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
DID
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
of std
802.11 based wifi phones. Giving QoS on wifi is a big pain.
Hope that helps,
Regards,
Mitul Limbani
Enterux Solutions
On May 6, 2012 11:34 PM, Nunya Biznatch aster...@ihearbanjos.com wrote:
I'm about to receive approval to design and deploy an Asterisk-based
phone system for my company. I
Used the Snom M9 Wifi DECT phones, they work like charm.
SIPDroid on Android phones work good too, however latency is going to be
nightmare for u in softphone n wifi kinda scenario.
Use good quality Access Points like Ruckus Wireless.
Mitul
On May 7, 2012 1:55 PM, Bart Coninckx
of the issues.
You should definitely give it a shot.
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
DID: +91-22-61447605
) very effectively, infact a Multi
Function device (MFD) like smart phone has this issue.
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
I suggest you put in dtmfmode=auto (so rfc2833 / inband ) can be selected
dynamically.
Wanted to check with the community if this feature holds true on latest
versions of Asterisk ?
Regards,
Mitul Limbani,
Chief Architech Founder,
Enterux Solutions Pvt. Ltd.
110 Reena Complex, Opp. Nathani
Hey Mahendra,
We happen to still stock the same X100P cards in India.
If you require one, do connect with me off the list.
Regards,
Mitul Limbani,
Chief Architech Founder,
Enterux Solutions Pvt. Ltd.
110 Reena Complex, Opp. Nathani Steel,
Vidyavihar (W), Mumbai - 400 086. India
http
yeah, put qualify=2000 to ensure that you shall get the latency perfectly.
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
Dear Moises,
Does Sangoma manufacture 4 port gsm card? or is that an Openvox card?
Mitul
On May 26, 2012 12:23 PM, Moises Silva moises.si...@gmail.com wrote:
On Sun, Apr 1, 2012 at 9:04 AM, bilal ghayyad bilmar...@yahoo.com wrote:
Dears;
I am looking to get a telephony card that has GSM
You need to look at Redfone fonebridges to achieve this.
Please connect with me offline, we have it working in India in our
CloudVoice Infraatructure.
Mitul Limbani
On May 31, 2012 12:40 PM, Amit Patkar | ATPL a...@avhan.com wrote:
Hi
Lot of users have deployed Asterisk in virtualized
Any changes inside chan_dahdi requires asterisk restart.
you can restart asterisk gracefully, where by asterisk will honor the
existing calls, but wont honor new calls till it restarts.
Regards,
Mitul Limbani,
Chief Architech Founder,
Enterux Solutions Pvt. Ltd.
110 Reena Complex, Opp. Nathani
a good idea.
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
DID: +91-22-61447605
Cell: +91-9820332422
On Sat, Jun
Welcome to da Matrix :)
Look at this issue : https://issues.asterisk.org/view.php?id=6683
And try different combinations suggested over there, you might get lucky :)
Regards,
Mitul Limbani,
Chief Architech Founder,
Enterux Solutions Pvt. Ltd.
110 Reena Complex, Opp. Nathani Steel,
Vidyavihar
For 150 phones I would suggest you dedicate a machine, mebbe a Core 2 Duo.
Voice quality on virtualized platforms is troublesome, hence not
recommended for production usage.
Home usage for less then 2-5 phones it works perfectly fine in virtualized
env.
Regards,
Mitul Limbani,
Chief Architech
Things that look simple r quite complex to build :-)
Indian Accent ASR on proper names is herculean task.
No speech recognition known to mankind as of date can handle so many
dialects being spoken in India, so in short what you want is nice to have,
but nearly impossible to develop.
Better try
Mebe your operator doesnt like the CallerID(num) set as NULL just remove
the callerid(num) statement and let the standard callerId get set by
network.
Regards,
Mitul Limbani,
Chief Architech Founder,
Enterux Solutions Pvt. Ltd.
110 Reena Complex, Opp. Nathani Steel,
Vidyavihar (W), Mumbai - 400
you need to either use chan_ss7 or libss7.
Also look for mailing list archives of asterisk-ss7
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
messages you mean SMS ?
If its SMS then thats not covered here in any of the stacks !!
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
Thats precisely what asterisk has to offer.
