Telco always says it is not their issue.
This is all over google, did you even check? Did you check your options in
chan_dahdi.conf?
hanguponpolarityswitch=yes
I am not sure if that is your problem but it would be helpful to list the
things you have found, tested, and ruled out.
As for prepend
Hi Brian,
Did you find a solution to your problem? or at least got a working dial-plan
for it? I have the same problem again as well and want to know what to do
with the dial-plan to off-set the effect at least since Telco says it's not
their issue.
Regards,
Bruce
On Thu, Apr 7, 2011 at 5:53 PM,
On a second thought, I don't need the predetermined delay. I can probably
just set that with additional w's in the DialBackground command (which I
made up).
So rather something like:
_X.,1,Progress
_X.,2,DialBackground(SIP/123456@provider,,D(ww${EwwXwwTwwEwwN}ww))
_X.,3,ConnectLegs
Thank
Hi,
has the following been done before respectively is it possible with
Asterisk? I searched the archives but couldn't locate anything.
1. Call to comes in via SIP.
2. Call is not answered yet but progress continues.
3. At the moment the call comes in something like this gets spawned in the
Dear, when I dial *30 in order to get instructions to blacklist an
extension, Idon't get the menu but I get a new dial tone.
What happen please ??? What can I do to solve this ???
Thanks a lot,
Alejandro
--
_
-- Bandwidth and C
- Original Message -
> On Thu, 2011-05-05 at 14:13 +, satish patel wrote:
> > Hi All,
> >
> > Just wondering is it safe to use asterisk 1.8 latest branch on
> > production ?
> >
> > http://svn.asterisk.org/svn/asterisk/branches/1.8/ Revision
> > 317100
> >
> > -S
> We've been run
At 05:39 AM 5/6/2011, you wrote:
Thanks for the feedback, Ira. It makes me very sad to hear what you
say and I hope that we can get more resources from the community to
assist in the process to make it more friendly. We want to get those
bug reports. The one thing I hate to hear when I'm trave
On 05/06/2011 01:30 PM, Bob Beers wrote:
Not sure if this will work, but I'd try adding, before line 86:
#Workaround for PAE
%if "%{paevar}" == "PAE"
Provides: kmod-dahdi-linux
%endif
Can't actually test it myself, sorry.
- Bob
You'd probably want to modify the kmodtool that comes with it,
>
The contents of this e-mail are intended for the named addressee only. It
contains information that may be confidential. Unless you are the named
addressee or an authorized designee, you may not copy or use it, or disclose it
to anyone else. If you received it in error please notify us immed
On Fri, May 6, 2011 at 2:48 PM, Vahan Yerkanian wrote:
> Has anyone used this board as an Asterisk server?
> http://www.supermicro.com/products/motherboard/ATOM/ICH9/X7SPE.cfm?typ=H&IPMI=Y
>
> I'm mostly interested about the possible compatibility issues this board may
> have with the AEX800 card.
Has anyone used this board as an Asterisk server?
http://www.supermicro.com/products/motherboard/ATOM/ICH9/X7SPE.cfm?typ=H&IPMI=Y
I'm mostly interested about the possible compatibility issues this board
may have with the AEX800 card.
--
___
On Fri, May 6, 2011 at 2:14 PM, Jerry Geis wrote:
> Is there a way in asterisk to Activate/Clear the blinking light on polycom
> phones
> indicating VM. Either from an AGI or some way in the dialplan?
>
> I want to be able to control this light for from my application.
> I have an AGI to do somet
On Fri, May 6, 2011 at 2:12 PM, Bob Beers wrote:
> On Fri, May 6, 2011 at 1:27 PM, Bob Beers wrote:
>> Hi Steven,
>>
>> Can you put the .spec file from dahdi-linux-kmod package up?
>
> Nevermind, I got it. :-)
>
> Looking at it now.
Not sure if this will work, but I'd try adding, before line 86
It was my problem ;)
https://issues.asterisk.org/view.php?id=18951
fixed in svn
On 6 May 2011 16:45, Steve Davies wrote:
> On 6 May 2011 16:30, Eric Wieling wrote:
>>> -Original Message-
>>> From: asterisk-users-boun...@lists.digium.com
>>> [mailto:asterisk-users-boun...@lists.digium.c
Is there a way in asterisk to Activate/Clear the blinking light on
polycom phones
indicating VM. Either from an AGI or some way in the dialplan?
