Hi Guys,
Hope fully somebody out there will have experienced this and can shed some
light on how it was overcome.
Current setup includes asterisk 1.6.2.11, GNU GK and a Quintum Tenor CMS on
the same lan. Earlier I was unable to make a sip call from the CMS back to a
sip client registered on my ast
On Thu, Sep 9, 2010 at 11:08 AM, Danny Dias 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 same problem
>
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 buil
On Thu, Sep 9, 2010 at 4:40 PM, Tim Nelson 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: pabelanger (Freenode)
b
On Thu, Sep 9, 2010 at 4:38 PM, Hose 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 Progress SIP Message.
Usually used
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
--
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
On Thursday 09 September 2010 12:46:10 Kyle Kienapfel wrote:
> On Thu, Sep 9, 2010 at 10:25 AM, 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 sec
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 C
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/
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk
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
big
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 publishe
-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 cu
On Thu, Sep 9, 2010 at 10:25 AM, 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 on
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 to
On 9 September 2010 17:52, Antonio Berrios
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
>> Dial(chan1&cha
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
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(chan1&chan2,,M(macro)) and use the macro to update the call
> recordi
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
-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
wrote:
> Can I in
On Thu, Sep 9, 2010 at 11:52 AM, Jonas Kellens 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 | IRC: pabelanger (Freenod
I hadn't heard about them until your mailing list post.
On Thu, Sep 9, 2010 at 8:48 AM, Barry Fawthrop 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 0x8836 initial
On 09/09/2010 05:37 PM, Paul Belanger wrote:
> On Thu, Sep 9, 2010 at 10:04 AM, Jonas Kellens
> 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 had libxml-
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 ??
Than
On Thu, Sep 9, 2010 at 11:37 AM, Paul Belanger
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
--
On Thu, Sep 9, 2010 at 10:04 AM, Jonas Kellens 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 | IRC: pabelanger (Fre
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
DT
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 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.
>
> >>> > Prefe
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 Asteri
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
> On 09/09/2010 09:01 AM, Tim
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 useab
_
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
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
*To
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
m
On 09/09/2010 09:01 AM, Tim Nelson wrote:
> - "Andrew Latham" 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 he may
_
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 othe
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 re
- "Andrew Latham" 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
--
__
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
pinpo
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 channe
_
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 sett
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.
http://e
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
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
HB
_
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,
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 SPA31
On Thu, Sep 9, 2010 at 4:26 AM, Deepika Nijhawan
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.
--
Paul Belanger | dCAP
Polybe
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: http://ubu
Jonas Kellens wrote:
> Hello list,
>
> getting warning : *syntax error, unexpected ''*
>
>
> dialplan :
>
> exten => pbx,n,Macro(CheckNetworkProblems,${custID})
> exten => pbx,n,NoOp(status = ${STATUS})
> exten => pbx,n,GoToIf($["${STATUS}"="congestion"]?backup:nocongestion)
>
>
> CLI :
>
>
Hello list,
getting warning : *syntax error, unexpected ''*
dialplan :
exten => pbx,n,Macro(CheckNetworkProblems,${custID})
exten => pbx,n,NoOp(status = ${STATUS})
exten => pbx,n,GoToIf($["${STATUS}"="congestion"]?backup:nocongestion)
CLI :
[Sep 9 12:27:07] -- Executing [...@cust:15]
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 getti
On Wed, Sep 8, 2010 at 4:24 PM, Adrià Vidal 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.
--
--
Adrià Vidal
--
_
-- Bandwidth and Colo
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 asterisk
hi,
any response ?
thanks,
Ashik
On Mon, Sep 6, 2010 at 12:01 PM, Ashik Ali wrote:
> 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 blog http://asterisknumb
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