Hello Guys,
Hope everyone is fine, I have one issue coming in asterisk , What i am doing
is i am generating a callback if some one calls at a specif access number on
asterisk,
Asterisk sends a busy signal to the calling party that he received a request
from party and then sends the call back to
JT djklut...@gmail.com writes:
I'm struggling with an intermittent crosstalk issue resulting in a
caller's audio being broadcasted to other calls (only one way as they are
unable to hear the others listening in).
This may be a long shot... I have experienced this when two SIP phones
had the
Hello
We need to know if a channel is not in use and can be used to dial a number
etc..
I have tried the ChanIsAvail function with different parameters.
ie ChanIsAvail(DAHDI/1DAHDI/2) ,ChanIsAvail(DAHDI/1,s) etc
no matter the channel is busy or not it always return 0 .
Please suggest
FYI
Warren Selby wcse...@selbytech.com writes:
I believe I spoke with Aastra and Snom at the Astricon tradeshow and
they said they support it on their newer models as well.
For Snom the enhancement request is SCPP-227, but I don't believe it has
been implemented. I can't find it in any release
Dear all,
i am using DGP 301 hard phone with my asterisk server.
1 : real time support is enabled ...all sip_buddies are stored in mysql
database...
2: when i register my phone for first time it works fine.receives 2 ,3 calls
then no call received
hangup cause is
On 14:59, Wed 25 Nov 09, ABBAS SHAKEEL wrote:
Hello
We need to know if a channel is not in use and can be used to dial a number
etc..
I have tried the ChanIsAvail function with different parameters.
ie ChanIsAvail(DAHDI/1DAHDI/2) ,ChanIsAvail(DAHDI/1,s) etc
no matter the channel is busy
Just for some information really : How many of you use multiple sip lines on
a phone ?.
I'm sitting here looking at my 7960, with it's 6 lines. I've every only used
one line, and I was wondering if I was a weirdo ;)
The only time I've ever found a use was when I had two systems (production
and
What version of Asterisk are you using?
I think this might be related to an issue that was resolved in version 1.4.27
http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-1.4.27-summary.html
- look in the list of Closed Items, second one down.
On Tue, Nov 24, 2009 at 11:03:16PM +, Mike wrote:
Folks,
I've got one of those GPO 1950's rotary dial phones that I'm trying to
get working in the UK. I've got pretty much everything working with my
TDM400, the phone rings and I can receive calls but I cannot dial with
the rotary
On 10:18, Wed 25 Nov 09, Julian Lyndon-Smith wrote:
Just for some information really : How many of you use multiple sip lines on
a phone ?.
I'm sitting here looking at my 7960, with it's 6 lines. I've every only used
one line, and I was wondering if I was a weirdo ;)
The only time I've
SIP schrieb:
Yes... you would have to register (and possibly pay for, dependent on
the ENUM registrar) each individual number. The idea behind ENUM is that
it's an E164 number that is already yours that maps to whatever you want
it to map to (email, SIP, etc). The key point here is that you
Thanks Michiel and Dan
@ Michiel i have checked the variables but they dont contain any value.
@Dan I am using 1.6.1.2 May be some issue with it ... In the mean while let
me test with an older version of asterisk
On Wed, Nov 25, 2009 at 3:19 PM, Dan Journo
d...@keshercommunications.comwrote:
Yes why not? when the agent is connected it can read the variables on the
calling channel what would you like to build with that? :)
l.
2009/11/24 Shaun Clark shaun_cl...@hotmail.com
Hello,
I was wondering if their is a way to use the Asterisk ACD to initiate a
call that will route
On Wed, Nov 25, 2009 at 12:26:10PM +0200, Tzafrir Cohen wrote:
On Tue, Nov 24, 2009 at 11:03:16PM +, Mike wrote:
Folks,
I've got one of those GPO 1950's rotary dial phones that I'm trying to
get working in the UK. I've got pretty much everything working with my
TDM400, the phone
Dan I have reverted to 1.4.27 but got no success. Same behaviour
Do anyone has any success with it ?
On Wed, Nov 25, 2009 at 3:54 PM, ABBAS SHAKEEL
shakeel.abbas@gmail.comwrote:
Thanks Michiel and Dan
@ Michiel i have checked the variables but they dont contain any value.
@Dan I am using
On 16:54, Wed 25 Nov 09, ABBAS SHAKEEL wrote:
Dan I have reverted to 1.4.27 but got no success. Same behaviour
Do anyone has any success with it ?
This ael snippet is working great for me on current -trunk.
I have been using this for some time now, it's from before 1.6 got
branched so it should
Greetings.
