Hi guys,
I'm trying to write hangup causes from asterisk into the CDR record.
Using version 1.4.24.1 at the moment, but no joy so far.
Has anyone implemented this?
Neeraj Chand
Support Analyst
Fiji Islands Australia
T: +6793342526 T: +61388924326
M:+6799344012 New Zealand
www.ocis.com.au T: +649 980 7022
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of
asterisk-users-request at lists.digium.com
Sent: Thursday, 21 May 2009 8:28 PM
To: asterisk-users at lists.digium.com
Subject: asterisk-users Digest, Vol 58, Issue 56
Send asterisk-users mailing list submissions to
asterisk-users at lists.digium.com
To subscribe or unsubscribe via the World Wide Web, visit
http://lists.digium.com/mailman/listinfo/asterisk-users
or, via email, send a message with subject or body 'help' to
asterisk-users-request at lists.digium.com
You can reach the person managing the list at
asterisk-users-owner at lists.digium.com
When replying, please edit your Subject line so it is more specific
than "Re: Contents of asterisk-users digest..."
Today's Topics:
1. Re: DAHDI fun and games (Danny Nicholas)
2. Re: Step-by-Step Asterisk and MeetMe Help (Tzafrir Cohen)
3. Re: Channels configuration with DAHDI (Dave Fullerton)
4. Re: ...is circuit busy message (Jeff LaCoursiere)
5. Re: Dialplan Priorities and Sort Order... (Alex Samad)
6. Re: Step-by-Step Asterisk and MeetMe Help (Jimmy Ezell)
7. Re: Open source SIP client (marek cervenka)
8. Re: Step-by-Step Asterisk and MeetMe Help (Jonathan Thurman)
9. Re: Step-by-Step Asterisk and MeetMe Help (ContactTel Business)
10. Re: Channels configuration with DAHDI (Daniel Bareiro)
11. 1.4.24.1 -> 1.6.0.9: segfault (sean darcy)
12. Voicemail playback NEWEST first vs. OLDEST first (Karl Fife)
13. Re: Step-by-Step Asterisk and MeetMe Help (Jeff LaCoursiere)
14. Re: Step-by-Step Asterisk and MeetMe Help (ContactTel Business)
15. Bridging INBOUND PRI to OUTBOUND PRI fails with Monitor()
(Barry L. Kline)
16. PSTN Connection (Manoj Panicker - FOES)
17. Re: Open source SIP client (Alex Samad)
18. Re: PSTN Connection (Paul Hales)
19. interruption in queue (Rilawich Ango)
20. Re: PSTN Connection (--[ UxBoD ]--)
21. Polycom Productivity Suite (Matt Darnell)
22. Fwd: Asterisk CCM, CME Integration (Arun Kumar)
----------------------------------------------------------------------
Message: 1
Date: Wed, 20 May 2009 16:07:48 -0500
From: "Danny Nicholas" <danny at debsinc.com>
Subject: Re: [asterisk-users] DAHDI fun and games
To: "'Asterisk Users Mailing List - Non-Commercial
Discussion'"
<asterisk-users at lists.digium.com>
Message-ID: <2897B95E2E394A7D9FAD95BFF31BF554 at db0002>
Content-Type: text/plain; charset="us-ascii"
Using "r/m" because DAHDI takes 10-15 seconds to get TELCO rings.
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Dave
Fullerton
Sent: Wednesday, May 20, 2009 4:03 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DAHDI fun and games
Danny Nicholas wrote:> Hi Listers,
>
> I'm running 1.4.25-rc1 on opensuse 11.0 with
> dahdi-linux-2.1.0.3, dahdi-tools-2.1.0.2, libpri-1.4.7 and
snapdsp.0.0.2.> Incoming calls work fine. Outgoing calls made directly (exten =>
> s,1,Dial(DAHDI/G1) then number work fine. The problem I have is
trying to> let Asterisk make the call (exten => s,1,Dial(DAHDI/G1/5551212,,r).
If I> use "m" (moh) the music plays 5-8 seconds after the other end
picks
up.> When using "r", I get 2-3 rings after other end picks up.
I've went
through> every flavor of dahdi-linux from 2.0.0 to 2.1.0-rc4 (which crashed me)
with> no joy. Any suggestions? Hardware is Dell Poweredge 1650/1550 and
> TDM410P/TDM400P.
Any reason you're using the r/m option at all? Since this is an analog
card I would leave the r/m off and just let asterisk use the in-band
progress from the telco.
-Dave
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
------------------------------
Message: 2
Date: Thu, 21 May 2009 00:11:24 +0300
From: Tzafrir Cohen <tzafrir.cohen at xorcom.com>
Subject: Re: [asterisk-users] Step-by-Step Asterisk and MeetMe Help
To: asterisk-users at lists.digium.com
Message-ID: <20090520211124.GM3227 at xorcom.com>
Content-Type: text/plain; charset=us-ascii
On Wed, May 20, 2009 at 01:07:25PM -0700, Jimmy Ezell wrote:
> multi-processor machine ( I had to remember to specify smp for the
kernel)
I repeat: why bother with such an old system? Really?
Recall the comment from the book. That book had nothing really specific
to Centos 4. Why do you shoot yourself in the foot by installing Centos4
now?
(not to mention Zaptel)
--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.cohen at xorcom.com
+972-50-7952406 mailto:tzafrir.cohen at xorcom.com
http://www.xorcom.com iax:guest at local.xorcom.com/tzafrir
------------------------------
Message: 3
Date: Wed, 20 May 2009 17:12:04 -0400
From: Dave Fullerton <dfullertasterisk at shorelinecontainer.com>
Subject: Re: [asterisk-users] Channels configuration with DAHDI
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <4A147224.6060302 at shorelinecontainer.com>
Content-Type: text/plain; charset=UTF-8; format=flowed
Daniel Bareiro wrote:> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
>
> Hi Tzafrir.
