2009/10/24 Darrick Hartman dhart...@djhsolutions.com
Oliver,
I have some comments, but am looking for a good answer to this as well.
1). 2.6.27 kernels to include a version of mISDN v2.0. It's not as up
to date as the full version on the mISDN.org site.
that would be great if someone
2009/10/25 Tzafrir Cohen tzafrir.co...@xorcom.com
On Sat, Oct 24, 2009 at 04:57:22PM +0200, Olivier wrote:
Hello,
I'm rather new to svn, so please, forgive me if this question sounds
naive
but is it normal that no trunk/ subdirectory is visible in
http://svn.asterisk.org/svn/libpri/ ?
Thank you Klaus and Martin for your answers!
It's very helpful!
--
-- --
Marc LEURENT
lf...@leurent.eu
Le vendredi, 23 octobre 2009 20.51:54, Martin a écrit :
You can call application Progress() from within dialplan and it will
cause the Asterisk to send a SIP reply 183
on the call that came
Tarek Sawah skrev:
you need to post you SIP.conf and your Extensions.conf so someone can
have a look at them and see if there is anything missing
what are the contexts you are using with your peers?
what is the dial plan triggered when calling your destination number?
Machine 1
Hello All;
I beleive as we have DUNDI in Asterisk, there should be a common community (a
website and so on) where those who has Asterisk boxes, they can exchange
traffic between each other.
Is there something like this?
I would like to have a service provider in low rates, any advise?
This question is more appropriate for the asterisk-biz list.
There are such clearinghouses for traffic peering indeed. I suspect
didx.net is likely to be brought up as an example.
bilal ghayyad wrote:
Hello All;
I beleive as we have DUNDI in Asterisk, there should be a common community (a
Ooops.. forgot. The versions of * are:
Machine 1: 1.6.1.4
Machine 2: 1.6.0.5
/Rob
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1. When i connected my analog phone to fxs card, i cannot get dial tone what
could be the problem?
I am using elastix 1.5.2 based on centos 5.2 Final.
2. On my 2 sip softphones using x-lite linux versions, i get one way audio how
do i solve this?. This problem is also present when i use a
Sir,
I am using Asterisk 1.4 and i want to record incoming call and i want to give
record file name will start with extension no. and add date and time. i know
it is possible with monitor but the problem is when i have receive incoming
call than i have ring 4 extension simultaneously and i
On Mon, Oct 26, 2009 at 05:02:18AM -0700, PATRICK KANGETHE wrote:
1. When i connected my analog phone to fxs card, i cannot get dial tone what
could be the problem?
What is the output of:
lsdahdi
dahdi_hardware
I am using elastix 1.5.2 based on centos 5.2 Final.
Consider also asking
Hi,
First a confession - The box in question is a 1.2.35 box, so this may
be solved in a newer version as I know the JB code is all hugely
changed, but... It may be worth checking into.
Scenario:
- IAX outbound call from Asterisk, which rings okay.
- Remote end sends ANSWER, which we
So this *should* work??
[outgoing]
- exten = s,1,Dial(DAHDI/1/5551212,20)
- exten = s,2,Noop(I hung up)
- exten = s,3,Hangup
- exten = h,1,Noop(you hung up)
- exten = h,2,Hangup
[incoming]
- exten = s,1,Answer
- exten = s,2,Noop(I hung up)
- exten = s,3,Hangup
- exten = h,1,noop(you hung up)
-
Thanks for all the information.
Benaiad: I will try adding in the hosts file and try it once again, also one
more that I was in regards to the harddrive, so I thought of replacing with
a SSD with high read and write speeds just to check whether its going to
reduce the dealy...
Regards
Sandesh
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Olivier a écrit :
|
|
| 2009/10/24 Jean-Denis Girard jd.gir...@sysnux.pf
| mailto:jd.gir...@sysnux.pf
|
| Olivier a écrit :
| | Hello,
| |
| | I'm evaluating to possibility to use chan_misdn as a short term
| | workaround, in case latest Dahdi is not
Hi folks,
I have a simple question regarding attended transfers. I have some
queues where agents take calls and I have configured attended transfers
between queues. That is, the agent dials the attended transfer extension
that routes it to the aproppiate transfer queue where the second agent
It's my understanding that the backport is available now in 1.4.
However, seem to be having some issues with it. Just wondering if I have
everything setup right.
I'm running 1.4.26.2 realtime.
queue_members:
`uniqueid` int(10) unsigned NOT NULL auto_increment,
`membername` varchar(40)
Hello!
I remember a while back I saw a way to answer a call from a device
that is not from the one ringing, but I don't remember what how to do
it. Any help would be great!
Thanks,
Elliot
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*8 is the default value in features.conf to pick up a ringing line if you
are in that ring group.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Elliot Murdock
Sent: Monday, October 26, 2009 2:52 PM
To:
I am using an 8 port tdm card and also I implemented a dialer using a
.call file generator. As you know on the .call you specify the channel to
call and then the contex/extension/priority to let dial plan continue when
the call is bridge.
My actual problem is that when the call process starts,
Agent 1 could park the call and have agent 2 pick it up from the lot.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Miguel Molina
Sent: Monday, October 26, 2009 12:00 PM
To: Asterisk Users Mailing List - Non-Commercial
Hi,
I have two servers ( A and B) in different towns.
Both servers have pstn attached to them. Now I need to have calls coming to
both servers to be answered on server A and then distributed between two sites.
What is the best way of doing ?
Having all calls to B forwarded to A on
It's not the DAHDI driver; it's the POTS service you are (presumably) using.
The DAHDI driver works fine with PRI/E1 interfaces, but POTS requires
human knowledge (it can't tell if a line is ringing/answered, etc). The
only reasonable solution I can suggest for this scenario is a
On Mon, 2009-10-26 at 14:58 -0500, Danny Nicholas wrote:
*8 is the default value in features.conf to pick up a ringing line if you
are in that ring group.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
One suggestion - use ex-girlfriend logic on server b to only allow pickup
of calls from Server A.
- exten = s,1/5551212,Answer
- exten = s,n,Hangup
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Asterisk Project Security Advisory - AST-2009-007
++
| Product | Asterisk |
I believe that for this, box B would have to answer the pstn line first …. I do
not want that to happen
Another question I have is how to configure Aastra phones to work with both
servers and continue to work when internet connection is down?
Thanks
From:da...@debsinc.com
In the example I gave you, B could only answer if the call was from box A
(substitute 5551212 for the number for box A).
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Robert
Augustyn
Sent: Monday, October 26, 2009 3:59 PM
For question #2, configure one line on each phone to A and one to B. If you
did an IAX2 connection from A to B, you could channel calls from A to B that
way and do PSTN answering on B if the internet connection was not up (IAX2
show peers).
_
From:
The Asterisk Development Team has announced the releases of Asterisk 1.6.1.8.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/
The release of 1.6.1.8 resolves an issue where an ACL check is not present for
verifying SIP INVITEs. For more
For some reason I am not able to set loopstart instead of kewlstart:
Console out put:
[Oct 26 20:58:40] == Parsing '/etc/asterisk/chan_dahdi.conf': [Oct 26
20:58:40] Found
[Oct 26 20:58:40] -- Registered channel 1, FXS Loopstart signalling
[Oct 26 20:58:40] -- Registered channel 2, FXS
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