Mitul
On Jul 23, 2012 9:53 AM, Kannan vasdevelo...@gmail.com wrote:
Hi List,
Is it possible for me to setup PBX, IVR and Conferencing platforms from a
single installation with Asterisk?
Thanks.
--
FreeSWITCH and
yeah well this is Asterisk Users community, I m not sure how others would
react to this, however these are my views.
If you need professional help, do connect with me off the list.
Regards,
Mitul Limbani,
Chief Architech Founder,
Enterux Solutions Pvt. Ltd.
110 Reena Complex, Opp
Try enabling crc4 on ur config.
Mitul
On Jul 24, 2012 10:38 PM, Russ Meyerriecks rmeyerrie...@digium.com
wrote:
On Tue, Jul 24, 2012 at 01:45:01PM -0300, equis software wrote:
Is a normal functionality?
when I do
#dahdi_cfg -vv
If I do this a lot of times...then
In rapid
If u r then try removing it.
Mitul
On Jul 24, 2012 11:09 PM, equis software equissoftw...@gmail.com wrote:
I´m using crc4 conf
span = 1,0,0,ccs,hdb3,crc4
bchan = 1-15,17-31
dchan = 16
The alarm clrears few seconds later...
On Tue, Jul 24, 2012 at 2:14 PM, Mitul Limbani mi...@enterux.in
Use AMI
Mitul
On Jul 25, 2012 2:37 AM, sathiish kumar sathiish.ku...@gmail.com wrote:
I wanna be able to pass arguments to macros when i initiate a call through
the originate command.. Is there any possible way of passing arguments to
the originate command in some way ?
--
NO SIP / IAX Trunking allowed on IP.
Apart from that everything is legitimate.
We do provide Hosted Asterisk connected with PRI circuits, do contact
off-line for more info.
Regards,
Mitul Limbani,
Chief Architech Founder,
Enterux Solutions Pvt. Ltd.
110 Reena Complex, Opp. Nathani Steel
EM signalling do not have seperate signalling channel.
Configure as em=1-31
Mitul
On Jul 26, 2012 6:40 AM, Jorge Mendoza jmendo...@tcc.com.pe wrote:
Hi,
We are trying to connect an Asterisk server with a Channel Bank with EM
interfaces using a RedFone TDMoE device.
The CB have a E1 CAS
signalling does not have
separate signalling channel, that is true for the T1 but not for E1. My
understanding is that E1 CAS pass the EM information in the bits abcd of
channel 16, the signalling channel.
Regards
--
Jorge Mendoza
--
*From: *Mitul Limbani mi
I think its not inbound call its outgoing, and during call progress the
remote end events are not passing back to sip.
Mitul
On Jul 27, 2012 10:36 PM, Raj Mathur (राज माथुर) r...@linux-delhi.org
wrote:
On Friday 27 Jul 2012, Tim Nelson wrote:
Another mystery for the list, hopefully someone
by writing dialplan
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
DID: +91-22-71967121
Cell: +91-9820332422
they get so much productivity
out of the same box.
If any one need redfone devices with high availability and high
performance, do connect with me off the list.
Regards,
Mitul Limbani
Enterux Solutions
www.enterux.com
On Jul 29, 2012 10:15 AM, Raj Mathur (राज माथुर) r...@linux-delhi.org
wrote
What you want can be done by OpenVBX, why dont you try exploring that model
?
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
Not a good idea :)
Asterisk at max we have seen supported 16 PRIs on single machine.
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
This has been happening since the asterisk 1.2 days, makes me believe it
has something to do with Analog FXO ckts provided.
Mitul Limbani
On Sep 12, 2012 10:18 AM, Vladimir Mikhelson v...@mikhelson.com wrote:
Raj,
I am just confirming it happens here as well.
CentOS 5.7. Asterisk 1.8.15.1
Raj,
Problem 1 is where asterisk times out and line gets free
However for problem 2
There is no way that this line frees up, as it depends upon remote side
infra of caller, if they calling from pri ckt they possibly could identify
our hangup signal, but if they calling from Analog exchange this
Operator sends callerId after 1st small ring (actually this is not audible
since its very small duration ring) post which all the data flows.
However, sometimes due to line distrubance this first small ring is missed.
