t; and usually better.
> >
> > --
> > JM or AJS
> >
> > Note: Originating address only accepts e-mail from list! If
> > replying off-
> > list, change address to asterisk1list at earthshod dot co dot uk .
> >
> > --
> >
might be secure,
but the vtiger CRM might have a security hole it. Read the forums for each
package and see if there is any issue.
So being in industry for more than 7 years, If I were you, I will go with
most secure open-source platform and modify GUI part based on my needs.
*Jai Rangi*
Cebod
Very common concerns from new Asterisk, Freeswitch, opensips and freepbx
owners, How can we monitor asterisk, what happens if service stop
responding.
Here is a small howto on monitoring asterisk with nagios. I am sure there
are plenty of options and suggestions, but this is one of them and has
Digital ocean offers ssd on all the virtual machines. Uptime is good.
Jai Rangi
Www.didforsale.com
www.cebodtelecom.com
www.cebod.com
On Mar 8, 2015, at 8:11 AM, Jeff LaCoursiere j...@jeff.net wrote:
Amazon instances are shared resources. I wouldn't want to count on timing or
disk
Agreed, network will be bottleneck even with ssd on shared resource. For a
stable env having a dedicated hosted server will be the best approach and
cheaper too.
Jai Rangi
Www.didforsale.com
www.cebodtelecom.com
www.cebod.com
On Mar 8, 2015, at 9:10 AM, Jeff LaCoursiere j...@jeff.net wrote
IP need to
fail or how many different users IP is trying to register on before
blocking the IP.
Jai Rangi
www.didforslae.com
On Fri, Jun 27, 2014 at 7:37 AM, Anurag Rana anuragrana31...@gmail.com
wrote:
Hi All.
Someone is attacking on my SIP server.
There are lot of requests coming in and I
I can vouch for newfies, but its not asterisk and there is some learning
curve, but comes with lots of features.
-Jai
www.didforsale.com
SIP Trunking Simplified
On Tue, Apr 22, 2014 at 2:54 PM, Nick Cameo sym...@gmail.com wrote:
Hello Everyone,
Thank you all for your response. The people I
Gopal,
This should have been on asterisk-biz list. You can try didofrsale.com. We
can offer your 10,000+ rate centers all under same tier. Contact us offline
to discuss further contact-sa...@didforsale.com
On Wed, Mar 5, 2014 at 10:33 AM, Gopalakrishnan N
gopalakrishnan...@gmail.com wrote:
Hello,
Anyone know good quality text to speach engine for building IVRs for
asterisk. Open-source will be nice, but I wont mind paying for thing really
good.
Regards,
-Jai
--
_
-- Bandwidth and Colocation Provided by
:
http://translate.google.com/translate_tts?tl=enq=i always find google
translate works well
http://translate.google.com/translate_tts?tl=frq=je trouve toujours
google translate fonctionne bien
On Jan 10, 2014 3:17 PM, Jai Rangi jpra...@gmail.com wrote:
Hello,
Anyone know good quality
something like the following in your
/etc/sudoers file:
nagios ALL=(ALL) NOPASSWD:
/path/to/plugins/directory/check_asterisk_channels
You can easily edit this to add more monitoring
Jai Rangi
On Thu, May 9, 2013 at 12:23 PM, motty cruz motty.c...@gmail.com wrote:
Hello,
i'm looking
Guys,
Since I am attached to did for sale:
My apology to every one who received the DIDForSale 2012 Achievement
email and you hated it.
As a asterisk user my question will be.
If some xyz company send you a so called spam email, what made you think
that you should spam the mailing lists. I am
Don,
I have removed yours right away.
Yes, I agree, But just like any company we have purchased/collected email
from different source. Also just like any company we are not perfect, we
make mistakes.
-Jai Rangi
On Wed, Jan 9, 2013 at 5:57 PM, Don Kelly d...@donkelly.biz wrote:
Jai
Jake,
We are DIDForSale support asterisk. We do IP based authentication and do
not require registration. You can test our DIDs without paying anything. I
am sending you the rates just to make it easy to compare apple to apple, no
run around for pricing ;) . Let me know if I can have an opportunity
www.didforslae.com have wide range of products to fit low usage to very
high usage. Dont want to put too much details here. Check it out let me
know if interested, since you are using I will help you waive activation
fee.
-Jai
On Thu, Mar 15, 2012 at 11:30 AM, Guy Gold g...@the-golds.us wrote:
I am sorry. Meant to send to biz list. Thank you for correcting me.
On Tue, Nov 22, 2011 at 5:57 AM, Patrick Lists
asterisk-l...@puzzled.xs4all.nl wrote:
On 11/22/2011 08:14 AM, Jai Rangi wrote:
[removed commercial offer]
You posted to the wrong list. The correct list for commercial
Make money while helping others to enjoy great VoIP Services and huge
savings on inbound SIP Trunking. There is no limit to how many friends and
business partners you can refer. The more friends you refer, the more money
you can make.
Just have your friend send us an email that he was referred by
209.216.2.203 is sip signaling server and 199.173.66.22 is media servers.
BTW Did you try config_1 option. Please send us your configuration and we
will help you configure it properly. Either you can post them here or you
can send them directly to contact-supp...@didforsale.com
Jai
You can use this link too.
http://www.didforsale.com/blog/how-to-setup-your-asterisk-server-with-didforsale
Keep the context as
context=from-trunk.
