Hi,
I have tried disabling and enabling crc4 before but that did not help.
I have not defined any signalling value under chan_dahdi.conf
Also, with respect to cabling we tried switching tx and rx but in that case
we see alarm on the dahdhi status.
--
put signalling=euroisdn in chan_dahdi.conf and restart asterisk.
Regards,
Mitul Limbani,
Chief Architech Founder,
Enterux Solutions Pvt. Ltd.
110 Reena Complex, Opp. Nathani Steel,
Vidyavihar (W), Mumbai - 400 086. India
http://www.enterux.com/
http://www.entvoice.com/
email: mi...@enterux.in
Dear friends,
I am not able to capture the CDR records for unanswered calls. Only one record
per call is coming in the CDR table. In the cdr.conf, I have enabled it by
setting :
unanswered = yes
From asterisk CLI also, I am getting this :
centos55server*CLI cdr show status
Call Detail
Already implemented Email to Fax in ICTFAX http://www.ictfax.org using both
sendmail and drupal mail handler module , you need to modify Fax part with
Voice call
Regards
*Tahir Almas*
Managing Partner
ICT Innovations
http://www.ictinnovations.com
Leveraging open source in ICT
On Sunday 23 September 2012, Ashish Agarwal wrote:
For some reason pri show spans does not show up. Can someone assist me to
fix this issue.
DAHDI is fussy about its configuration files. A single misconfiguration
anywhere
can break the whole thing, so get it working with just one span first.
I have noticed a peculiar problem recently with the way that the
failover operates in my dialplan.
I normally have:
1,Dial(SIP/provider-1/extension)
n,Dial(SIP/provider-2/extension)
(or something similar).
This has up until now worked flawlessly.
If there is an error with
Why not use the DIALSTATUS channel variable to determine if a fail over is
necessary?
- Logan
On Sep 24, 2012 6:00 AM, Thomas Kenyon dig...@sanguinarius.co.uk wrote:
I have noticed a peculiar problem recently with the way that the failover
operates in my dialplan.
I normally have:
Hello everyone,
I stuck in problem I have creating a time based IVR and its working fine.
If my IVR playing in office hour it would standard IVR and if not they we
have play a greeting message and place that call to voice mail of
a extension.
My problem is this I am able to transfer the call on
Hi Jordan,
Thanks for all, but i found this bug in Asterisk :
https://issues.asterisk.org/jira/browse/ASTERISK-16465
Attached the patch to fix the problem, if the online site does not work.
Thanks for all
Best Regards
-Messaggio originale-
Da: asterisk-users-boun...@lists.digium.com
Dear List memebers,
We as voice quality testing sotware vendor are very interested in your
opinion on demands (Asterisk owners, call centers, etc) for a voice
quality impairments detection library we plan to release. Currently the
library can detect:
- clicking
- clipping
- stuck
- SNR
- echo
Hello all,
I inherited an Asterisk 1.2 machine and I have a question about the order
of operations.
I want to give people the ability to dial specifics and block others. For
example, lets say NYC
[allowed]
exten = _1212555., 1,Authenticate(pins||3,j)
exten = _1212555.,
On Monday 24 September 2012, Asterisk Newb wrote:
Hello all,
I inherited an Asterisk 1.2 machine and I have a question about the order
of operations.
I want to give people the ability to dial specifics and block others. For
example, lets say NYC
[allowed]
exten = _1212555.,
On Mon, Sep 24, 2012 at 12:43 PM, A J Stiles
asterisk_l...@earthshod.co.ukwrote:
Asterisk always tests against the most specific (= hardest-to-match)
wildcarded extensions first, regardless of the actual order in the
dialplan.
Since _1212555. is harder to match than _1212., the former will
You are doing it wrong. I know 50 bazillion Asterisk dialplan examples on the
internet do it the same way. It is still wrong.
When you do a Dial on the dialplan you need check the value of DIALSTATUS or
HANGUPCAUSE before dialing again. Both variables will give you some indication
of why
I think a lot of people leave it out in examples for simplicity's sake. It
doesn't instil proper practices in folks' heads.
- Logan
On Sep 24, 2012 12:06 PM, Eric Wieling ewiel...@nyigc.com wrote:
You are doing it wrong. I know 50 bazillion Asterisk dialplan examples on
the internet do it the
On Mon, Sep 24, 2012 at 09:02:47AM +0100, A J Stiles wrote:
Make sure you set the jumpers on the card correctly for E1
operation (they are often set as T1 on delivery)...
