You need 2 but they can point to the same table
sipusers =
sippeers =
You can get table definitions by downloading the source and then looking
in the
contrib/realtime/
directory
Ish
On Thu, 2012-05-03 at 04:56 +, Kamlesh Kumar wrote:
Hello,
For realtime configuration, in
On Wed, May 02, 2012 at 11:18:54AM -0500, Stephen J Alexander wrote:
Hello all,
I'm trying to solve a problem on a T1 span setup wherein calls are
apparently not hanging up properly.
CAS or PRI?
The system in question is using a Xorcom Astribank with 1 full and 1
partial T1 span, and
On 05/03/2012 07:17 AM, Fernando Berretta wrote:
Hi,
I'm analyzing how to make Asterisk communications secured End-To-End,
and not sure which is the best approach, SRTP + TLS seems to be secured
but.. at least by default, doesn't appear to be End-To-End allowing
Asterisk administrators to
Hi
We are using Spandsp + FreeSWITCH for receiving Fax over T.30 E1/PRI and
the results make us sad :(
I suppose Asterisk also has the option of using spandsp or a commercial
version from Commetrex. What are your experiences with receiving Fax on
spandsp or commetrex on Asterisk ?
Does it
Tzafrir,
Thanks for your response. I'll check into those items.
Regards,
Stephen J Alexander
MPBX, LLC
http://mpbx.com
832-713-6729
On Thu, May 3, 2012 at 4:39 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote:
On Wed, May 02, 2012 at 11:18:54AM -0500, Stephen J Alexander wrote:
Hello all,
If you're going full time hosted fax you will ultimately end up buying
a t.38/sip gateway like an Audiocodes Mediant.
On Thu, May 3, 2012 at 5:27 AM, Anita Hall anita.h...@simmortel.com wrote:
Hi
We are using Spandsp + FreeSWITCH for receiving Fax over T.30 E1/PRI and the
results make us sad
On 05/03/2012 10:35 PM, cjwstudios wrote:
If you're going full time hosted fax you will ultimately end up buying
a t.38/sip gateway like an Audiocodes Mediant.
Many people handling hundreds of thousands of FAXes per day would
disagree with that assessment.
On Thu, May 3, 2012 at 5:27 AM, Anita
Hello,
We are currently working on a project where using .call file on asterisk
spool, outbound calls will be made from a pri line and a voice clip will be
played.
We know that pri has a capacity of handling only 30 channels at a time.
Therefore, my worry is what happens if we write 100 files at
The other 70 will result into failure with .call file approach.
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
Hi Anita
On 4/05/2012, at 12:27 AM, Anita Hall wrote:
Hi
We are using Spandsp + FreeSWITCH for receiving Fax over T.30 E1/PRI and the
results make us sad :(
I am presuming you do mean T.30 (standard fax protocol but people don't mention
it much) not T.38 as I am not familiar with that
Hi Ashish
On 4/05/2012, at 3:41 AM, Ashish Agarwal wrote:
Hello,
We are currently working on a project where using .call file on asterisk
spool, outbound calls will be made from a pri line and a voice clip will be
played.
We know that pri has a capacity of handling only 30 channels at
IAX Modem with Hylafax is a perfect combo as such, it just works !!
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
In my experience the first 30 calls will go simulatenously, then the remainder
will go as lines become available. You should use /g1 or /r1 to allow the call
file to pick an open channel. Mitul is somewhat correct; all 100 calls will
try to process at once, so the 70 “laggards” will have to
So what is a better approach to achieve this
On May 3, 2012 9:20 PM, Mitul Limbani mi...@enterux.in wrote:
The other 70 will result into failure with .call file approach.
Regards,
Mitul Limbani,
Chief Architech Founder,
Enterux Solutions Pvt. Ltd.
110 Reena Complex, Opp. Nathani Steel,
The best approach to this would be to have a sender that uses AMI to
monitor channels and release .call files as channels become available.
About 100 lines in PERL.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ashish Agarwal
Sent:
If you set the ctime (or maybe mtime) of your spool file to a date in the
future, then asterisk won't process the .call file until that future time.