>
> El mi?rcoles 20 de mayo del 2009 a las 10:00:46 -0300,
> Tzafrir Cohen escribi?:
>
>> On Wed, May 20, 2009 at 07:03:15AM -0300, Daniel Bareiro wrote:
>
>> Hint: you don't need to set 'signalling' for analog
channels. Or just
>> set it explicitly to "auto". This is for Asterisk >= 1.6.0
. Simply
>> reduces the complication a bit...
>
> Thanks for the tip. I will remember it for when I use Asterisk 1.6 :-)
>
>>> I load the modules wctdm and dahdi. But when I execute in Asterisk
>>> CLI "dahdi show channels", I get the following error
message:
>>>
>>>
>>> No such command 'dahdi show channels' (type 'help dahdi
show' for
>>> other possible commands)
>
>> Try running:
>>
>> asterisk -r
>>
>> and in that prompt:
>>
>> module unload chan_dadhi.so
>> module load chan_dadhi.so
>>
>> and tell us the output you got.
>
>
> # asterisk -r
> Asterisk 1.4.24.1, Copyright (C) 1999 - 2008 Digium, Inc. and others.
> Created by Mark Spencer <markster at 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.
>
=======================================================================>
Connected to Asterisk 1.4.24.1 currently running on alderamin (pid >
19777)> Verbosity is at least 7
> alderamin*CLI>
> alderamin*CLI> module unload chan_dadhi.so
> alderamin*CLI> module load chan_dadhi.so
> [May 20 17:52:19] WARNING[10345]: loader.c:359 load_dynamic_module:
> Error loading module 'chan_dadhi.so':
> /usr/lib/asterisk/modules/chan_dadhi.so: cannot open shared object
file:> No such file or directory
> [May 20 17:52:19] WARNING[10345]: loader.c:653 load_resource: Module
> 'chan_dadhi.so' could not be loaded.
> alderamin*CLI>
>
>
> Mmmm... it would seem to be a bug:
>
> /usr/lib/asterisk/modules/chan_dadhi.so: cannot open shared object
file:> No such file or directory
>
Sounds like DAHDI was installed/compiled *after* Asterisk was compiled.
Recompile Asterisk again and make sure
/usr/lib/asterisk/modules/chan_dahdi.so is created when you make
install.
-Dave
------------------------------
Message: 4
Date: Wed, 20 May 2009 21:16:03 +0000 (UTC)
From: Jeff LaCoursiere <jeff at jeff.net>
Subject: Re: [asterisk-users] ...is circuit busy message
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <alpine.BSF.2.00.0905202112310.46773 at phoenix.jeff.net>
Content-Type: TEXT/PLAIN; charset=US-ASCII; format=flowed
On Wed, 20 May 2009, John Regal wrote:
> Thanks for the reply and apologize for the double post. My original
post> landed in another thread and thought it may have been missed...
>
> I questioned my voip provider before posting and they told me they
have> other asterisk customers that are making hundreds of simultaneous
calls> without problems with the same account type that I have. They
indicated that> they do not limit my simultaneous connections. I am now going to have
them> trace my connection but hoped to learn of a possible configuration
setting I> could check first.
> Thanks again for the help.
The message you are hearing is coming from your asterisk server, and it
is
because there was either no response to your call attempt or your call
attempt weas refused by your VoIP provider. In the "no response"
scenario
it may be because you are strapped for bandwidth, but if you were THAT
strapped you would have many additional problems, and the calls that
were
going through would have serious audio problems.
So I would focus on the idea that your VoIP provider is refusing the
calls
that are failing.
You could prove this with a packet trace. Use wireshark/tcpdump to
capture your attempts and see if you can find the session that is
refused.
You will either see the refusal come back or the request timed out...
Good luck,
j
------------------------------
Message: 5
Date: Thu, 21 May 2009 07:35:26 +1000
From: Alex Samad <alex at samad.com.au>
Subject: Re: [asterisk-users] Dialplan Priorities and Sort Order...
To: asterisk-users at lists.digium.com
Message-ID: <20090520213526.GE11230 at samad.com.au>
Content-Type: text/plain; charset="us-ascii"
On Wed, May 20, 2009 at 03:16:34PM -0400, M Hulber
wrote:>
>
> Alex Samad wrote:
> > On Tue, May 19, 2009 at 02:05:47PM -0400, M Hulber wrote:
> >
[snip]
> >
> I left the busy after dial because this is what the original poster
> had. In this case, if the channel does not get hungup then the next
> execution will be a busy, letting the caller know the call was not
> completed. In the dialplan macro I normally use it checks for the
call > status and acts accordingly as seen below. If you are new to Asterisk
> syntax this is probably confusing. If you are following it, I don't
> exit on a BUSY because I frequently get a BUSY when there is actually
a > congestion or channel problem. Anyhow, how often is a line actually
> busy these days?
>
> exten => s,n,Set(DIALS1="IAX2/xxxxxxxx at
carrier1-out/${ARG1},90,T")
> exten => s,n,Set(DIALS2="IAX2/xxxxxxxx at
carrier2-out/${ARG1},90,T")
> exten => s,n,Set(DIALS3="SIP/${ARG1}@carrier3-out,90,T")
>
> exten => s,n,Set(DialNum=3)
> exten => s,n,Set(DialCount=0)
>
> exten => s,n(dial),Set(DialCount=$[1 + ${DialCount}])
> exten => s,n,GotoIf($[${DialCount} > ${DialNum}]?h,1)
> exten => s,n,Dial(${DIALS${DialCount}})
> exten => s,n,Goto(dial)
>
> exten => s-CONGESTION,1,Congestion(5)
> exten => s-CONGESTION,n,Macro(rhangup)
>
> exten => s-BUSY,1,Playtones(busy)
> exten => s-BUSY,n,Busy(5)
> exten => s-BUSY,n,Macro(rhangup)
>
> exten => h,1,GotoIf($[${DIALSTATUS} = BUSY]?s-BUSY,1)
> exten => h,n,GotoIf($[${DIALSTATUS} = CHANUNAVAIL]?s-CONGESTION,1)
> exten => h,n,GotoIf($[${DIALSTATUS} = CONGESTION]?s-CONGESTION,1)
> exten => h,n,Macro(rhangup)
>
> exten => t,1,Macro(rhangup)
>
Wow thats a neet way to dial multiple providers, can you make it into a
macro and passin an array of numbers ? and maybe another param to
specify how many elements in the array ?
