Hi!
I am running asterisk ver 1.2.4 and have faced this error:
Try a downgrade to Asterisk 0.7.1 ;-
Philipp
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New to Asterisk? Join us for a live
hi,
any response ?
thanks,
Ashik
On Mon, Sep 6, 2010 at 12:01 PM, Ashik Ali beaasteriskg...@gmail.comwrote:
Hi all,
I am able to understand your solutions. Depending upon the india number
reading method, I changed number reading setting in say.conf language. For
more details visit my
I am not getting anything in debug because call is not reaching us from
other end, it is inbound connection over ipsec.
Thanks.
From: Deepika Nijhawan [mailto:deepika.nijha...@oxygen8.com]
Sent: 08 September 2010 17:10
To: 'asterisk-users@lists.digium.com'
Subject: IPSec on
On Wed, Sep 8, 2010 at 4:24 PM, Adrià Vidal adriavi...@gmail.com wrote:
Sorry was my fault , res_calendar was ok, but ical and caldav need other
libs (neon,ical...) that were
not installed in my system.
--
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Adrià Vidal
--
_
I don't know exactly what help you expect to receive in this forum.
Asterisk itself has nothing to do with VPNs of any kind, and you should
take your questions regarding the setup and configuration of them to the
appropriate place.
On 09/09/10 18:26, Deepika Nijhawan wrote:
I am not
Hello list,
getting warning : *syntax error, unexpected 'token'*
dialplan :
exten = pbx,n,Macro(CheckNetworkProblems,${custID})
exten = pbx,n,NoOp(status = ${STATUS})
exten = pbx,n,GoToIf($[${STATUS}=congestion]?backup:nocongestion)
CLI :
[Sep 9 12:27:07] -- Executing [...@cust:15]
Jonas Kellens wrote:
Hello list,
getting warning : *syntax error, unexpected 'token'*
dialplan :
exten = pbx,n,Macro(CheckNetworkProblems,${custID})
exten = pbx,n,NoOp(status = ${STATUS})
exten = pbx,n,GoToIf($[${STATUS}=congestion]?backup:nocongestion)
CLI :
[Sep 9
Hello Asterisk community,
I'm experiencing some problems with a Digium TE4XXP, the thing is that i'm
sharing IRQ with some megasas device:
169: 69917985 0 0 0 0 0
0 0 IO-APIC-level megasas, wct4xxp
I've been searching here:
On Thu, Sep 9, 2010 at 4:26 AM, Deepika Nijhawan
deepika.nijha...@oxygen8.com wrote:
I am not getting anything in debug because call is not reaching us from
other end, it is inbound connection over ipsec.
Asterisk != IPSec
Like Rob said, repost your question once you have IPSec working.
--
Hi,
I have created one SIP extension in Asterisk and configured that extension
in SPA 3102. And connected one FAX machine to the SPA3102 and one to
Asterisk.
The problem is if I try to send FAX from SPA3102 to Asterisk i am not able
to send. But if the same if I try to send from Asterisk to
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Dias
Sent: Thursday, September 09, 2010 6:39 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] How to avoid interruptions with DIGIUM
Hi,
Please excuse me for addressing this Linux issue on this list, however I
hope that some of you have found a solution thats matches the * use and
also easy to install without very deep knowledge of Linux.
My wish are a program that maintain a mirror copy of the HD.
Thank you!
Best regards
modprobe blacklisting may be of help...
~
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, Sep 9, 2010
HB wrote:
Please excuse me for addressing this Linux issue on this list, however I
hope that some of you have found a solution thats matches the * use and
also easy to install without very deep knowledge of Linux.
My wish are a program that maintain a mirror copy of the HD.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ashik Ali
Sent: Thursday, September 09, 2010 2:06 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] agi playback to execute say.conf
Hello list,
is it possible to set a variable (channel variable) from within another
channel ?!
I'm currently working with 2 channels that I bridge afterwards. It would
be good to set a variable in one channel when something occurs in the
other channel.
If some variable is not set in
Can you clarify 'send FAX from SPA3102 to Asterisk' ? Are you running Fax for
Asterisk software? Where do you expect your call to terminate?
As I understand it, without FFA or some of the patches floating around,
Asterisk is not T.38 gateway aware. Without more information, it's hard to
- Andrew Latham lath...@gmail.com wrote:
modprobe blacklisting may be of help...
The module sharing interrupts with the card is his storage controller
(megasas). Blacklisting the storage controller module? That is not a good
idea...
Try putting the card in another slot.