Attempting to connect Asterisk to LDAP database using res_config_ldap
module. While trying to register sip client (Ekiga softphone),
according to slapd.log, asterisk connects to LDAP server, asks for
some attributes to modify (they do exist, and asterisk user has all
permissions to do
I use two 'lines' though 'Line appearances' would be a better term, though
still confusing in my book.
One line for incoming, one line that auto-answers for paging.
Cisco really has so many line appearances on their phones to enable BLF using
SIP over TCP.
-Dave
From:
snip
Thats probably it You're relying on Asterisks software echo
canceling I have seen mixed results. Have you tried adjusting
gains? I'd do the following
1. Turn off echo canceler (makes it more obvious whilst you're trying
to remove it)
2. Turn down both gains
3. listening'
I wonder if this is related to my problem where the channel returns with
a status of BUSY even if it is on hook -- this is a dahdi channel.
ABBAS SHAKEEL shakeel.abbas@gmail.com wrote:
Dan I have reverted to 1.4.27 but got no success. Same behaviour
Do anyone has any success with it ?
If I get an echo cancellation module for my Digium TE121 card, will I need
to do any adjustments/configuration in Asterisk?
You should probably still set the gain using rxgain and txgain. IME,
it's much easier setting gains on a PRI than it is on a POTS line,
though. I've worked with a couple
I have two Asterisk server, running on Asterisk 1.6:
SRV1 = 192.168.0.5 on Asterisk 1.6.1.4
SRV2 = 192.168.0.20 on Asterisk 1.6.1.8
I want create a link for exchange call.
To clarify and expand on Aggio's response. You either need to have a
peer and user on both machines, or you
On Wed, Nov 25, 2009 at 5:18 AM, Julian Lyndon-Smith aster...@dotr.com wrote:
Just for some information really : How many of you use multiple sip lines on
a phone ?.
I'm sitting here looking at my 7960, with it's 6 lines. I've every only used
one line, and I was wondering if I was a weirdo ;)
I use two ‘lines’ though ‘Line appearances’ would be a better term, though
still confusing in my book.
I have five line appearances on the Snom190 on my desk. I regularly
use two line appearances, and on occasion, I have used three to juggle
back and forth between calls.
I would guess that a
On Mon, 2009-11-23 at 14:46 -0500, Dovey Forman wrote:
Regarding the email to multiple receipients, it is available on an ad-hoc
basis from the phone?
IE; call into the voicemail system, enter x digit to send a voicemail to
multiple users, record the message, then enter the destination
I use Polycom 501's with 2 LA's for production and 1 for testing.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Noah Miller
Sent: Wednesday, November 25, 2009 10:06 AM
To: Asterisk Users Mailing List -
hi,
we are running a switchvox system, and i would like to know what the
practice is for users who are working party in the main office and on some
other days with their laptops either from home of on the road...
right now i told them to unplugg the hardphone, coz having a softphone and
the
On Wed, 2009-11-25 at 08:46 -0500, David Gibbons wrote:
I use two ‘lines’ though ‘Line appearances’ would be a better term,
though still confusing in my book.
One line for incoming, one line that auto-answers for paging.
Cisco really has so many line appearances on their phones to enable
Rob;
That would be great. You could send directly to me @ dovey.for...@idt.net
or respond to this list.
I appreciate it!
--Dovey
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Robert
Lister
Sent: Wednesday,
I setup another extension for the softphone and enable followme on
their main extension to ring both. For example 8678 is the main and
38678 is their softphone. For users with more phones I just keep going
up 48678. This makes it fairly seamless to the end user and easy
enough to remember when
Thanks Michiel Covici
@ Michiel i will try the script
@Covice yes it is a DAHDI channel
On Wed, Nov 25, 2009 at 8:12 PM, cov...@ccs.covici.com wrote:
I wonder if this is related to my problem where the channel returns with
a status of BUSY even if it is on hook -- this is a dahdi channel.
At 02:18 AM 11/25/2009, you wrote:
Just for some information really : How many of you use multiple sip
lines on a phone ?.
I'm sitting here looking at my 7960, with it's 6 lines. I've every
only used one line, and I was wondering if I was a weirdo ;)
I've fought with the same question. When I
snip
Cisco 7960 does not do BLF (at least not on the SIP firmware) but the
7961 might. It's a shame they haven't added such features, but there we
go.)
/snip
Are you sure about this? I believe the 79xx series on 8x SIP firmware loads
does BLF with SIP/TCP, just not SIP/UDP.
-Dave
snip
On Wed, Nov 25, 2009 at 8:18 AM, Julian Lyndon-Smith aster...@dotr.comwrote:
Just for some information really : How many of you use multiple sip lines
on a phone ?.