[snip]
>
-------------- next part --------------
A non-text attachment was scrubbed...
Name: not available
Type: application/pgp-signature
Size: 197 bytes
Desc: Digital signature
Url :
http://lists.digium.com/pipermail/asterisk-users/attachments/20090521/ec
fe0455/attachment-0001.pgp
------------------------------
Message: 6
Date: Wed, 20 May 2009 14:36:38 -0700
From: "Jimmy Ezell" <jezell at hmhca.com>
Subject: Re: [asterisk-users] Step-by-Step Asterisk and MeetMe Help
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Message-ID:
<E77AB304D41C084090D498682873252FC1B606 at nts-10.ca.hmhengineers.com>
Content-Type: text/plain; charset="iso-8859-1"
>On Wed, May 20, 2009 at 01:07:25PM -0700, Jimmy Ezell wrote:
>
>> multi-processor machine ( I had to remember to specify smp
>for the kernel)
>
>I repeat: why bother with such an old system? Really?
>
>Recall the comment from the book. That book had nothing really specific
>to Centos 4. Why do you shoot yourself in the foot by
>installing Centos4
>now?
>
>(not to mention Zaptel)
>
>--
> Tzafrir Cohen
Tzafrir thanks for the comments. I am not done playing with this and in
the end I may well use newer software as you suggest.
According to wikipedia CentOS 4.7 was released OCT. 2008 (7 months ago)
is that really consider that old? I am looking to setup a phone system
that I would hope would not require any major software upgrades for many
years.
Jimmy>
------------------------------
Message: 7
Date: Thu, 21 May 2009 01:17:24 +0200 (CEST)
From: marek cervenka <cervajs at fpf.slu.cz>
Subject: Re: [asterisk-users] Open source SIP client
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <alpine.LRH.2.00.0905210116570.17084 at axpsu.fpf.slu.cz>
Content-Type: TEXT/PLAIN; charset=US-ASCII; format=flowed
> can anybody help me to give Opensource SIP client information which
can be modified as per our requirment
http://www.qutecom.org
---------------------------------------
Marek Cervenka
======================================
------------------------------
Message: 8
Date: Wed, 20 May 2009 16:33:15 -0700
From: Jonathan Thurman <jthurman42 at gmail.com>
Subject: Re: [asterisk-users] Step-by-Step Asterisk and MeetMe Help
To: asterisk-users at lists.digium.com
Message-ID:
<f7cbcc6e0905201633n75f67015m7143d0652c691dd0 at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
>From the front page ( http://wiki.centos.org/FrontPage ):
"*What is CentOS?*
CentOS is an Enterprise Linux distribution based on the freely available
sources from Red Hat Enterprise
Linux<ftp://ftp.redhat.com/pub/redhat/linux/enterprise/>.
Each CentOS version is supported for 7 years (by means of security
updates).
A new CentOS version is released every 2 years and each CentOS version
is
regularly updated (every 6 months) to support newer hardware. This
results
in a secure, low-maintenance, reliable, predictable and reproducible
Linux
environment."
CentOS 4 ( http://wiki.centos.org/FAQ/CentOS4 ):
"We intend to support CentOS-4 updates until Feb 29, 2012"
CentOS 5 ( http://wiki.centos.org/FAQ/CentOS5 ):
"We intend to support CentOS 5 until Mar 31st, 2014"
So if you don't want major upgrades for a while you might want to go
with
the latest version. To put it into Microsoft terms... the minor
version is
like a service pack. So CentOS 4.7 is really a base lined version 4,
service pack 7. You get the new features in major releases (like there
are
no more "smp" kernels in 5 to deal with)
-Jonathan
On Wed, May 20, 2009 at 2:36 PM, Jimmy Ezell <jezell at hmhca.com> wrote:
>
> >On Wed, May 20, 2009 at 01:07:25PM -0700, Jimmy Ezell wrote:
> >
> >> multi-processor machine ( I had to remember to specify smp
> >for the kernel)
> >
> >I repeat: why bother with such an old system? Really?
> >
> >Recall the comment from the book. That book had nothing really
specific> >to Centos 4. Why do you shoot yourself in the foot by
> >installing Centos4
> >now?
> >
> >(not to mention Zaptel)
> >
> >--
> > Tzafrir Cohen
>
> Tzafrir thanks for the comments. I am not done playing with this and
in> the end I may well use newer software as you suggest.
>
> According to wikipedia CentOS 4.7 was released OCT. 2008 (7 months
ago) is> that really consider that old? I am looking to setup a phone system
that I> would hope would not require any major software upgrades for many
years.>
>
> Jimmy
> >
>
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20090520/12
dc7cf2/attachment.htm
------------------------------
Message: 9
Date: Wed, 20 May 2009 20:00:06 -0400
From: "ContactTel Business" <lists at contacttel.com>
Subject: Re: [asterisk-users] Step-by-Step Asterisk and MeetMe Help
To: "'Asterisk Users Mailing List - Non-Commercial
Discussion'"
<asterisk-users at lists.digium.com>
Message-ID: <004901c9d9a7$19d16950$4d743bf0$@com>
Content-Type: text/plain; charset="us-ascii"
Many years in telecom and computer world is around 100 year in real
life..
10 years ago i was a millionaire in the dot com boom and 24 years old
with a
P2 300 computer.., 20 years ago i was military engineer and running on
3.76
MHz 386's amber screens.. last year it was dual cores, today its
quad/opt
cores, and tomorrow morning it's going to be quantum physics/organic
computers and VOIP will be of the past, since Voice over Something else
will
arrive.
You can't put a system and let it go for 3-4 years unless you don't have
any
growth, ( new drives = new technology , IDE/SATA/ISCSI) new RAM/ NEW
CPU/
etc all these need software upgrades eventually..
As far as my personal experience i reformat my desktops /fully, semi
annually, and all servers get a facelift every other month ( new glib
for
new freeswitch updates, new ZAP hardware ? then you need new zaptel..
wait
zaptel aka dhadi needs X, X needs Y.. and so on..
