Hello Lee,
Telekom Malaysia provide PRI lines. We've been actively using their services
for the past few years with success. Let me know if you need contacts.
Regards,
Arstan
On Thu, Jan 20, 2011 at 9:56 AM, Lee, John (Sydney)
john@compuware.comwrote:
We are setting up an office in
.
From: asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] On Behalf Of Arstan Jusupov
Sent: Thursday, 20 January 2011 1:33 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users
That's quite possible. We handle around 100 similtaneous calls(PRI +
SIP) with a decent dell server with only 4gb ram.
On Wed, Feb 2, 2011 at 6:22 AM, Juan David Diaz juanch...@gmail.com wrote:
Hi Asterisk Users,
I would like to handle about 250 simultaneous (calls agents only) calls
with PRI
Hi William,
just to know that gtalk/asterisk works in your environment you could
quickly create a virtual server and install an asterisk 1.8 with this
guide
http://highsecurity.blogspot.com/2010/11/googlevoice-asterisk-18-with-freepbx.html
which works fine for me.
this way you know for sure that
I highly recommend Yealink phones.
On Wed, Mar 9, 2011 at 7:01 PM, Raj Mathur r...@linux-delhi.org wrote:
Hi,
Would you recommend some standalone SIP phones that work well with
Asterisk? Personal experience preferred.
Thanks,
-- Raj
--
It may sound silly but did you configure/open firewall ports on amazon ec2? The
instance itself as we as from the amazon ec2 panel?
Sent from my iPhone
On Mar 10, 2012, at 7:16 AM, sean darcy seandar...@gmail.com wrote:
On 03/09/2012 04:16 PM, sean darcy wrote:
I'm trying to move the
iPhone
On Mar 10, 2012, at 10:20 AM, sean darcy seandar...@gmail.com wrote:
On 03/09/2012 07:20 PM, Arstan Jusupov wrote:
It may sound silly but did you configure/open firewall ports on amazon ec2?
The instance itself as we as from the amazon ec2 panel?
Sent from my iPhone
On Mar 10, 2012
Thanks, I will try asterisk 1.8 tomorrow and see.
Sent from my iPhone
On Mar 13, 2012, at 8:24 PM, Kevin P. Fleming kpflem...@digium.com wrote:
On 03/13/2012 07:19 AM, Arstan Jusupov wrote:
So since remote UPDATE is not supported, this project of mine would fail. Is
that correct?
If your
I understand you want to choose the easy way but I really think you should not
be lazy and go phone by phone and write down the Mac address. Of course if
that's ever possible...
For future ease of administering those phones, like if you want to do
provisioning, troubleshooting etc etc. Better
You can take a look at phpivr project -
https://sites.google.com/site/grygoriim/devel/phpivr
Sent from my iPhone
On Mar 29, 2012, at 8:49 PM, Eric Wieling ewiel...@nyigc.com wrote:
core show application saydigits
core show application SayUnixTime
Or better yet core show applications
Sent from my iPhone
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users
First of all, I want apologize for the first two blank emails that I sent out
by mistake.
I have Xorcom USB fxo channel bank, asterisk 1.6, freepbx 2.8. Up to now, the
lines connected from Telekom did not have caller id feature enabled, now that
we enabled we cannot see incoming caller id
Another option is to get those routers that are capable of running dd-wrt
firmware with USB ports(for storage)
This option is rather good if you don't need any VoIP cards and if you are OK
to use sip/iax2 etc trunks.
I have my wifi router with dd-wrt firmware running asterisk for home use.
I think there is Google calendar with public holidays listings for nearly every
country. At least I know there is one for Malaysia. And Google calendars are
available through number of ways I suppose.
Sent from my iPhone
On May 19, 2012, at 12:35 AM, Ing CIP. Alejandro Celi
tcpdump and wireshark would help I guess. Just sniff for sip traffic and look
out for what's happening there. My 2 cents
Sent from my iPhone
On May 19, 2012, at 8:33 PM, David Wessell da...@ringfree.biz wrote:
I'm in the process of setting up an asterisk box that will stand
between PBX's and
I am sending and receiving fax.
I have an issue where sending and receiving is intermittent. Provider is
claiming that It doesn't always receives t.38.
So I thought if I could see if Asterisk is sending and receiving t.38 as it
should be.
Oh yeah, I am using ATA with t.38 support which is
Thanks Kevin,
updtl debug is what I am looking for, I guess.
Arstan
Sent from my iPhone
On May 24, 2012, at 11:25 PM, Kevin P. Fleming kpflem...@digium.com wrote:
On 05/24/2012 10:19 AM, Arstan Jusupov wrote:
I am sending and receiving fax.
I have an issue where sending and receiving
Why don't you use AMI? There's are phpami project if you google.
Sent from my iPhone
On May 25, 2012, at 1:51 AM, Kamlesh Kumar kamlesh_...@hotmail.com wrote:
Hi,
I'm using AMI to get the extension status but always get -1 i.e. extension
not found.
#!/usr/bin/php -q
?php
I think what you want is CEL logging, cdr has design issues. If I am not
mistaken it's covered in asterisk wiki itself.
Sent from my iPhone
On May 29, 2012, at 8:57 PM, Marek Cervenka cerv...@fpf.slu.cz wrote:
hi,
i read a lot about CDR problems
this document is the best description of
I highly recommend Yealink phones. They have variety of choices - from basic to
video phones. Easy to configure manually via web ui and also supports auto
provisioning.
Price wise quite affordable. We currently deploy those phones in our asterisk
projects. From office Pbx to call centers.
Why don't you just generate call files for each of the servers on the same
server? Anyhow you are not sharing one single pool of call files among servers,
I suspect that's where network drive would come in handy.
Sent from my iPhone
On Jul 6, 2012, at 6:56 PM, Chandrakant Solanki
You have to use from-zaptel in your context, and define your Zap DIDs in
elastix. And lastly set inbound route with your zap did defined and forward
that to your ivr.
Sent from my iPhone
On Jul 20, 2012, at 7:30 AM, Satria Anamarta anam.satri...@gmail.com wrote:
Hi,
Let say I have 8 PSTN
.
Thanks,
Regards,
Arstan Jusupov
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk
Correct me if I'm wrong but phono works with voxeo tropo.
Sent from my iPhone
On Aug 8, 2012, at 7:24 AM, Matt Riddell li...@venturevoip.com wrote:
On 2/08/2012, at 2:27 PM, Arstan Jusupov arst...@gmail.com wrote:
Dear list,
I am looking for an open source SIP client(or any SDK) that can
24 matches
Mail list logo