--Tim
--
Hello list,
how come on my Asterisk 1.6.2.11, I have no help available ?!
asterisk*CLI core show application Dial
-= Info about application 'Dial' =-
[Synopsis]
Not available
[Description]
Not available
[Syntax]
Not available
[Arguments]
Not available
[See Also]
Not available
Kind
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Thursday, September 09, 2010 8:56 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Set channel variable from within
On 09/09/2010 09:01 AM, Tim Nelson wrote:
- Andrew Latham lath...@gmail.com wrote:
modprobe blacklisting may be of help...
The module sharing interrupts with the card is his storage controller
(megasas). Blacklisting the storage controller module? That is not a good
idea...
No, but
Hi Tim,
Thanks for your reply. I am not using any fax software. I just created two
extensions in trixbox (Asterisk 1.4), one extension I configured in one
SPA3102 here FAX machine is connected in the phone port of SPA3102.
As the same the other extension is configured in another SPA3102 and FAX
On 09/09/2010 04:12 PM, Danny Nicholas wrote:
*From:* asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas
Kellens
*Sent:* Thursday, September 09, 2010 8:56 AM
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Thursday, September 09, 2010 9:45 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Set channel variable from within
Thursday, September 9, 2010, 4:32:29 PM, Gopalakrishnan wrote:
I am sending FAX from one extension to another extension. I am not able to
send.
Preferred Codec:G711u
You forget to mentoin where do you live? In some countries the G711a
codec and in onther countries the G711u codec useable.
Thanks Kevin,
But today i saw a Kernel Panic into my server, for no any apparent
reasondoes
this parameter could help: pci=routeirq
By the way, we are using DELL servers, i've also used Sangoma, and always
the same problem
Thanks!
2010/9/9 Kevin P. Fleming kpflem...@digium.com
On
Hi All, sorry for the off topic.
I own 3 vegastream 50 BRI-s 4 ISDN ports gateways with firmware version 6
and I need some advanced parameters available only from firmware version 7.
I am sure that I need those parameters because changing the vega gateway with a
20$ cologne pci card in an
I am from India and I hope I have to use G711u...If I am not wrong
On Thu, Sep 9, 2010 at 8:36 PM, Gergo Csibra csi...@gmail.com wrote:
Thursday, September 9, 2010, 4:32:29 PM, Gopalakrishnan wrote:
I am sending FAX from one extension to another extension. I am not able
to
send.
The issue we are having is that in-call RFC2833 DTMF digits are being
dropped with Broadvox and Level 3. This is happening with Grandstream GXP
and Snom phones. We did some testing with the vendors and here is one of
the responses we got back. Is there any way to force asterisk to modify the
On Thu, Sep 9, 2010 at 10:04 AM, Jonas Kellens jonas.kell...@telenet.be wrote:
asterisk*CLI core show application Dial
did you have libxml-doc installed when you build asterisk?
*CLI module load app_dial.so
--
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com |
On Thu, Sep 9, 2010 at 11:37 AM, Paul Belanger
paul.belan...@polybeacon.com wrote:
did you have libxml-doc installed when you build asterisk?
s/-doc/-dev
--
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
blog.polybeacon.com
--
Does anyone have a packet capture of a 3Com 3102 phone registering with
an NBX that I could take a look at ?
What is the expected traffic flow, all I get is the 0x8836 initial
packet from the phone but have no NBX to validate
Please still seeking help
No one using 3com with asterisk ??
On 09/09/2010 05:37 PM, Paul Belanger wrote:
On Thu, Sep 9, 2010 at 10:04 AM, Jonas Kellensjonas.kell...@telenet.be
wrote:
asterisk*CLI core show application Dial
did you have libxml-doc installed when you build asterisk?
*CLI module load app_dial.so
Indeed I did not
I hadn't heard about them until your mailing list post.
On Thu, Sep 9, 2010 at 8:48 AM, Barry Fawthrop ba...@isscp.com wrote:
Does anyone have a packet capture of a 3Com 3102 phone registering with
an NBX that I could take a look at ?
What is the expected traffic flow, all I get is the
On Thu, Sep 9, 2010 at 11:52 AM, Jonas Kellens jonas.kell...@telenet.be wrote:
Can I install libxml-doc now without having to rebuild asterisk ?!
No, install libxml-dev then rerun ./configure, make install
--
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com |
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul Belanger
Subject: Re: [asterisk-users] info about application not available
asterisk1.6.2.11
On Thu, Sep 9, 2010 at 11:52 AM, Jonas Kellens
Hi Everyone,
My experience is only with the Canadian providers. What options/providers
are there in Dallas and Philadelphia other than Verizon when it comes to
internet? Something in the order of at least 10mbps down and up - I
understand that and higher bandwidths are easily available in USA due
Steve Davies wrote:
Hi,
I am using 1.6.2.11, and I need to be able to include the name of the
channel that answered a call in the call-recording filename.