Mike
ContacTel.COM
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Jonathan
Thurman
Sent: May-20-09 7:33 PM
To: asterisk-users at lists.digium.com
Subject: Re: [asterisk-users] Step-by-Step Asterisk and MeetMe Help
>From the front page ( http://wiki.centos.org/FrontPage ):
"What is CentOS?
CentOS is an Enterprise Linux distribution based on the freely available
<ftp://ftp.redhat.com/pub/redhat/linux/enterprise/> sources from Red
Hat
Enterprise Linux. Each CentOS version is supported for 7 years (by means
of
security updates). A new CentOS version is released every 2 years and
each
CentOS version is regularly updated (every 6 months) to support newer
hardware. This results in a secure, low-maintenance, reliable,
predictable
and reproducible Linux environment."
CentOS 4 ( http://wiki.centos.org/FAQ/CentOS4 ):
"We intend to support CentOS-4 updates until Feb 29, 2012"
CentOS 5 ( http://wiki.centos.org/FAQ/CentOS5 ):
"We intend to support CentOS 5 until Mar 31st, 2014"
So if you don't want major upgrades for a while you might want to go
with
the latest version. To put it into Microsoft terms... the minor
version is
like a service pack. So CentOS 4.7 is really a base lined version 4,
service pack 7. You get the new features in major releases (like there
are
no more "smp" kernels in 5 to deal with)
-Jonathan
On Wed, May 20, 2009 at 2:36 PM, Jimmy Ezell <jezell at hmhca.com> wrote:
>On Wed, May 20, 2009 at 01:07:25PM -0700, Jimmy Ezell wrote:
>
>> multi-processor machine ( I had to remember to specify smp
>for the kernel)
>
>I repeat: why bother with such an old system? Really?
>
>Recall the comment from the book. That book had nothing really specific
>to Centos 4. Why do you shoot yourself in the foot by
>installing Centos4
>now?
>
>(not to mention Zaptel)
>
>--
> Tzafrir Cohen
Tzafrir thanks for the comments. I am not done playing with this and in
the
end I may well use newer software as you suggest.
According to wikipedia CentOS 4.7 was released OCT. 2008 (7 months ago)
is
that really consider that old? I am looking to setup a phone system
that I
would hope would not require any major software upgrades for many years.
Jimmy
>
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20090520/05
1c70dd/attachment.htm
------------------------------
Message: 10
Date: Wed, 20 May 2009 21:19:18 -0300
From: Daniel Bareiro <daniel-listas at gmx.net>
Subject: Re: [asterisk-users] Channels configuration with DAHDI
To: asterisk-users at lists.digium.com
Message-ID: <slrnh197i3.3j6.daniel-listas at marian.freesoftware.org>
Content-Type: text/plain; charset=UTF-8
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Hi Dave.
El mi?rcoles 20 de mayo del 2009 a las 18:12:04 -0300,
Dave Fullerton escribi?:
>>>> I load the modules wctdm and dahdi. But when I execute in
Asterisk
>>>> CLI "dahdi show channels", I get the following error
message:
>>>>
>>>>
>>>> No such command 'dahdi show channels' (type 'help
dahdi show' for
>>>> other possible commands)
>>> Try running:
>>>
>>> asterisk -r
>>>
>>> and in that prompt:
>>>
>>> module unload chan_dadhi.so
>>> module load chan_dadhi.so
>>>
>>> and tell us the output you got.
>> # asterisk -r
>> Asterisk 1.4.24.1, Copyright (C) 1999 - 2008 Digium, Inc. and others.
>> Created by Mark Spencer <markster at 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.
>>
=======================================================================>>
Connected to Asterisk 1.4.24.1 currently running on alderamin (pid >>
19777)>> Verbosity is at least 7
>> alderamin*CLI>
>> alderamin*CLI> module unload chan_dadhi.so
>> alderamin*CLI> module load chan_dadhi.so
>> [May 20 17:52:19] WARNING[10345]: loader.c:359 load_dynamic_module:
>> Error loading module 'chan_dadhi.so':
>> /usr/lib/asterisk/modules/chan_dadhi.so: cannot open shared object
file:>> No such file or directory
>> [May 20 17:52:19] WARNING[10345]: loader.c:653 load_resource: Module
>> 'chan_dadhi.so' could not be loaded.
>> alderamin*CLI>
>>
>>
>> Mmmm... it would seem to be a bug:
>>
>> /usr/lib/asterisk/modules/chan_dadhi.so: cannot open shared object
file:>> No such file or directory
> Sounds like DAHDI was installed/compiled *after* Asterisk was
> compiled. Recompile Asterisk again and make sure
> /usr/lib/asterisk/modules/chan_dahdi.so is created when you make
> install.
Mmmm... but I believe that it had done already in that order. In fact, I
reviewed the existence of the module and it was in the directory. For
that
reasonI said that perhaps it was bug by the following thing:
[May 20 20:49:07] WARNING[23599]: loader.c:359 load_dynamic_module:
Error loading module 'chan_dadhi.so':
/usr/lib/asterisk/modules/chan_dadhi.so: cannot open shared object file:
^^^^^^^^^^^^^
No such file or directory
[May 20 20:49:07] WARNING[23599]: loader.c:653 load_resource: Module
'chan_dadhi.so' could not be loaded.
Apparently Asterisk is looking for the module using an incorrect name.
Whatever happens, I compile Asterisk again but I got the same error
message.
Thanks for your reply.
Regards,
Daniel
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.9 (GNU/Linux)
iEYEARECAAYFAkoUnkMACgkQZpa/GxTmHTfKeACffr1Q2vgJnyDVrA+hCN3DtHd5
e4UAoJUSl6JjjMCX+SiUA2/cyHJhwtfN
=wCSV
-----END PGP SIGNATURE-----
------------------------------
Message: 11
Date: Wed, 20 May 2009 20:22:16 -0400
From: sean darcy <seandarcy2 at gmail.com>
Subject: [asterisk-users] 1.4.24.1 -> 1.6.0.9: segfault
To: asterisk-users at lists.digium.com
Message-ID: <gv26rp$38a$1 at ger.gmane.org>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
I'm testing an upgrade of an i686 production machine running 1.4.24.1 to
1.6.0.9. I've installed dahdi-linux-2.1.0.4.
But:
asterisk -cvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvv
Asterisk 1.6.0.9, Copyright (C) 1999 - 2008 Digium, Inc. and others.