Regards,
Mitul Limbani,
Chief Architech Founder,
Enterux Solutions Pvt. Ltd
question worth asking in mysql user list then here !!
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
DID: +91-22
this is quite complicated to be setup. however you can try using :
asterisk 1.4.11 with libpri patch for h324m and app_h324m.
Regards,
Mitul Limbani,
Chief Architech Founder,
Enterux Solutions Pvt. Ltd.
110 Reena Complex, Opp. Nathani Steel,
Vidyavihar (W), Mumbai - 400 086. India
http
Are you sure if PRI is the signalling or its EM based E1 links.
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
DID
Signalling frm remote side is down.
Also just add crc4 in span 1,0,0 in dahdi/system.conf just like other spans.
what is signalling= defined in your asterisk/chan_dahdi.conf ?
Regards,
Mitul Limbani,
Chief Architech Founder,
Enterux Solutions Pvt. Ltd.
110 Reena Complex, Opp. Nathani Steel
On Mon, Sep 24, 2012 at 10:11 AM, Raj Mathur (राज माथुर)
r...@linux-delhi.org wrote:
On Monday 24 Sep 2012, Mitul Limbani wrote:
Signalling frm remote side is down.
Also just add crc4 in span 1,0,0 in dahdi/system.conf just like other
spans.
what is signalling= defined in your
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
I guess you are looking for event handler, which can be polled
programatically n not via manual command entry?
Mitul
On Oct 18, 2012 8:53 PM, Danny Nicholas da...@debsinc.com wrote:
The AMI Command function issues CLI commands, but carry on.
** **
*From:*
Short answer is, its not possible
Long answer, why it is not !!
U would have to write a dahdi module for this 3G modem to help asterisk
understand it as standard gsm channel.
Hope that help,
Mitul
On Oct 18, 2012 9:16 PM, Mahendra Dobariya mahendra_mahen...@hotmail.com
wrote:
hi,
I want to
Need dummy to provide timing on machines that do not have a tdm board. Also
meetme dependency was on dummy or one of the tdm card.
I believe meetme has been rewritten since then.
Mitul
On Oct 23, 2012 9:58 PM, Warren Selby wcse...@selbytech.com wrote:
If I remember correctly, dahdi dummy was
yealink T18 and T20 are decent phones available for $60
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
DID: +91-22
Not possible to have same sip usernames.
However you can create
custA_user1 == 101
custB_user1 == 101
In the dialplan context.
Mitul
On Oct 30, 2012 12:47 PM, Darin Iv adari...@gmail.com wrote:
Hi all,
I need to configure DIDs for different companies and they should reach on
different
FYI
SIP usernames =! Extensions
You have to use unique sip usernames to be identified inside dialplan for
mapping to extensions
[contextA]
Exten = 101,1,Dail(sip/custA_user1)
[contextB]
Exten = 101,1,Dial(sip/custB_user2)
Hope this makes it clear.
Mitul
On Oct 30, 2012 1:16 PM, Darin Iv
Stop asking same questions !!!
On Oct 31, 2012 11:54 PM, Darin Iv adari...@gmail.com wrote:
Is it possible to bul multitenant system using some third party opensouce
application My design is like this.
Company A:
Context Company_A
IVR Company A
Extensions: 101,102,103,104 etc.
Company B:
Dont enable everything that you see in installation without doing homework.
Just delete the extracted directory and reextract from tarball n follow the
INSTALL.txt peacefully !!!
Mitul
On Nov 4, 2012 12:30 PM, akhilesh chand omakhileshch...@gmail.com wrote:
Hello, I am working with CentOS 5.3,
AFAIK its a propreitary card from Aculab and wont work on Asterisk unless
you buy software or support or both from them.
My advice is to dump it n get a digium card in same or lesser cost which
you need to pay aculab.
Mitul Limbani
On Nov 12, 2012 1:23 PM, RAJNI VANZA rajniva...@gmail.com wrote
Any changes inside chan_dahdi requires you to unload module chan_dahdi and
load module chan_dahdi, in case you dont wish to.restart asterisk.
pridialplan = national or unknown should help you solve the problem,
however you need to unload n load dahdi module.
Mitul
On Nov 21, 2012 10:26 PM,
You might want to share the know how over here if its not a chan_sip patch.