I'm sitting here looking at my 7960, with it's 6 lines. I've every only
used one line, and I was wondering if I was a
hi ryan,
thx for ur suggestions. so , if i would go that route, that would mean i end
up with n-extensions per user based on n-locations. questions:
- how would i set the 'main' extension, so that other people see only one
extension in the phonebook / have to remember?
- is the caller-id when that
Hi
I have been using the CHANNEL variable as a way of checking if a user is
allowed to make outgoing calls, and what their source caller ID should be
(these values are in a database).
This works all of the time with SIP and most of the time with IAX, however
sometimes with IAX the channel
On 17:03, Wed 25 Nov 09, Robert Lister wrote:
On Wed, 2009-11-25 at 08:46 -0500, David Gibbons wrote:
I use two ???lines??? though ???Line appearances??? would be a better term,
though still confusing in my book.
One line for incoming, one line that auto-answers for paging.
Cisco
Using an Asterisk system running 1.2 with Aastra phones.
Would be a cause of static for inbound/outbound and ext to ext calls?
Its voip both in and out.
We swapped, phones, cordes, switches etc…..
Typically a reboot of the phone resolves the problem…person also swears
there is nothing on
snip Cisco 7960 does not do BLF (at least not on the SIP firmware) but the
7961 might. It's a shame they haven't added such features, but there we
go.)
It does with the skinny firmware :)
The skinny channel driver also comes with the 'random crash' feature ;-p. But
truth be told I only every
Hello all,
Do you know if it IS possible to use multiple lines/extensions on SIP with a
Cisco 7960 or other phone models? My boss wanted to have 1 physical phone but
have it register to a couple of different extensions, then use different
ringtones to identify which line was ringing when a
Travis Elsberry wrote:
Hello all,
Do you know if it IS possible to use multiple lines/extensions on SIP
with a Cisco 7960 or other phone models? My boss wanted to have 1
physical phone but have it register to a couple of different
extensions, then use different ringtones to identify
Norbert Zawodsky skrev:
SIP schrieb:
Yes... you would have to register (and possibly pay for, dependent on
the ENUM registrar) each individual number. The idea behind ENUM is that
it's an E164 number that is already yours that maps to whatever you want
it to map to (email, SIP, etc).
Hello,
We are in the process of splitting our phone system into two separate
logical systems for our two departments. One of the goals of this
switch is to restrict members of one department from transferring calls
to the other, but not restrict them from calling that department
themselves. So
Is it a single user? Or every single phone?
If it's a single user, and you can get hold of a UPS with power conditioning on
it, try plugging the various devices into it - there might be some dirty power
coming along.
From: asterisk-users-boun...@lists.digium.com
But then you create phonenumbers in enum, which doesn't exist as
pstn-numbers.
Not the idea behind enum.
On the other hand, if you owned 10 or 100 pstn-numbers in series, you
could get the last one or two digits delegated to your dns-server.
Leif
Brilliant, thanks a lot.
Best regards,
Örn
On Tue, Nov 24, 2009 at 1:39 PM, Santiago Gimeno
santiago.gim...@gmail.comwrote:
Hi,
I think it can be related to https://issues.asterisk.org/view.php?id=16268
Best regards,
Santi
2009/11/24 Örn Arnarson o...@arnarson.net
Hello again,
I
Can you add an agent dynamically to a queue with an external number, i.e.
cell phone as an extension? If so how? Thanks!
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or update
Hello.
I have a question regarind sound files in asterisk 1.6. I have a sound package
in ulaw format and I would like to know if I have a sip extension with
allow=alaw would asterisk convert that file to the codec the user is allowed to?
I am having a problem playing a file that exist in
Way back when, the wctdm driver needed a fix to make it more agreeable
to pulse dials in the US. I suspect this is also the case in the UK.
Speed and make break ratio are more critical , as pulse detection isn't
nearly as smart as a PSTN exchange
Search the wiki for more details.
How that
On Nov 25, 2009, at 3:07 PM, Dovey Forman wrote:
Would be a cause of static for inbound/outbound and ext to ext calls?
Its voip both in and out.
We swapped, phones, cordes, switches etc…..
Typically a reboot of the phone resolves the problem…person also
swears there is nothing on or
I believe ${IAXPEER(CURRENTCHANNEL)} should help you with the current
IAX2 name ... you can make DumpChan() to understand what kind of channel
variables you can use there.
--
razu
On 11/25/2009 09:57 PM, Nic Colledge wrote:
Hi
I have been using the CHANNEL variable as a way of checking if a
51 matches
Mail list logo
