my lsdahdi output is;
1. [r...@elastix ~]# lsdahdi
### Span 1: WCTDM/8 "YSTDM8xx REV E Board 9" (MASTER)
1 FXSFXOKS (In use)
2 FXSFXOKS (In use)
3 EMPTY
4 FXSFXOKS (In use)
5 FXOFXSKS (In use) RED
6 FXO
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
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 inf
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: asterisk-users-boun...@lists.digium.
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
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 [mail
Asterisk Project Security Advisory - AST-2009-007
++
| Product | Asterisk |
|+---|
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 O
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
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
polarity/sil
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 telco’s
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 Dis
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, aste
*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: Asteris
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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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) defau
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
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Olivier a écrit :
|
|
| 2009/10/24 Jean-Denis Girard 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 stable enough for wha
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
On
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
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 immediately
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 aski
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 want
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 windo
Ooops.. forgot. The versions of * are:
Machine 1: 1.6.1.4
Machine 2: 1.6.0.5
/Rob
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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
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?
Regards
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 --
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 cam
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