Hello Community,
I have installed Dahdi on Centos on many system and succesfully used that..
But today i have bad luck...
This is the error that i am facing
[r...@localhost dahdi-linux-complete-2.2.1+2.2.1]# make all
make -C linux all
make[1]: Entering directory
`/usr/src/dahdi-linux-complete-2.
Can you post outputs for the following commands;
#asterisk -rx 'pri show spans'
#asterisk -rx 'zap show channels'
#wanpipemon -i w1g1 -c Ta
Sam
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jaap Winius
Sent
--
Jeffery
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Hello,
i´ve got this when i asterisk has died / killed and was restarted but i
dont have seen that it will collapse then.
i also got this after restarting asterisk from the CLI with restart now.
so dont worry ;)
best regards
steve smith
Danny Nicholas schrieb:
>
> You’d think that this is/wa
Danny,
I haven't been able to test further, but I've seen the same issue since
I upgraded to 1.6.2.6. The astcanary does not appear to stay up more
than a few seconds when asterisk is initially started. On this same
system using 1.6.2.2 (the previous version I was running prior to the
upgrad
If * answers the call, it will be on the "loop" but with canreinvite or
directrtp the media can be out of * and redirected to the final end point even
if signaling goes through *.
For the trunk, you can have multiple simultaneous calls. I do not know about
Mitel's licensing but with only one tr
I finally got it at "Calories Consumed". Geesh. Good one! :)
j
On Thu, 1 Apr 2010, Olle E. Johansson wrote:
> FOR IMMEDIATE RELEASE
> Puerto Escondido, Mexico, April 1st, 2010:
>
> Digium launches Asterisk VCC (TM) - a new virtual communication platform
> for enterprises, the public sector a
On 4/1/2010 5:09 PM, Juan E. Rodríguez wrote:
> Depends on the configuration you make. For example, if you want to route the
> call giving the Mitel a new desrination or prefix, you can use Transfer
> dialplan app. Transfer before answering the call will be redirected with SIP
> 302.
>
> If the
> This begs the question of when the actual violation occurs. In other
> words, is this really a "usage" issue, or does the violation occur
> at install time even though the non-GPL component is not "usable"?
It's hard to see how the violation could occur unless and until the
resulting program w
Depends on the configuration you make. For example, if you want to route the
call giving the Mitel a new desrination or prefix, you can use Transfer
dialplan app. Transfer before answering the call will be redirected with SIP
302.
If the call is to be anwered on *, then canreinvite set to yes o
On 4/1/2010 1:52 PM, Kevin P. Fleming wrote:
> Chris Miller wrote:
>
>> A comment in the spec file would have been nice... Does anyone know
>> if this a real technical issue, or simply a licensing conflict
>> between GPL and Digium?
>
> It is not a technical issue; it is an issue because some of th
Hi All,
I have a question about how a particular situation would work between two
PBX systems:
If I were to connect my Mitel box to my Asterisk box via a SIP trunk (same
rack, same network), and then pass a call from the Mitel to Asterisk to
perform some functions (lookups, maybe routing), and th
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi, Jim.
On Sun, 28 Mar 2010, Jim Dickenson wrote:
> I think if you are installing dahdi complete from source you do
> "make all" and "make install" and "make config"
Something that I forgot to ask previously is if the update of Asterisk
or DAHDI is
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi, Alyed.
On Sun, 28 Mar 2010, Alyed wrote:
>> I didn't know that there was Digium's GUI. It is FLOSS? I was looking
>> for in the site of Digium in the download section, but the unique
>> thing that I saw that it speaks of a GUI is AsteriskNow, tha
Youd think that this is/was some kind of April fool message, but it is a
real 1.6 warning
http://lists.digium.com/pipermail/asterisk-commits/2008-May/022745.html
Since 1.6 has more multi-thread capabilities, the good folks at
Digium/Asterisk made this warning program to keep runaway threads f
Hint Apr 1
~
Andrew "lathama" Latham
lath...@gmail.com
* Learn more about OSS http://en.wikipedia.org/wiki/Open-source_software
* Learn more about Linux http://en.wikipedia.org/wiki/Linux
* Learn more about Tux http://en.wikipedia.org/wiki/Tux
On Thu, Apr 1, 2010 at 6:05 PM, Marcus Vinic
People,
Anybody knows what mean this message in my CLI:
[Apr 1 16:58:34] WARNING[3845]: asterisk.c:3050 canary_thread: The canary is
no more. He has ceased to be! He's expired and gone to meet his maker! He's
a stiff! Bereft of life, he rests in peace. His metabolic processes are now
hi
I was missing libtool-ltdl-devel. Thanks!