At a guess we need to use the Queue(name,,macro) or
Dial(chan1chan2,,M(macro)) and use the macro to update the call
recording
bilal ghayyad wrote:
Hi All;
I would like to use Asterisk for a call center, but really does not know if
Asterisk support the following in a good way:
1) Ability to do an inteligent routing, so to route the call to the proper
skill group based on the caller information?
2) If I can
On 9 September 2010 17:52, Antonio Berrios
anto...@sheffieldcitytaxis.com wrote:
Steve Davies wrote:
Hi,
I am using 1.6.2.11, and I need to be able to include the name of the
channel that answered a call in the call-recording filename.
At a guess we need to use the Queue(name,,macro) or
Is there an archive of security advisories for Asterisk? We recently
upgraded a customer from 1.2 to 1.4 and now they are asking for
documentation of all security and bug related fixes. I know the
advisories get published on this list but is there an easier way to find
them than trying
On Thu, Sep 9, 2010 at 10:25 AM, Carlos Chavez cur...@telecomabmex.comwrote:
Is there an archive of security advisories for Asterisk? We
recently
upgraded a customer from 1.2 to 1.4 and now they are asking for
documentation of all security and bug related fixes. I know the
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Chavez
Subject: [asterisk-users] Archive of security advisories?
Is there an archive of security advisories for Asterisk? We
recently
upgraded a
On Thu, Sep 09, 2010 at 12:25:03PM -0500, Carlos Chavez wrote:
Is there an archive of security advisories for Asterisk? We recently
upgraded a customer from 1.2 to 1.4 and now they are asking for
documentation of all security and bug related fixes. I know the
advisories get published
As far as Dallas, it completely depends on where you are. The only provider
that blankets an area with fiber is Verizon and that is really only 2-3
cities around Dallas and it is usually residential, not business. They
aren't in Dallas itself. Time Warner and Cogent have a lot of coverage in
On Thu, 2010-09-09 at 15:29 +0200, hbk wrote:
My wish are a program that maintain a mirror copy of the HD.
http://www.drbd.org/
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New to Asterisk?
Have a Cisco 7975g running SIP firmware version 8.3.4. Many things are broken
with Asterisk.
1) BLF doesn't work
2) MWI doesn't work
3) Sometimes the calls get stuck on the display
4) Sometimes MOH works
5) Headset jack doesn't work
Can anyone recommend a version of the SIP firmware for the
On Thursday 09 September 2010 12:46:10 Kyle Kienapfel wrote:
On Thu, Sep 9, 2010 at 10:25 AM, Carlos Chavez
cur...@telecomabmex.comwrote:
Is there an archive of security advisories for Asterisk? We
recently
upgraded a customer from 1.2 to 1.4 and now they are asking for
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Answer
Can someone clarify what early media is? I noticed that NOT answering
a call before dumping them into a queue that has music on hold will not
set up a leg to push music back over the calling SIP channel. Tossing
an Answer command
Greetings all-
During some recent testing and debugging, I wanted to use the 'fxstest'
application. However, I found it hasn't been built when doing the standard
'make, make install' shtick with dahdi-linux-complete-2.3.0.1+2.3.0...
Can anyone tell me how to build fxstest?
Thanks!
--Tim
--
On Thu, Sep 9, 2010 at 4:38 PM, Hose hose+aster...@bluemaggottowel.com wrote:
Can someone clarify what early media is?
Basically playing audio to the channel before actually answering the
channel (IE: Answer()). You usually use Progress() at the start of
your dial plan to send 183 Session
On Thu, Sep 9, 2010 at 4:40 PM, Tim Nelson tnel...@rockbochs.com wrote:
Can anyone tell me how to build fxstest?
No, but if you output the error message we can help point you in the
right direction.
--
Paul Belanger | dCAP
Polybeacon | Consultant
Jabber: paul.belan...@polybeacon.com | IRC:
On 9/9/10 1:40 PM, Tim Nelson wrote:
During some recent testing and debugging, I wanted to use the 'fxstest'
application. However, I found it hasn't been built when doing the standard
'make, make install' shtick with dahdi-linux-complete-2.3.0.1+2.3.0...
Can anyone tell me how to build
On Thu, Sep 9, 2010 at 11:08 AM, Danny Dias ing.diasda...@gmail.com wrote:
Thanks Kevin,
But today i saw a Kernel Panic into my server, for no any apparent
reasondoes
this parameter could help: pci=routeirq
By the way, we are using DELL servers, i've also used Sangoma, and always
the
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