Created by Mark Spencer <markster at 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
== Parsing '/etc/asterisk/logger.conf': == Found
Asterisk Event Logger Started /var/log/asterisk/event_log
Asterisk Dynamic Loader Starting:
== Parsing '/etc/asterisk/modules.conf': == Found
== Parsing '/etc/asterisk/dnsmgr.conf': == Found
== Parsing '/etc/asterisk/http.conf': == Found
................
== Parsing '/etc/asterisk/manager.conf': == Found
[May 20 18:43:54] NOTICE[750]: manager.c:3903 __init_manager: Invalid
keyword <displaysystemname> = <yes> in manager.conf [general
........
== Parsing '/etc/asterisk/smdi.conf': == Found
[May 20 18:43:54] NOTICE[750]: res_smdi.c:1272 load_module: No SMDI
interfaces are available to listen on, not starting SMDI listener.
...........
== Parsing '/etc/asterisk/musiconhold.conf': == Found
[May 20 18:43:54] WARNING[750]: res_musiconhold.c:1496 load_moh_classes:
A directory must be specified for class 'default'!
[May 20 18:43:54] WARNING[750]: res_musiconhold.c:1657 load_module: No
music on hold classes configured, disabling music on hold.
== Registered application 'MusicOnHold'
...............
== Registered application 'DateTime'
app_sayunixtime.so => (Say time)
== Registered application 'SetCallerPres'
app_setcallerid.so => (Set CallerID Presentation Application)
== Registered file format gsm, extension(s) gsm
format_gsm.so => (Raw GSM data)
== Registered application 'BackgroundDetect'
app_talkdetect.so => (Playback with Talk Detection)
Segmentation fault
strace was little help:
strace asterisk -c
.......
.open("/usr/lib/asterisk/modules/format_gsm.so", O_RDONLY) = 12
read(12, "\177ELF\1\1\1\0\0\0\0\0\0\0\0\0\3\0\3\0\1\0\0\0
\10\0\000"...,
512) = 512
fstat64(12, {st_mode=S_IFREG|0755, st_size=150128, ...}) = 0
mmap2(NULL, 16240, PROT_READ|PROT_EXEC, MAP_PRIVATE|MAP_DENYWRITE, 12,
0) = 0xb7364000
mmap2(0xb7367000, 4096, PROT_READ|PROT_WRITE,
MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 12, 0x2) = 0xb7367000
close(12) = 0
gettimeofday({1242859458, 600093}, NULL) = 0
stat64("/etc/localtime", {st_mode=S_IFREG|0644, st_size=1267, ...}) =
0
stat64("/etc/localtime", {st_mode=S_IFREG|0644, st_size=1267, ...}) =
0
stat64("/etc/localtime", {st_mode=S_IFREG|0644, st_size=1267, ...}) =
0
gettid() = 921
futex(0x81abca4, 0x5 /* FUTEX_??? */, 1) = 1
futex(0x8195988, FUTEX_WAKE, 1) = 1
futex(0x81abca4, 0x5 /* FUTEX_??? */, 1) = 1
futex(0x8195988, FUTEX_WAKE, 1) = 1
.open("/usr/lib/asterisk/modules/app_talkdetect.so", O_RDONLY) = 12
read(12, "\177ELF\1\1\1\0\0\0\0\0\0\0\0\0\3\0\3\0\1\0\0\0\360\v\0"...,
512) = 512
fstat64(12, {st_mode=S_IFREG|0755, st_size=155069, ...}) = 0
mmap2(NULL, 12176, PROT_READ|PROT_EXEC, MAP_PRIVATE|MAP_DENYWRITE, 12,
0) = 0xb7361000
mmap2(0xb7363000, 4096, PROT_READ|PROT_WRITE,
MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 12, 0x1) = 0xb7363000
close(12) = 0
gettimeofday({1242859458, 601194}, NULL) = 0
stat64("/etc/localtime", {st_mode=S_IFREG|0644, st_size=1267, ...}) =
0
stat64("/etc/localtime", {st_mode=S_IFREG|0644, st_size=1267, ...}) =
0
stat64("/etc/localtime", {st_mode=S_IFREG|0644, st_size=1267, ...}) =
0
gettid() = 921
futex(0x81abca4, 0x5 /* FUTEX_??? */, 1) = 1
futex(0x8195988, FUTEX_WAKE, 1) = 1
futex(0x81abca4, 0x5 /* FUTEX_??? */, 1) = 1
futex(0x8195988, FUTEX_WAKE, 1) = 1
.open("/usr/lib/asterisk/modules/app_random.so", O_RDONLY) = 12
read(12, "\177ELF\1\1\1\0\0\0\0\0\0\0\0\0\3\0\3\0\1\0\0\0\260\10"...,
512) = 512
fstat64(12, {st_mode=S_IFREG|0755, st_size=135210, ...}) = 0
mmap2(NULL, 9356, PROT_READ|PROT_EXEC, MAP_PRIVATE|MAP_DENYWRITE, 12, 0)
= 0xb735e000
mmap2(0xb7360000, 4096, PROT_READ|PROT_WRITE,
MAP_PRIVATE|MAP_FIXED|MAP_DENYWRITE, 12, 0x1) = 0xb7360000
close(12) = 0
--- SIGSEGV (Segmentation fault) @ 0 (0) ---
+++ killed by SIGSEGV +++
Process 921 detached
Anyone else seen this?
sean
------------------------------
Message: 12
Date: Wed, 20 May 2009 19:28:19 -0500
From: "Karl Fife" <karlfife at gmail.com>
Subject: [asterisk-users] Voicemail playback NEWEST first vs. OLDEST
first
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Message-ID: <BC16DF1437384D7C88BE5DA5ECBF8155 at kfife2>
Content-Type: text/plain; charset="iso-8859-1"
Is there a way to make the asterisk voicemail app play back messages in
NEWEST FIRST order, instead of OLDEST FIRST? I see the situation
repeatedly where someone needs to dip into their voicemail archive to
get something from a recently saved voicemail message, and they have to
slog through lots of irrelevant stuff to get there.