I want to be able to control this light for from my application.
I have an AGI to do something similiar to VM and want to light /clear
the light mysel
On Fri, May 6, 2011 at 1:27 PM, Bob Beers wrote:
> Hi Steven,
>
> Can you put the .spec file from dahdi-linux-kmod package up?
Nevermind, I got it. :-)
Looking at it now.
Did you CC [Packager: Jason Parker ]?
- Bob
--
_
-- Ba
Currently we have following working page now i want to add custom ring type so
people pay attention. Anybody know about what would be the variable to change
custom ringer
[all-page]
exten => s,1,Set(TIMEOUT(absolute)=15)
exten => s,n,AGI(page.agi)
exten => s,n,SIPAddHeader(Alert-Info: Ring A
You are using an old format for specifying the mailbox. See "core show
application voicemail" for the correct usage. Also read ALL the UPGRADE*.txt
files.
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf
On Fri, May 6, 2011 at 6:30 PM, Rizwan Hisham wrote:
> I am in desperate need of this feature. I want to play background music
> during a call while the 2 parties are having some lovely conversation (or
> maybe give them a sort of cursing background if they are cursing each
> other).
Let's start
On Fri, May 6, 2011 at 11:58 AM, wrote:
> I am trying to install dahdi-linux from packages onto an OEL5u3 server which
> has an old kernel (5.2.6.18_128) and is a PAE variant. As there are no kmod
> packages now available for this kernel I am having to build them from source
> packages.
>
>
[snip
Boa tarde a todos,
Colegas estou a procura de um gateway GSM para ligar ao servidor asterisk de
nossa empresa, o objetivo é interligar clientes e parceiros comerciais a
nossa central, reduzindo custo das ligações para celular, porem ao ver o
regulamento dos planos oferecidos pelas operadoras vi qu
change "u500" in extension.conf because asterisk 1.8 user "500,u" like
following
exten => open-NOANSWER,1,VoiceMail(5800,u)
> Date: Fri, 6 May 2011 09:49:17 -0700
> From: bilmar...@yahoo.com
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] Configuring Voicemail in Asterisk 1
Hi All;
Already in the voicemail.conf file, I added the extension 500 and kindly find
below my voicemail configuration:
[Internal]
0 => 1234,Gama Operator,opera...@gama.com
500 => 1234,Operator,opera...@gama.com
501 => 1234,Employer Name,employer_em...@gama.com
502 => 1234,Employer Name,employ
>
The contents of this e-mail are intended for the named addressee only. It
contains information that may be confidential. Unless you are the named
addressee or an authorized designee, you may not copy or use it, or disclose it
to anyone else. If you received it in error please notify us immed
Has anyone else noticed that QueueCallerAbandon is not showing up in the AMI
after the 1.8.3.3? Am I missing something? I'm getting what seems like
everything else but QueueCallerAbandon.
v/r,
Me
--
_
-- Bandwidth and Colocation P
Hi,
On Thu, May 05, 2011 at 05:30:04PM -0400, Paul Belanger wrote:
> On 11-05-05 05:14 PM, Mark G Thomas wrote:
> >Hi,
> >
> >I think this must be a bug introduced with 1.6.2.17.something.
> >
> >When I upgrade from asterisk-1.6.2.16.1 to asterisk-1.6.2.17.2 or 1.6.2.18,
> >my AEL Dial() commands
I am trying to install dahdi-linux from packages onto an OEL5u3 server which
has an old kernel (5.2.6.18_128) and is a PAE variant. As there are no kmod
packages now available for this kernel I am having to build them from source
packages.
I have installed the dahdi-firmware-2.0.0-1_centos5 RPM
On 6 May 2011 16:30, Eric Wieling wrote:
>> -Original Message-
>> From: asterisk-users-boun...@lists.digium.com
>> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
>> Cassius Smith
>> Sent: Friday, May 06, 2011 11:23 AM
>> To: Asterisk Users Mailing List - Non-Commercial Discu
After many moons I have revisited this problem and found a solution that moves
the problem further up the stack. I will post my new problem separately but
just for completeness here is the solution.