I recommend creating your call files with a random ctime/mtime for 0 - 240
seconds in the future and make sure you have a random retry time in
How can I check how many lines are currently being used?
On May 3, 2012 9:23 PM, Duncan Turnbull dun...@e-simple.co.nz wrote:
Hi Ashish
On 4/05/2012, at 3:41 AM, Ashish Agarwal wrote:
Hello,
We are currently working on a project where using .call file on asterisk
spool, outbound calls
easiest way is service asterisk status or asterisk -rx core show
channels verbose
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ashish Agarwal
Sent: Thursday, May 03, 2012 11:29 AM
To: Asterisk Users Mailing List - Non-Commercial
Ashish Agarwal писал 03.05.2012 18:53:
So what is a better
approach to achieve this
I've switched to AMI originate, call files did
not satisfy me for some reasons.
Besides, originating calls from
script gives you full control on when does each call start ant stop,
thus allowing you to
Or you could use a System call in the hangup dialplan and trigger a new
call as soon as an old one just finished. Maybe a silly idea but it shpuld
just work.
Ioan
--
_
-- Bandwidth and Colocation Provided by
Hi all.
I've got a perl script that connects to Asterisk's management interface using
Asterisk::AMI. So far, its proven to be very useful.
I'm hoping to use this to detect and respond to asterisk restarts and sip
reloads.
However, my script gets disconnected quite frequently, causing false
We handle tousands of faxes a day with asterisk 1.8 and spandsp but we
dictate what ATA the customers can use. Most of the faxes come in and are
processed as e-fax from asterisk.. It really comes down to how good are
your black art skills when It comes to fax and source providers channels
On 12-05-03 01:45 PM, Mike Diehl wrote:
Hi all.
I've got a perl script that connects to Asterisk's management interface using
Asterisk::AMI. So far, its proven to be very useful.
I'm hoping to use this to detect and respond to asterisk restarts and sip
reloads.
However, my script gets
On 12-05-03 03:47 PM, Paul Belanger wrote:
On 12-05-03 01:45 PM, Mike Diehl wrote:
Hi all.
I've got a perl script that connects to Asterisk's management
interface using Asterisk::AMI. So far, its proven to be very useful.
I'm hoping to use this to detect and respond to asterisk restarts and
Hello,
I want to send out 1000 faxes. I have an excel sheet of numbers and I have
Asterisk 1.8 installed from repository. I don't want to use a fax machine
or any ATAs or analogue equipment. How would Asterisk help me with faxing
these? and what add-ons do I need to make this possible?
I can
On Thursday 03 May 2012 1:47:09 pm Paul Belanger wrote:
On 12-05-03 01:45 PM, Mike Diehl wrote:
Hi all.
I've got a perl script that connects to Asterisk's management interface
using Asterisk::AMI. So far, its proven to be very useful.
I'm hoping to use this to detect and respond to
On 05/03/2012 01:28 PM, Bruce B wrote:
I want to send out 1000 faxes. I have an excel sheet of numbers and I
have Asterisk 1.8 installed from repository. I don't want to use a fax
machine or any ATAs or analogue equipment. How would Asterisk help me
with faxing these? and what add-ons do I
Lee,
Much appreciated for the input.
I am running all VoIP. SIP and IAX2 to our ITSPs. Please elaborate on
HylaFax and IAXmodems. Is there a guide posted on to get it running, or is
it part of the repository? Once installed how would one send .pdf as fax?
Thanks,
Bruce
On Thu, May 3, 2012 at
On 5/3/12 9:16 PM, Bruce B wrote:
Lee,
Much appreciated for the input.
I am running all VoIP. SIP and IAX2 to our ITSPs. Please elaborate on
HylaFax and IAXmodems. Is there a guide posted on to get it running, or
is it part of the repository? Once installed how would one send .pdf as fax?
James,
That is amazing details. I can use all of this. Thank you for sharing.
I am assuming you installed res_fax from repository?
*yum install asterisk18-res_fax_digium.i386*
And how did you install SpanDSP? Is there a guide you used?
I am aiming for multi-channels fax so the digium one won't
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