I have seen this question come up previously on this list without an
answer. I'm hoping that someone can shed light on how to do it, or
confirm that it is NOT currently supported.
I've looked at the new 1.6 voicemail.conf and it doesn't seem have any
parameters that speak to that feature, nor an voicemailmain parameter in
1.4 or 1.6.
Can anyone confirm that this is not supported, or enlighten us on
how-to?
Many thanks.
-Karl
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20090520/00
8201e2/attachment-0001.htm
------------------------------
Message: 13
Date: Thu, 21 May 2009 00:42:46 +0000 (UTC)
From: Jeff LaCoursiere <jeff at jeff.net>
Subject: Re: [asterisk-users] Step-by-Step Asterisk and MeetMe Help
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <alpine.BSF.2.00.0905210041350.46773 at phoenix.jeff.net>
Content-Type: TEXT/PLAIN; charset=US-ASCII; format=flowed
So you were fourteen and a military engineer?
j
On Wed, 20 May 2009, ContactTel Business wrote:
> Many years in telecom and computer world is around 100 year in real
life..> 10 years ago i was a millionaire in the dot com boom and 24 years old
with a> P2 300 computer.., 20 years ago i was military engineer and running on
3.76> MHz 386's amber screens.. last year it was dual cores, today its
quad/opt> cores, and tomorrow morning it's going to be quantum physics/organic
> computers and VOIP will be of the past, since Voice over Something
else will> arrive.
>
>
>
> You can't put a system and let it go for 3-4 years unless you don't
have any> growth, ( new drives = new technology , IDE/SATA/ISCSI) new RAM/ NEW
CPU/> etc all these need software upgrades eventually..
>
>
>
> As far as my personal experience i reformat my desktops /fully, semi
> annually, and all servers get a facelift every other month ( new glib
for> new freeswitch updates, new ZAP hardware ? then you need new zaptel..
wait> zaptel aka dhadi needs X, X needs Y.. and so on..
>
>
>
> Mike
>
> ContacTel.COM
>
>
>
>
>
>
>
> From: asterisk-users-bounces at lists.digium.com
> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Jonathan
> Thurman
> Sent: May-20-09 7:33 PM
> To: asterisk-users at lists.digium.com
> Subject: Re: [asterisk-users] Step-by-Step Asterisk and MeetMe Help
>
>
>
> From the front page ( http://wiki.centos.org/FrontPage ):
>
> "What is CentOS?
> CentOS is an Enterprise Linux distribution based on the freely
available> <ftp://ftp.redhat.com/pub/redhat/linux/enterprise/> sources from Red
Hat> Enterprise Linux. Each CentOS version is supported for 7 years (by
means of> security updates). A new CentOS version is released every 2 years and
each> CentOS version is regularly updated (every 6 months) to support newer
> hardware. This results in a secure, low-maintenance, reliable,
predictable> and reproducible Linux environment."
>
> CentOS 4 ( http://wiki.centos.org/FAQ/CentOS4 ):
> "We intend to support CentOS-4 updates until Feb 29, 2012"
>
> CentOS 5 ( http://wiki.centos.org/FAQ/CentOS5 ):
> "We intend to support CentOS 5 until Mar 31st, 2014"
>
>
> So if you don't want major upgrades for a while you might want to go
with> the latest version. To put it into Microsoft terms... the minor
version is> like a service pack. So CentOS 4.7 is really a base lined version 4,
> service pack 7. You get the new features in major releases (like
there are> no more "smp" kernels in 5 to deal with)
>
> -Jonathan
>
>
>
> On Wed, May 20, 2009 at 2:36 PM, Jimmy Ezell <jezell at hmhca.com>
wrote:
>
>
>> On Wed, May 20, 2009 at 01:07:25PM -0700, Jimmy Ezell wrote:
>>
>>> multi-processor machine ( I had to remember to specify smp
>> for the kernel)
>>
>> I repeat: why bother with such an old system? Really?
>>
>> Recall the comment from the book. That book had nothing really
specific>> to Centos 4. Why do you shoot yourself in the foot by
>> installing Centos4
>> now?
>>
>> (not to mention Zaptel)
>>
>> --
>> Tzafrir Cohen
>
> Tzafrir thanks for the comments. I am not done playing with this and
in the> end I may well use newer software as you suggest.
>
> According to wikipedia CentOS 4.7 was released OCT. 2008 (7 months
ago) is> that really consider that old? I am looking to setup a phone system
that I> would hope would not require any major software upgrades for many
years.>
>
> Jimmy
>
>>
>
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
>
------------------------------
Message: 14
Date: Wed, 20 May 2009 21:02:21 -0400
From: "ContactTel Business" <lists at contacttel.com>
Subject: Re: [asterisk-users] Step-by-Step Asterisk and MeetMe Help
To: "'Asterisk Users Mailing List - Non-Commercial
Discussion'"
<asterisk-users at lists.digium.com>
Message-ID: <007101c9d9af$cc4f19d0$64ed4d70$@com>
Content-Type: text/plain; charset="us-ascii"
Hehe i meant 15 but i knew one would spot that..
I was 17 in fact, left at 22, yeah demolition, construction, sniper,
road
demolish, anti tank craters, and all the bells and whistles,
>>-----Original Message-----
>>From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-
>>bounces at lists.digium.com] On Behalf Of Jeff LaCoursiere
>>Sent: May-20-09 8:43 PM
>>To: Asterisk Users Mailing List - Non-Commercial Discussion
>>Subject: Re: [asterisk-users] Step-by-Step Asterisk and MeetMe Help
>>
>>
>>So you were fourteen and a military engineer?
>>
>>j
>>
>>On Wed, 20 May 2009, ContactTel Business wrote:
>>
>>> Many years in telecom and computer world is around 100 year in
real
>>life..