Mitul
On Nov 28, 2012 12:28 AM, Ron Wheeler rwhee...@artifact-software.com
wrote:
On 27/11/2012 12:58 PM, Christopher Harrington wrote:
It's an open source project. Pay a programmer or make the modification
yourself
Mebbe you guys should try snom m9 dect ip phone, i have been using it since
over 3 years now without any of these issues.
Mitul
On Dec 12, 2012 4:25 AM, Kai-Uwe Jensen kujen...@gmail.com wrote:
Using a Gigaset C610IP here, and am very happy with the features. The base
station can handle two
latency in DAHDI/Mobile connections.
Latency on DAHDI - heard this for first time
TDM networks have zero latency we face latency only on IP (SIP) networks.
Mitul Limbani
--
_
-- Bandwidth and Colocation Provided by http
+1 here.
On Jan 10, 2013 5:50 AM, Steve Totaro stot...@totarotechnologies.com
wrote:
On Wed, Jan 9, 2013 at 7:03 PM, chris tknch...@gmail.com wrote:
On Wed, Jan 9, 2013 at 2:02 PM, Doug Lytle supp...@drdos.info wrote:
What were the senders IP(s)?
Will have to look it up when I get home.
I would suggest to use linux ha and use same ip, which can failover to
second standby server using heartbeat.
This activity takes less then 5secs.
Mitul
On Jan 18, 2013 9:42 PM, Ishfaq Malik i...@pack-net.co.uk wrote:
On Fri, 2013-01-18 at 18:06 +0200, Onur Cem Çelebi wrote:
Hello,
we
Hi,
You might want to use ${MACRO_EXTEN} variable inside to preserve exten
variable of the original dialplan exten variable.
Mitul
On Feb 24, 2013 4:04 PM, Leandro Dardini ldard...@gmail.com wrote:
I just discover an hidden problem with AEL macro I want to have your
feedback. If you use a
to use?
Any DB integration layer inside IVR?
Mitul Limbani
On Mar 8, 2013 5:20 PM, nik600 nik...@gmail.com wrote:
Dear all
i'm planning a migration to asterisk for a high volume IVR service
(from 1000 to 1500 concurrent call)
The IVR service is based only on DTMF tones so the features required
convert the calls from PRI to SIP and throw it inside the VirtualBox
Asterisk, thats the ONLY WAY OUT
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
Hey,
Can you send me URL to download the tar ball pls?
Mitul Limbani
On Saturday, March 16, 2013, Robert Krakora wrote:
Hi,
If anyone is interested, I have ported app_rtsp.c to Asterisk 11.x. I
have tested it with GStreamer RTSP server and a C920 webcam streaming H264
SVC video from one
Use pri_net as signalling mode inside chan_dadhi.conf in /etc/asterisk/
folder.
You can set this up using any pri card thats supported on Asterisk.
Mitul
On Mar 31, 2013 12:25 PM, Dimitar Dimitrov ddimit...@consult.bg wrote:
Hello everyone.
I am looking for a E1 PRI card which supports
Why dont u run a reverse dialer on the admin contacts phone number. Leave
him clueless as well.
Mitul
On Apr 5, 2013 1:25 AM, Joseph syscon...@gmail.com wrote:
I receive several calls from this scamer: Senior SafeAlert
It is an automated call and they keep rotating their caller ID so it is
,
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
DID: +91-22-71967121
Cell: +91-9820332422
,
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
DID: +91-22-71967121
Cell: +91-9820332422
to a parking slot right from the originate, because that is cheaper than
running conferences, and then joining the second call right to the parked
call, so that all we have to do is two originates.
l.
2013/5/14 Mitul Limbani mi...@enterux.in javascript:_e({}, 'cvml',
'mi...@enterux.in');
Dial first
Not recommended to run Asterisk on Virualization.
Mitul
On May 18, 2013 11:33 PM, Rafael dos Santos Saraiva rafaels...@gmail.com
wrote:
Hi
I would like the opinion of you and if anyone has a similar scenario. I
have a project for installation of a Asterisk server in a client with about
400
Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Regards,
Mitul Limbani,
Chief Architech Founder,
Enterux Solutions Pvt. Ltd.
