs and config files are useful for getting to the bottom of this?
[/quote]

Again, talking over the phones works great. Only the system-generated messages cannot be heard.

Here are some details about the system.

Running Asterisk 2.4.10, FreePBX 2.3.1.0, compiled for x86_64.
The sytem runs behind a firewall, which allows VOIP calls. Shouldn’t matter, as this problem pertains to system voice messages.

Log:
linuxmce@dcerouter:~$ sudo asterisk -rvvvvvvvvvv
Asterisk 1.4.10, Copyright © 1999 - 2007 Digium, Inc. and others.
Created by Mark Spencer markster@digium.com
Asterisk comes with ABSOLUTELY NO WARRANTY; type ‘core show warranty’ for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type ‘core show license’ for details.

== Parsing ‘/etc/asterisk/asterisk.conf’: Found
== Parsing ‘/etc/asterisk/extconfig.conf’: Found
Connected to Asterisk 1.4.10 currently running on dcerouter (pid = 7095)
Verbosity was 0 and is now 10
– Executing [*97@from-internal:1] Answer(“SIP/202-cc00ce60”, “”) in new stack
– Executing [*97@from-internal:2] Wait(“SIP/202-cc00ce60”, “1”) in new stack
– Executing [*97@from-internal:3] Macro(“SIP/202-cc00ce60”, “user-callerid|”) in new stack
– Executing [s@macro-user-callerid:1] NoOp(“SIP/202-cc00ce60”, “user-callerid: device 202”) in new stack
– Executing [s@macro-user-callerid:2] Set(“SIP/202-cc00ce60”, “AMPUSER=202”) in new stack
– Executing [s@macro-user-callerid:3] GotoIf(“SIP/202-cc00ce60”, “0?report”) in new stack
– Executing [s@macro-user-callerid:4] GotoIf(“SIP/202-cc00ce60”, “0?start”) in new stack
– Executing [s@macro-user-callerid:5] Set(“SIP/202-cc00ce60”, “REALCALLERIDNUM=202”) in new stack
– Executing [s@macro-user-callerid:6] NoOp(“SIP/202-cc00ce60”, “REALCALLERIDNUM is 202”) in new stack
– Executing [s@macro-user-callerid:7] Set(“SIP/202-cc00ce60”, “AMPUSER=202”) in new stack
– Executing [s@macro-user-callerid:8] Set(“SIP/202-cc00ce60”, “AMPUSERCIDNAME=Gigaset”) in new stack
– Executing [s@macro-user-callerid:9] GotoIf(“SIP/202-cc00ce60”, “0?report”) in new stack
– Executing [s@macro-user-callerid:10] Set(“SIP/202-cc00ce60”, “AMPUSERCID=202”) in new stack
– Executing [s@macro-user-callerid:11] Set(“SIP/202-cc00ce60”, “CALLERID(all)=“Gigaset” <202>”) in new stack
– Executing [s@macro-user-callerid:12] Set(“SIP/202-cc00ce60”, “REALCALLERIDNUM=202”) in new stack
– Executing [s@macro-user-callerid:13] NoOp(“SIP/202-cc00ce60”, "TTL: ARG1: ") in new stack
– Executing [s@macro-user-callerid:14] GotoIf(“SIP/202-cc00ce60”, “0?continue”) in new stack
– Executing [s@macro-user-callerid:15] Set(“SIP/202-cc00ce60”, “__TTL=64”) in new stack
– Executing [s@macro-user-callerid:16] GotoIf(“SIP/202-cc00ce60”, “1?continue”) in new stack
– Goto (macro-user-callerid,s,23)
– Executing [s@macro-user-callerid:23] NoOp(“SIP/202-cc00ce60”, “Using CallerID “Gigaset” <202>”) in new stack
– Executing [*97@from-internal:4] Macro(“SIP/202-cc00ce60”, “get-vmcontext|202”) in new stack
– Executing [s@macro-get-vmcontext:1] Set(“SIP/202-cc00ce60”, “VMCONTEXT=novm”) in new stack
– Executing [s@macro-get-vmcontext:2] GotoIf(“SIP/202-cc00ce60”, “0?200:300”) in new stack
– Goto (macro-get-vmcontext,s,300)
– Executing [s@macro-get-vmcontext:300] NoOp(“SIP/202-cc00ce60”, “”) in new stack
– Executing [*97@from-internal:5] MailboxExists(“SIP/202-cc00ce60”, “202@novm”) in new stack
– Executing [*97@from-internal:6] GotoIf(“SIP/202-cc00ce60”, “0?mbexist”) in new stack
– Executing [*97@from-internal:7] VoiceMailMain(“SIP/202-cc00ce60”, “”) in new stack
– <SIP/202-cc00ce60> Playing ‘vm-login’ (language ‘en’)

I hear nothing. I hang up, which gets:

-- Executing [h@from-internal:1] Macro("SIP/202-cc00ce60", "hangupcall") in new stack
-- Executing [s@macro-hangupcall:1] ResetCDR("SIP/202-cc00ce60", "w") in new stack
-- Executing [s@macro-hangupcall:2] NoCDR("SIP/202-cc00ce60", "") in new stack
-- Executing [s@macro-hangupcall:3] GotoIf("SIP/202-cc00ce60", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing [s@macro-hangupcall:6] GotoIf("SIP/202-cc00ce60", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing [s@macro-hangupcall:9] GotoIf("SIP/202-cc00ce60", "1?theend") in new stack
-- Goto (macro-hangupcall,s,11)
-- Executing [s@macro-hangupcall:11] Hangup("SIP/202-cc00ce60", "") in new stack

== Spawn extension (macro-hangupcall, s, 11) exited non-zero on ‘SIP/202-cc00ce60’ in macro ‘hangupcall’
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on ‘SIP/202-cc00ce60’

3

Hi in 99% of the time this is due to zaptel hardware being present but not fully configured.

Ian

4

In addition to what Ian said, it could also be the phones using g729 and you do not have either a g729 license, or the g729 encoded sound files. Or possibly you have the rtp ports firewalled on the asterisk server.

5

Dave and Ian,

Thanks for the replies. I do not have zaptel hardware, just the linksys. There IS a zaptel trunk present, without the hardware. This just came with the LinuxMCE standard configuration – could this cause problems, too?
I will look into the codec problem. Is it possible though that with a codec problem, talking works but the voice message don’t? In any case, is there a way to force phones to use a publicly available codec?

Thanks again
Mark

6

Dave and Ian,

Thanks for the replies. I do not have zaptel hardware, just the linksys. There IS a zaptel trunk present, without the hardware. This just came with the LinuxMCE standard configuration – could this cause problems, too?
I will look into the codec problem. Is it possible though that with a codec problem, talking works but the voice message don’t? In any case, is there a way to force phones to use a publicly available codec?
Everything on the asterisk side of the firewall is demilitarized; all LAN members can speak freely.

Thanks again
Mark

7

Since your phones are all on the lan and bandwidth should not be an issue try changing to use only the ulaw codec.

In sip.conf make sure the codecs are specified like this:

disallow=all
allow=ulaw