Original problem: trying to build kmod-dahdi-linux for out of date PAE kernel.
Errors:
rpmbuild
Hi All,
I am in desperate need of this feature. I want to play background music
during a call while the 2 parties are having some lovely conversation (or
maybe give them a sort of cursing background if they are cursing each
other). I found this post which talks about creating a ghost call with the
I seem to recall this issue mentioned on asterisk-dev. Check issues.digium.com
and see if there is anything similar to your issue.
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
> Cassius Smith
> Sent:
Hi,
I use follow me and have several SIP phones answering, works nice but:
All phones that did not answer a call have the number in missed call
list even if answered by other ext.
CDR gets messy too. Difficult to see if call is answered.
I was thinking of possible solution: Turn of missed call o
Hi all,
I have a production server running with about 90 Cisco 79[46]1's and SIP
release 8.5(2)SR1 from last year. I was running Asterisk 1.6.2.9 and
upgraded last night after hours. (Seemed low risk to me!)
Much to my surprise, not a single one of the Cisco 79XX phones would
register. Since it's
Look at function CURL
-Original Message-
From: Daniel Isenmann
Sender: asterisk-users-boun...@lists.digium.com
Date: Fri, 6 May 2011 13:04:09
To: asterisk-users@lists.digium.com
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] TCP Trig
Daniel,
Have you thought about using CURL from the Dialplan?
Henk
Daniel Isenmann schreef:
Hi,
is it possible to configure a TCP trigger to a predefined address and
port on a incoming call request?
Some background:
For example “Client 1” tries to call “Client 2”, “Client 1” is sending
Hi,
is it possible to configure a TCP trigger to a predefined address and port on
a incoming call request?
Some background:
For example "Client 1" tries to call "Client 2", "Client 1" is sending the call
request to Asterisk (SIP-Server). Asterisk open a connection to the predefined
address an
>
>Thanks for the feedback, Ira. It makes me very sad to hear what you say and I
>hope that we can get more resources from the community to assist in the
>process to make it more friendly. We want to get those bug reports. The one
>thing I hate to hear when I'm travelling at conferences is that "oh
5 maj 2011 kl. 18.30 skrev Ira:
> At 07:56 AM 5/5/2011, you wrote:
>> So how can we fix this? How can we get more people involded? What makes
>> projects like FedoraTesting[3] and DebianTesting[4] popular? How can the
>> Asterisk project reproduce their success?
>
> Well, it's not a lot of
New Text at Bottom:
---
hi:
my current system is 1.6.2. I have dahdi hardware card. I must
disable res_timing_timerfd module or sometimes phone calls would
become silent suddenly.
I don't know the situation in 1.8. I heard that timing is still a
problem in 1.8. should I kee
hi:
my current system is 1.6.2. I have dahdi hardware card. I must
disable res_timing_timerfd module or sometimes phone calls would
become silent suddenly.
I don't know the situation in 1.8. I heard that timing is still a
problem in 1.8. should I keep using "res_timing_dahdi" or I can use
"res
On Thu, 2011-05-05 at 14:13 +, satish patel wrote:
> Hi All,
>
> Just wondering is it safe to use asterisk 1.8 latest branch on
> production ?
>
> http://svn.asterisk.org/svn/asterisk/branches/1.8/ Revision
> 317100
>
> -S
We've been running 1.8.3.2 with the patch to fix the local chan
At 01:07 PM 5/5/2011, you wrote:
Fair enough, what are some examples of questions you have? It only
takes a moment to create a new wiki page and start documenting
them. If you willing to provide the questions and feedback, I'm
more then happy to write them on the wiki.
So to start, I'm a us
I am still using Asterisk 1.4 because of the Asterisk GUI. I don't
understand why it was ever dropped, it's easy to setup (no SQL
databases), quick, works well and in my experiance it gets along with
manual config file changes.
The only real issue I've encountered with 1.4 is Digium can't seem to
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