On Thu, Apr 1, 2010 at 2:48 PM, Jared Smith wrote:
> - "Nathan Pryor" wrote:
>
> "make menuconfig" does not show cdr_odbc as a selectable compile option. I
> have compiled and installed both unixODBC and freetds from source and have
> verified both
I'm doing some automated calling by putting .call files in the Outgoing
folder of Asterisk. I'm concerned this might be a stupid question, but I'm
pretty sure I've done my research well and I'm unable to come up with an
answer on my own.
I want to know: what happens to the .call files after the "M
Chris Miller wrote:
> A comment in the spec file would have been nice... Does anyone know
> if this a real technical issue, or simply a licensing conflict
> between GPL and Digium?
It is not a technical issue; it is an issue because some of the modules
in -addons have licenses that are pure GPL
- "Nathan Pryor" wrote:
"make menuconfig" does not show cdr_odbc as a selectable compile option. I have
compiled and installed both unixODBC and freetds from source and have verified
both successfully connect to my sql server. Both were installed to standard
locations (/usr/lib). I had n
Yes, after installation of the drivers I've tried:
./configure
./configure --with-unixodbc=/usr
./configure --with-unixodbc=/usr/lib
None turn on cdr_odbc. I'm at a loss here. Is there anything in the
configure output I should be looking for?
On Thu, Apr 1, 2010 at 2:10 PM, Tilghman Lesher wro
It seems that asterisk-addons and one or more of Digium's licensed
modules such as res_fax_digium have a conflict that doesn't seem to
be documented anywhere I can find.
In a nutshell, asterisk14-addons-core has a fake provide for
asterisk-gplonly :
#
# core subpackage
#
%package core
Summar
On Thursday 01 April 2010 13:36:00 Nathan Pryor wrote:
> "make menuconfig" does not show cdr_odbc as a selectable compile option. I
> have compiled and installed both unixODBC and freetds from source and have
> verified both successfully connect to my sql server. Both were installed to
> standard l
"make menuconfig" does not show cdr_odbc as a selectable compile option. I
have compiled and installed both unixODBC and freetds from source and have
verified both successfully connect to my sql server. Both were installed to
standard locations (/usr/lib). I had no problem compiling cdr_odbc on my
Back to your original point, this is NOT the day to post a real news item;
we all know that's Friday after 4:30 EST.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen
Sent: Thursday, April 01, 2010 1
On Thu, Apr 01, 2010 at 07:05:07PM +0100, --[ UxBoD ]-- wrote:
> - "Tzafrir Cohen" wrote:
>
> > On Thu, Apr 01, 2010 at 02:41:15PM +0200, Tommy Botten Jensen wrote:
> > > -BEGIN PGP SIGNED MESSAGE-
> > > Hash: SHA512
> > >
> > > You know there are 1st of april jokes, and there are ev
- "Tzafrir Cohen" wrote:
> On Thu, Apr 01, 2010 at 02:41:15PM +0200, Tommy Botten Jensen wrote:
> > -BEGIN PGP SIGNED MESSAGE-
> > Hash: SHA512
> >
> > You know there are 1st of april jokes, and there are evil 1st of
> april
> > jokes.
>
> Huh? Have you actually bothered reading the
On Thursday 01 April 2010 10:36:23 Vitali Fomine wrote:
> > Thought it was funny myself! Not as funny as the other one just posted
> > about the VoIP clients written in C# and .net under Microsoft though. Now
> > that was funny!
>
> It was not a joke. What is wrong with c#,.net?
You probably shoul
Hi,
I'm seeing a lot of "Exceptionally long voice queue length" errors in
my logs, and then I seem to have a problem
where Asterisk will drop the registration for a significant number of
phones (they go UNREACHABLE), but then they
come back approximately a minute later.
Is this some sort of load pr
Hi,
thank you very much. this solved my problem.
greets
felix
Am 31.03.2010 um 22:52 schrieb Mark Michelson:
> Felix Tiefenthaler wrote:
>> Hi list,
>>
>> can anyone tell me how to reset/delete all modifications (personal
>> greeting message, personal name, ...) I made in my voicemail?
>> I j
Figured out my issue. My contacts are in -directory.cfg when it
should be in -directory.xml.
When did Polycom switched from CFG to XML?
