* You are viewing the archive for May 28th, 2012

audiohook errors


I´m facing some issues on asterisk 1.8.10. I can see this on the console:

[May 28 15:46:19] ERROR[28099]: lock.c:438 __ast_pthread_mutex_unlock: audiohook.c line 705 (audio_audiohook_write_list): Error releasing mutex: Operation not permitted
[May 28 15:46:19] ERROR[28099]: lock.c:280 __ast_pthread_mutex_lock: audiohook.c line 688 (audio_audiohook_write_list): Error obtaining mutex: State not recoverable
[May 28 15:46:19] ERROR[28099]: lock.c:407 __ast_pthread_mutex_unlock: audiohook.c line 705 (audio_audiohook_write_list): mutex ‘&(audiohook)->lock’ freed more times than we’ve locked!

any ideas???

best regards

How to configure Asterisk 1.8 to run with IPv6 in a LAN network

On 2012-05-25 16:47, jose carlos de souza wrote:
> Hi,
> I need help.
> Sorry for my bad english, i’m from Brasil. And just starting to leran
> Linux because of Asterisk.
> I want to configure Asterisk 1.8 with ipv6 in my Lan network at home.
> I’ve done a clean install of it and SIP clients it’s working well in my
> home with Ipv4 (execept of the bad quality of the voice, that it’s
> coming out, but that’s not important, what i relly want to see it’s ipv6
> working)
> I’m using Debian 6.0.4 on a Virtual machine (VMware), Linphone 3.5.2 as
> my SIP client and a Router DIR-600 from D-LINK as DHCP sever to my LAN
> networking. (With a DHCP IP Address Range of: to
> What i change in configurations to work with ipv6 was:
> 1
> change my network interfaces in /etc/network/interfaces , i add this:
> iface eth0 inet6 static
> addres fe80::c0a8:6e
> netmask 64
> gateway fe80::c0a8:1
> ———————————————————————
> 2
> Change sip.conf
> UDPbindaddr = ::
> TCPbindaddr = ::
> ————————————————————————————————
> 3
> On Linphone check the box “use IPv6 insted of IPv4″
> Your SIP Identity: sip:9005@fe80::c0a8:6e>

>I alredy put the brackets. Like: 9006@[fe80::c0a8:6e] and still dosent work.

> I’m calling from a windows 7(the one with the virtual machine on it) to
> another windows 7 both with dual-stack ip protocol, and both are in my
> LAN network.
> The call’s only works when i use the IPv4. for example:
> when my linphone its configure to sip:9005@ and Proxy
> addres:
> I have created on Asterisk 10 SIP extensions, 9001 to 9010 on.
> ————————————————————————————————–
> I whant to know what im doing wrong.
> Or what files do i have to changes and where are they located? Wicth
> changes should i made?
> Thanks for the attention and help.

How to tell VPM presence without restarting?

The wct4xxp module will log kernel messages when starting up, indicating
whether a VPM module (VPM400 or VPM450) was found.

I have some systems that have been running long enough that the messages
files from the last reboot have long since been rotated out and deleted.
They are older systems running zaptel 1.2.27.

Is there any way using one of the zt tools or /proc to determine whether
a VPM module is present, without doing a restart of zaptel or the machine
to see the init log messages?

I’m trying to debug a problem for which this is relevant information.


Which combination of codecs are required?


In Voicemail.conf 

If I  am using

format=h263|gsm ,and i want to store only audio , then it is not storing.In log it shows that video is deposite less then 5 second. If i want to store video and audio both then it will store properly.

If am using  

format=gsm|h263 ,then my Xlite  softphone will go to haung.

I just want to store audio and video both or some time only audio .

1)Plz guide me which combination of codec will be usefull.

2)Is there is any serial number signifance in format,ie one time if i use as format=h263|gsm and second time i am using format=gsm|h263,why  is diffrence  come?


Durgesh Mishra

Rancore Technologies.