110 Reena Complex, Opp. Nathani
Without posting exact error messages, dont expect help !!
Mitul
On May 28, 2013 1:01 PM, upendra uppi...@gmail.com wrote:
hi,
anyone can help me to debug this ??
--
upendar
On Mon, May 27, 2013 at 4:09 PM, upendra uppi...@gmail.com wrote:
hi,
chan_local and res_crypto are building
in the memuselect the chan_sip module driver
showing as XXX to enable for building.
--
Upendra.
On Tue, May 28, 2013 at 1:03 PM, Mitul Limbani mi...@enterux.in wrote:
Without posting exact error messages, dont expect help !!
Mitul
On May 28, 2013 1:01 PM, upendra uppi...@gmail.com wrote:
hi
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Regards,
Mitul Limbani,
Chief Architech Founder,
Enterux Solutions Pvt. Ltd.
110 Reena Complex, Opp. Nathani Steel,
Vidyavihar (W), Mumbai - 400 086. India
http
://lists.digium.com/mailman/listinfo/asterisk-users
--
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
DID: +91-22
Interesting.
You might want to consider paying some expert for consulting ?
Mitul
On Jun 22, 2013 7:21 PM, Nick Khamis sym...@gmail.com wrote:
Hello Everyone,
We are currently having talks with various service providers, and
trying to determine what the best way is to interconnect in order
Chan_zap has been deprecated more then 2-3 yrs back. You might have to ping
ipcortex helpdesk to get fix.
Mitul
On Jul 11, 2013 4:32 PM, Xavier Singer - EcuTek xav...@ecutek.com wrote:
We use an IPcortex PABX running Asterisk 1.2.39-BRIstuffed-0.3.0-PRE-1y-y.
We have recently implemented Call
/asterisk-users
--
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
DID: +91-22-71967196
Cell: +91-9820332422
of a
particular sequence of events off-hand where it would happen though.
Richard
--
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
Are these end points Hard IP Phones having g729 codec?
If yes then you dont need any license. Just download passthrough g729
license.
Mitul
On Oct 2, 2013 1:17 PM, Frederic Van Espen frederic...@gmail.com wrote:
On 10/02/2013 09:33 AM, s m wrote:
and the last question is how many license
Nailed it to the point Matt +1 on.this entire philosophy of open source.
Mitul
On Oct 14, 2013 7:19 PM, Matthew Jordan mjor...@digium.com wrote:
On Sun, Oct 13, 2013 at 2:06 PM, CDR vene...@gmail.com wrote:
snip
I need Digium to store this IP in the CDR. I will be honest with the
Paste ur chan_dahdi.conf n sip.conf on both servers to pastebin and fwd
link here.
Mitul
On Oct 20, 2013 11:07 AM, akhilesh chand omakhileshch...@gmail.com
wrote:
Dear All,
I have pri with E1 facility that have 30 line and 100 pri number which is
provided by service provider.Number started
negotiations
Mitul Limbani
www.facebook.com/enterux
www.facebook.com/entvoice
On Oct 29, 2013 1:30 AM, Ron Wheeler rwhee...@artifact-software.com
wrote:
I am reaching the same level of frustration.
I have tried to find the source of the problems.
We have IAX2 to our VoIP provider and SIP
Just dont configure those spans n related channels inside chan_dahdi.conf
Mitul
On Nov 1, 2013 3:38 PM, Dmitry Melekhov d...@belkam.com wrote:
Hello!
Just got new server with TE420.
Not all four spans will be used immediately, but spans not configured or
not connected blink red light.
Is
? or if there are suggestions on best way to approach this problem.
Thanks,
--
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
If using IAX then I would recommend setting up DUNDi or Switch statement in
dialplan.
Mitul
On Nov 17, 2013 12:50 PM, Steve Edwards asterisk@sedwards.com wrote:
On Sat, 16 Nov 2013, Doug wrote:
I want to be able to pass any number (variable length) to a context and
then forward that to
Try following in chan_dahdi
immediate = yes
echocancel = no
dtmfmode = auto
Mitul
On Nov 29, 2013 1:42 PM, isr...@gmail.com wrote:
Are you using a mp3 file?