From: hin lee
To: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Fri, March 19, 2010 12:00:14 PM
Subje
On Thu, Apr 01, 2010 at 02:41:15PM +0200, Tommy Botten Jensen wrote:
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA512
>
> You know there are 1st of april jokes, and there are evil 1st of april
> jokes.
Huh? Have you actually bothered reading the site it points to?
The date there clearly state
I found a sap400, wich is FXO and seems to work fine with asterisk. It
offers the posibility to plug 4 analog phonelines.
Thanks again.
Kosa
- Un mundo mejor es posible -
Ioan Indreias escribió:
> Both SPA2102 and SPA9000 have FXS ports. You need to use SPA3102 (or
> other ATA which have FXO po
Ondrej Valousek wrote:
> Hello,
>
> Did anyone manage to force asterisk to put Remote-party-ID attribute on
> the SIP outgoing call? I.e. When A calls B, I want that A gets a name of
> B displayed on his phone.
> Note that name of A gets displayed on the B's phone fine, but this is
> not what I
> Thought it was funny myself! Not as funny as the other one just posted
> about the VoIP clients written in C# and .net under Microsoft though. Now
> that was funny!
It was not a joke. What is wrong with c#,.net?
Best regards,
Vitali Fomine
--
___
Try using sendrpid=yes on sip.conf
Regards,
Juan
Ondrej Valousek wrote:
> Hello,
>
> Did anyone manage to force asterisk to put Remote-party-ID attribute on
> the SIP outgoing call? I.e. When A calls B, I want that A gets a name of
> B displayed on his phone.
> Note that name of A gets displaye
Hello,
Did anyone manage to force asterisk to put Remote-party-ID attribute on
the SIP outgoing call? I.e. When A calls B, I want that A gets a name of
B displayed on his phone.
Note that name of A gets displayed on the B's phone fine, but this is
not what I want.
This works with Cisco Call man
On Thu, 1 Apr 2010, Tommy Botten Jensen wrote:
> You know there are 1st of april jokes, and there are evil 1st of april
> jokes.
> ... I actually felt a bit nauseous
Thought it was funny myself! Not as funny as the other one just posted
about the VoIP clients written in C# and .net under Micros
Hi all,
My problem boils down to these errors:
... Unable to create channel of type 'ZAP' (cause 34 -
Circuit/channel congestion)
== Everyone is busy/congested at this time
This is triggered by lines in extentions.conf such as:
exten => _X.,1,Dial(ZAP/g1/${EXTEN},,W)
The sy
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA512
You know there are 1st of april jokes, and there are evil 1st of april
jokes.
... I actually felt a bit nauseous
- - Tommy
Den 01. april 2010 07:53, skrev Olle E. Johansson:
> FOR IMMEDIATE RELEASE
> Puerto Escondido, Mexico, April 1st, 2010:
>
Redfone uses and improved, in house developed TDMoE driver, officially
supported by same Redfone.
Redfone´s support site maintains tdmoe driver updated and "certified" to
operate in every zaptel and dahdi versions.
Txs
Jorge Churio
Redfone Communications
--
_
I'm not quite sure what do you mean with MSC.
Anyway, I assume your environment is like
[PSTN (Public Switched Telephone Network)]<-->[DTM
Switch]<---SS7 (PRI line)>[Asterisk
Box] IP net
If you mean MSC Mo
Press Release
For Immediate Release
OfficeSIP Communications
http://www.officesip.com/
i...@officesip.com
OfficeSIP Communications Makes Its VoIP SIP Products Open Source
OfficeSIP Communications makes its two enterprise VoIP SIP clients
officially open-source. OfficeSIP Softphone and OfficeSIP
Both SPA2102 and SPA9000 have FXS ports. You need to use SPA3102 (or
other ATA which have FXO ports).
HTH,
Ioan.
On Thu, Apr 1, 2010 at 12:29 AM, Kosa wrote:
> I have two linksys spa2102 and a sap9000 but as far as I know I need
> something else to connect the asterisk box to the analog phoneli
We were hoping voicemail would become Tweets and that Tweets from your
bathroom scale could be sent as audio using calls files. I guess that
will be the next minor version?
/r
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-- Bandwidth and Colocation Provided by http://
Just can't wait for the live calorie counter! :)
l.
2010/4/1 Olle E. Johansson
> FOR IMMEDIATE RELEASE
> Puerto Escondido, Mexico, April 1st, 2010:
>
> Digium launches Asterisk VCC (TM) - a new virtual communication platform
> for enterprises, the public sector and the home.
> =
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