>>> 10 years ago i was a millionaire in the dot com boom and 24 years
old>>with a
>>> P2 300 computer.., 20 years ago i was military engineer and running
>>on 3.76
>>> MHz 386's amber screens.. last year it was dual cores, today
its
>>quad/opt
>>> cores, and tomorrow morning it's going to be quantum
physics/organic
>>> computers and VOIP will be of the past, since Voice over Something
>>else will
>>> arrive.
>>>
>>>
>>>
>>> You can't put a system and let it go for 3-4 years unless you
don't
>>have any
>>> growth, ( new drives = new technology , IDE/SATA/ISCSI) new RAM/
NEW
>>CPU/
>>> etc all these need software upgrades eventually..
>>>
>>>
>>>
>>> As far as my personal experience i reformat my desktops /fully,
semi
>>> annually, and all servers get a facelift every other month ( new
glib>>for
>>> new freeswitch updates, new ZAP hardware ? then you need new
zaptel..>>wait
>>> zaptel aka dhadi needs X, X needs Y.. and so on..
>>>
>>>
>>>
>>> Mike
>>>
>>> ContacTel.COM
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> From: asterisk-users-bounces at lists.digium.com
>>> [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of
>>Jonathan
>>> Thurman
>>> Sent: May-20-09 7:33 PM
>>> To: asterisk-users at lists.digium.com
>>> Subject: Re: [asterisk-users] Step-by-Step Asterisk and MeetMe Help
>>>
>>>
>>>
>>> From the front page ( http://wiki.centos.org/FrontPage ):
>>>
>>> "What is CentOS?
>>> CentOS is an Enterprise Linux distribution based on the freely
>>available
>>> <ftp://ftp.redhat.com/pub/redhat/linux/enterprise/> sources
from
Red>>Hat
>>> Enterprise Linux. Each CentOS version is supported for 7 years (by
>>means of
>>> security updates). A new CentOS version is released every 2 years
and>>each
>>> CentOS version is regularly updated (every 6 months) to support
newer>>> hardware. This results in a secure, low-maintenance, reliable,
>>predictable
>>> and reproducible Linux environment."
>>>
>>> CentOS 4 ( http://wiki.centos.org/FAQ/CentOS4 ):
>>> "We intend to support CentOS-4 updates until Feb 29,
2012"
>>>
>>> CentOS 5 ( http://wiki.centos.org/FAQ/CentOS5 ):
>>> "We intend to support CentOS 5 until Mar 31st, 2014"
>>>
>>>
>>> So if you don't want major upgrades for a while you might want
to go
>>with
>>> the latest version. To put it into Microsoft terms... the minor
>>version is
>>> like a service pack. So CentOS 4.7 is really a base lined version
4,>>> service pack 7. You get the new features in major releases (like
>>there are
>>> no more "smp" kernels in 5 to deal with)
>>>
>>> -Jonathan
>>>
>>>
>>>
>>> On Wed, May 20, 2009 at 2:36 PM, Jimmy Ezell <jezell at
hmhca.com>
>>wrote:
>>>
>>>
>>>> On Wed, May 20, 2009 at 01:07:25PM -0700, Jimmy Ezell wrote:
>>>>
>>>>> multi-processor machine ( I had to remember to specify smp
>>>> for the kernel)
>>>>
>>>> I repeat: why bother with such an old system? Really?
>>>>
>>>> Recall the comment from the book. That book had nothing really
>>specific
>>>> to Centos 4. Why do you shoot yourself in the foot by
>>>> installing Centos4
>>>> now?
>>>>
>>>> (not to mention Zaptel)
>>>>
>>>> --
>>>> Tzafrir Cohen
>>>
>>> Tzafrir thanks for the comments. I am not done playing with this
and>>in the
>>> end I may well use newer software as you suggest.
>>>
>>> According to wikipedia CentOS 4.7 was released OCT. 2008 (7 months
>>ago) is
>>> that really consider that old? I am looking to setup a phone
system
>>that I
>>> would hope would not require any major software upgrades for many
>>years.
>>>
>>>
>>> Jimmy
>>>
>>>>
>>>
>>> _______________________________________________
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com
-->>>
>>> asterisk-users mailing list
>>> To UNSUBSCRIBE or update options visit:
>>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
>>>
>>>
>>>
>>
>>_______________________________________________
>>-- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>>asterisk-users mailing list
>>To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-users
------------------------------
Message: 15
Date: Wed, 20 May 2009 21:52:18 -0400
From: "Barry L. Kline" <blkline at attglobal.net>
Subject: [asterisk-users] Bridging INBOUND PRI to OUTBOUND PRI fails
with Monitor()
To: asterisk-users at lists.digium.com
Message-ID: <4A14B3D2.5050609 at attglobal.net>
Content-Type: text/plain; charset=ISO-8859-1
I wrote a note earlier about this problem but have done quite a bit more
debugging. Now I'm stuck at what to do next.
I have inbound calls being answered by our Asterisk box, which then
dials our answering service and bridges those calls. The inbound and
outbound are both PRIs. The answering service takes our calls on a
PRI.
If I don't use the Monitor() application, things work find and have been
for a few thousand calls. If I add the Monitor() application, no audio
ever gets passed from the caller to the answering service.
I have noted the following things while testing with Monitor():
1) If I have it call my cell phone instead of the service, it works
fine.
2) If I have it call my home phone instead of the services, it works
fine.
3) I tried calling another number (in another state) that I know
terminates into a PRI and it worked fine.
4) If I call the service without Monitor(), it works fine.
Throw in Monitor() and it's virtually guaranteed not to work.
My dial plan and debug output for both the working and failing call is
at http://www.pastebin.ca/1429504 .
Things start to diverge around lines 28-31 and 68-72. Can anyone tell
me what I can do to further trace this problem? Thanks in advance for
anything you may be able to offer.