I have noticed that using control playback with a mp3 file I cannot use
the keypad to control the playback
-Original Message-
thanks a
lot for your help and support now ican stop the speech and go to my context
i really appreciate your help and support
2013/11/29 Mitul Limbani mi...@enterux.in javascript:_e({}, 'cvml',
'mi...@enterux.in');
Try following in chan_dahdi
immediate = yes
echocancel = no
dtmfmode
As per that theory 3CX should have been public by now !!
Mitul
On Dec 4, 2013 8:49 PM, CDR vene...@gmail.com wrote:
Digium is 100% lost in the map. If they would come up with a Paid
version of Asterisk, one that would use the .NET framework in Windows,
something simple to install, they could
Use FreeSWITCH !! Thats what you want on your winblows system, so suit
yourself my friend.
Mitul
On Dec 5, 2013 12:43 AM, Ruddy Gbaguidi plugwo...@micnes.com wrote:
I never tought this is become a Linux vs Windows fight.
We have been using asterisk on linux from a long time now and happy with
. There is no DNS so straight
IP addressing is used.
--
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
DID: +91
://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
--
Regards,
Mitul Limbani
like Inphonex, Broadvoice... and etc
Is there any suggestions for the service providers.
Regards
--
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
Hello,
Using Single Server with multiple VMs essentially kills the purpose, coz it
doesnt protect against physical hardware failures.
To save costs, use low end box as failover, to keep u in business, till
primary box goes live.
Mitul
On Mar 6, 2014 8:51 PM, Thorolf Godawa nos...@godawa.de
You can achieve this by setting relevant sip flags in the dialplan back and
forth.
Mitul
On Mar 12, 2014 11:18 PM, DHAVAL INDRODIYA dhaval.it01...@gmail.com
wrote:
Thanks Amit,
I want following scenario.
INCOMINGCALL --- MSC (SIP-T) PBX (Asterisk)
OUTGOINGCALL --- PBX (Asterisk)
V_/_ Heckler Koch - the original point and click interface
--
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
Hello,
I was able to use webrtc2sip and connect audio calls in g729 passthrough
and ulaw modes over a callus webpage js.
However not tested Video.
and it worked good even on AST 1.8.XX
Regards,
Mitul Limbani,
Chief Architech Founder,
Enterux Solutions Pvt. Ltd.
110 Reena Complex, Opp
appreciated.
If this can be simply implemented using asterisk and call folder, even
better
PS Our preferred version of * is 1.8.x
Kind Regards,
Nick from Toronto.
--
Regards,
Mitul Limbani,
Chief Architech Founder,
Enterux Solutions Pvt. Ltd.
110 Reena Complex, Opp. Nathani Steel,
Vidyavihar
Hello,
Do respond back Offline ..
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
DID: +91-22-71967196
Cell: +91
Put line side echo cancelation chip on ur PRI card.
On 25-Jun-2014 10:35 PM, Anurag Rana anuragrana31...@gmail.com wrote:
Hi,
I am using Twinkle to call mobile phone but there is too much noise on the
mobile user's end. Mobile user's voice is echoed back to user. While on
twinkle end
I think your asterisk server is behind firewall or some sort of NAT where
the out to in packets are getting masqueraded with local or DMZ IP of your
firewall / gateway box.
Fix this first to get fail2ban detect the correct public IP.
Otherwise fail2ban will ban your local GW IP due to which you
, Mitul Limbani mi...@enterux.in wrote:
I think your asterisk server is behind firewall or some sort of NAT
where the out to in packets are getting masqueraded with local or DMZ IP
of your firewall / gateway box.
Fix this first to get fail2ban detect the correct public IP.
Otherwise fail2ban
Move the .wav to diff server which has the processor to keep converting
files in runtime.
Asterisk would never have direct file save to mp3 due to patent
restrictions.
And pls Dont hijack the thread of packet filter. Open new email thread !!!
On 01-Jul-2014 9:43 PM, andrew Colin
No way to avoid bw charges for any of the client if it is behind any sort
of NAT.
On 08-Jul-2014 8:52 PM, Sameer Rathod sam...@hostnsoft.com wrote:
Hi Eric,
I am behind nat
Is there any solution for the same.
My goal is to deduct the balance
for the call but free my asterisk server from
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