A minor FYI not directly related to this discussion: Since
DAHDI-Linux 2.6.0 the jumpers can be overridden in the
[allowed]
exten = _1212321.,s,Goto(denied,s,1)
exten = _1212333.,s,Goto(denied,s,1)
exten = _1212456.,s,Goto(denied,s,1)
exten = _1212555., 1,Authenticate(pins||3,j)
exten = _1212555., 2,Dial(SIP/${EXTEN)@mycarrier)
exten = 102,Hangup
[denied]
exten = s,1,Playback(num-outside-area)
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Asterisk Newb
Sent: Monday, September 24, 2012 1:31 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Dial plan order of operations
Hi,
Does anybody here have experience using Asterisk as an FXO to emulate a
E1/T1/PRI line for test purpose?
Gateway or PCI card?
Thank you!
HB
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to
Going to n+101 was deprecated in Asterisk 1.2 or 1.4. Don't use it.. Read the
docs for Authenticate and see what diaplan variables you can check.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Asterisk
On Mon, Sep 24, 2012 at 2:55 PM, Eric Wieling ewiel...@nyigc.com wrote:
Going to n+101 was deprecated in Asterisk 1.2 or 1.4. Don't use it..
Read the docs for Authenticate and see what diaplan variables you can
check.
Thanks, situated the problem with the following:
exten =
Currently running version 1.8.16.0 and trying to manage confbridge rooms and
users. When I try to use the confbridge cli command I get a command not found
error.
CLI confbridge
No such command 'confbridge' (type 'core show help confbridge' for other
possible commands)
I've tried googling
Gary Carr wrote:
Currently running version 1.8.16.0 and trying to manage confbridge
rooms and users. When I try to use the confbridge cli command I get
a command not found error.
CLI confbridge
No such command 'confbridge' (type 'core show help confbridge' for
other possible commands)
On Mon, 24 Sep 2012, Asterisk Newb wrote:
Thanks, situated the problem with the following:
exten = _212555.,1,Authenticate(/etc/asterisk/pins||3,j)
exten = _212555.,2,Dial(SIP/${EXTEN:3}@level3,90,tr)
Two suggestions:
1) Using the 'n' priority will make your dialplans more maintainable.
2)
Hello,
I'd like to start using realtime hints in my asterisk 1.8 dialplan, but I
am unable. I haven't understood if they have to be put inside the
extensions realtime table (with priority -1) or if a dedicated realtime
hints table can be made. Neither ways seem to work. Have you any working
On Mon, 24 Sep 2012 23:37:32 +0200
Leandro Dardini ldard...@gmail.com wrote:
I'd like to start using realtime hints in my asterisk 1.8 dialplan,
but I am unable. I haven't understood if they have to be put inside
the extensions realtime table (with priority -1) or if a dedicated
realtime
Eric Wieling wrote:
You are doing it wrong. I know 50 bazillion Asterisk dialplan examples on the
internet do it the same way. It is still wrong.
When you do a Dial on the dialplan you need check the value of DIALSTATUS or
HANGUPCAUSE before dialing again. Both variables will give you some
We use something like below
[blf]
exten =_ZXX!,hint,SIP/${ODBC_FINDEXTN(sd.name,${EXTEN})}
This uses an odbc call to create the hint when the phone asks for it.
Using snom 760 and 821
Cheers
Stephen
--
_
-- Bandwidth and
Hoping for some clarification. I would like to setup a NORMAL (not
T.38) fax machine on an ATA, and have the ATA be a T.38 gateway to a
remote asterisk (1.8) server, which is doing T.38 relay (passthru) to a
provider.
Some amount of googling today seems to imply that most ATAs are just
Hi Mitul,
Thanks for reply.
If you can possible then please let me know download location and steps or
document for setup Video IVRS on Asterisk.
Thanks in advance.
--
Best Regards,
Rajni Vanza
Consultant Technology
---
Working On
Hello, all!
My name is Andrew. I have OpenVox TDM400P card with 2 modules (FXO and
FXS). It plugged to server asterisk (1.8.13.1) under gentoo (3.3.8-gentoo)
via dahdi (2.6.1 from gentoo portages). All works, but I have a problem:
When I whant to transfer call from FXS to other phone after
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