Barry
------------------------------
Message: 16
Date: Thu, 21 May 2009 07:06:02 +0400
From: "Manoj Panicker - FOES" <manoj.panicker at emirates.com>
Subject: [asterisk-users] PSTN Connection
To: <asterisk-users at lists.digium.com>
Message-ID:
<AC5F42F85475254AAB74B5A2B0653E4401F76400 at
DXBHQMBEX10.corp.emirates.com>
Content-Type: text/plain; charset="us-ascii"
Hi
Which is the best interface card to connect PSTN line with
Asterisk. Can somebody please help. My intention is to route the
incoming PSTN calls to internal IP Phones through Asterisk and Vice
versa. The Asterisk is in LAN and is reachable from all the IP phones in
the LAN.
Thanks
Manoj
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20090521/84
91ae86/attachment-0001.htm
------------------------------
Message: 17
Date: Thu, 21 May 2009 14:13:31 +1000
From: Alex Samad <alex at samad.com.au>
Subject: Re: [asterisk-users] Open source SIP client
To: asterisk-users at lists.digium.com
Message-ID: <20090521041330.GA30923 at samad.com.au>
Content-Type: text/plain; charset="us-ascii"
On Tue, May 19, 2009 at 10:38:24AM +1000, Paul Hales
wrote:>
> Not true. I am always wrong.
> (wait...is that a paradox?)
only on the 42nd time
>
> PaulH
>
>
[snip]
> ContactTel Business wrote:
-------------- next part --------------
A non-text attachment was scrubbed...
Name: not available
Type: application/pgp-signature
Size: 197 bytes
Desc: Digital signature
Url :
http://lists.digium.com/pipermail/asterisk-users/attachments/20090521/fc
197395/attachment-0001.pgp
------------------------------
Message: 18
Date: Thu, 21 May 2009 14:20:21 +1000
From: Paul Hales <pdhales at optusnet.com.au>
Subject: Re: [asterisk-users] PSTN Connection
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <4A14D685.2040305 at optusnet.com.au>
Content-Type: text/plain; charset=ISO-8859-1
Digium PSTN cards seem to work.
PaulH
Manoj Panicker - FOES wrote:>
> Hi
> Which is the best interface card to connect* PSTN* line with
> Asterisk. Can somebody please help. My intention is to route the
> incoming PSTN calls to internal IP Phones through Asterisk and Vice
> versa. The Asterisk is in LAN and is reachable from all the IP phones
> in the LAN.
>
> Thanks
> Manoj
>
>
------------------------------------------------------------------------>
> _______________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
------------------------------
Message: 19
Date: Thu, 21 May 2009 14:10:18 +0800
From: Rilawich Ango <maillisting at gmail.com>
Subject: [asterisk-users] interruption in queue
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID:
<6fbb529e0905202310u42d75f2boe5bec86b81bb44a8 at mail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1
HI,
I want to allow user to press 0 to the voicemail if the user don't
want to wait in the queue. Below is what I set but it doesn't work.
Anyone can help? ango
file: features.conf
[applicationmap]
opervm => 0,self/both,Macro,opervm
file: extensions.conf
...
exten => 5555,n(queue),Set(DYNAMIC_FEATURES=opervm)
exten => 5555,n,Queue(5555|tThH|||180)
...
[macro-opervm]
exten => s,1,NoOp(--openvm--)
exten => s,n,VoiceMail(3634 at default,u)
exten => s,n,Hangup
------------------------------
Message: 20
Date: Thu, 21 May 2009 09:01:39 +0100 (BST)
From: "--[ UxBoD ]--" <uxbod at splatnix.net>
Subject: Re: [asterisk-users] PSTN Connection
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID:
<30533477.221242892899791.JavaMail.root at office.splatnix.net>
Content-Type: text/plain; charset=utf-8
----- "Paul Hales" <pdhales at optusnet.com.au> wrote:
> Digium PSTN cards seem to work.
>
>
>
> PaulH
OpenVox works well.
Best Regards,
--
SplatNIX IT Services :: Innovation through collaboration
------------------------------
Message: 21
Date: Wed, 20 May 2009 22:04:21 -1000
From: Matt Darnell <mattdarnell at gmail.com>
Subject: [asterisk-users] Polycom Productivity Suite
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID:
<52b8ace90905210104u4239b2e2t3a539b34d75599a3 at mail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1
Has anyone been able to do the following:
1. Set the phone to automatically record all calls to the USB stick,
now you have to press three keys.
2. Put Record on the main screen when a call is active. This would
eliminate having to press the 'more' softkey.
Thanks,
Matt
------------------------------
Message: 22
Date: Thu, 21 May 2009 13:58:10 +0530
From: Arun Kumar <arunvoip at gmail.com>
Subject: [asterisk-users] Fwd: Asterisk CCM, CME Integration
To: "ccie_voice at onlinestudylist.com" <ccie_voice at
onlinestudylist.com>,
Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>, Commercial and
Business-Oriented
Asterisk Discussion <asterisk-biz at lists.digium.com>
Message-ID:
<a70a109b0905210128u32b3a41cude2dc79189178dbd at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
Hi All,
please provide some help.
I'm just posting this questions to both forums as its related to both.
In
hope to get some help on below issue:
Asterisk 1.4.x
CCM = 4.x
CME = 4.x
codec = g711ulaw
Here is my setup:
600X Phones ----> Asterisk ---- SIP Trunk ----> Call Manager -----> CME
-----> 461X Phones
461X Phones ----> CME -----> my dial peer points to Asterisk IP for 600X
Phones
so in the above setup I'm able to call from Asterisk to my CME and
vice-versa.
here is my problem: when I call from 6004 to my cme extension 4615, on
4615
I've configured noans timeout to 15 and then it goes to my unity express
(cue) for voicemail so when I call my cme extension it rings for few
seconds
and then on my asterisk cli I see "500 Internal Server Error" back
from
my
CCM IP and getting standard asterisk message saying "all circuits are
busy
now" . as per my understanding it should go to my cue.
please advise and let me know if you need any other details.
Regards
Arun
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
http://lists.digium.com/pipermail/asterisk-users/attachments/20090521/12
12870a/attachment.htm
------------------------------
_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
End of asterisk-users Digest, Vol 58, Issue 56
**********************************************