-Jai
On Wed, Mar 9, 2011 at 5:01 PM, Jai Rangi jpra...@didforsale.com wrote:
209.216.2.203 is sip signaling server and 199.173.66.22 is media
Asterisk security has always been a big concern. I am sure most of asterisk
pros have taken care of these type of attacks. For non pros I am sharing a
shell script here.
http://www.didforsale.com/blog/?p=253
If you care feel free is use it.
-Jai
On Tue, Nov 2, 2010 at 9:27 AM, Cary Fitch
://lists.digium.com/mailman/listinfo/asterisk-users
Jai Rangi
1949 419 7634
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs
I think the key point is how many calls per second. That's what mysql is
concerned about. Other than that it is just asterisk. Did you monitor the
mysql, try log-slow-queries and set the time to 1 second.
-Jai
On Wed, Oct 21, 2009 at 12:57 PM, das sandesh sandesh...@gmail.com wrote:
Hi Steve,
The thing is, concurrent calls won't make any difference, it's the calls
per second.
And really you're unlikely to use too many queries per sec.
Exactly and you can see the slow-log-queries if mysql is taking time.
-Jai
On Wed, Oct 21, 2009 at 3:51 PM, Matt Riddell li...@venturevoip.com
Nagios has a plugin check_sip that can be used for this.
-Jai
On Sat, Oct 17, 2009 at 5:30 PM, Dan Journo
d...@keshercommunications.comwrote:
Hello,
I was wondering if anyone has any insights on the best way to automatically
monitor an asterisk box to check it is constantly available
SIP Status: 183 Session
Progress
4 0.046546 192.168.4.23 - 192.168.3.222 SIP/SDP Status: 200 OK,
On Fri, Sep 4, 2009 at 11:55 PM, Olle E. Johansson o...@edvina.net wrote:
5 sep 2009 kl. 04.58 skrev Jai Rangi:
Hello,
I have a issue between asterisk and windows based VoIP system
But this is my questions why it is sending invites again in 6-10 when the
call is already established.
-Jai
On Sat, Sep 5, 2009 at 3:22 AM, Olle E. Johansson o...@edvina.net wrote:
5 sep 2009 kl. 09.06 skrev Jai Rangi:
Thank you for your response,
But we do get response from client
Hello,
I have a issue between asterisk and windows based VoIP system (Client).
Vendor SIP Server -- My asterisk -- Client
Here is ethereal trace between asterisk and client.
1 0.00 192.168.3.222 - 192.168.4.23 SIP/SDP Request: INVITE
sip:1978525...@192.168.4.23
*All,
To meet the target for the month, we are running a special promotion.
$5 activation fee waived for all new DID purchases.*
Buy DIDs from DIDForSale http://www.didforsale.com/ today and *your $5
activation fees will be WAIVED* for all the DIDs purchased before July 20
2009.
There is no
Sincere Apologies--
Send the mail to wrong list, Meant to send to asterisk-biz list.
-J
On Wed, Jul 8, 2009 at 11:35 PM, Jai Rangi jpra...@gmail.com wrote:
*All,
To meet the target for the month, we are running a special promotion.
$5 activation fee waived for all new DID purchases.*
Buy
Yes, its working :)
Jai Rangi
ww.didforsale.com
On Thu, Apr 30, 2009 at 12:12 PM, James A. Shigley
j...@answeringserv.comwrote:
Had an inbound email server issue, just double checking it is working
again.
James Shigley
*Monroe Telephone Answering Service*
409-981-9213**
Infinity
Vikas,
www.didforsale.com can get you the DIDs, please contact me off list.
Jai Rangi
jpra...@didforsale.com
On Wed, Feb 25, 2009 at 1:35 PM, Vikas topg...@gmail.com wrote:
Since it's not clear from this thread of conversation, do you need 100
unique DIDs?
I apologize for not being more
such a project,
just so long as we have enough contributors.
**
We have some documentation and I can contribute that. Also we can provide
the physical resources (Domain, Web hosting, bandwidth, storage, database
etc). Ofcourse need a team with designated responsibilities.
-Jai Rangi
Shane,
You can try, www.didforsale.com. We allow free testing with no purchase
required. See what others are saying, http://www.didforsale.com/blog/?p=103
-Jai
On Mon, Jan 26, 2009 at 11:12 AM, Thomas Mullins
tsmull...@wise.k12.va.uswrote:
My coworker and I have built an Asterisk box.
ngrep port 5060
or tcpdum port 5060
By default asterisk runs on port 5060, that way you can see if your getting
the signal or not.
Jai Rangi
Buy SIP DID www.didforsale.com
free Trial now purchase required
On Tue, Jan 13, 2009 at 1:13 PM, David @ULC ucoms2...@gmail.com wrote:
I also tried
.i386.rpm
Is you sip configuration right?
cant tell without looking at it.
Jai Rangi
Buy SIP DID www.didforsale.com
free Trial no purchase required
On Tue, Jan 13, 2009 at 1:44 PM, David @ULC ucoms2...@gmail.com wrote:
[r...@vicidialnow ~]# ngrep port 5060
-bash: ngrep: command not found
[r
Alex,
I must say wow, great explanation. It was a wonderful reading.
Best,
-Jai
On Tue, Jan 13, 2009 at 1:49 AM, Alex Balashov abalas...@evaristesys.comwrote:
Hi Randulo,
I think this topic is probably more appropriate for asterisk-biz, as was
the aforementioned rant about one particular
Increase the timeout in my.cnf in mysql.
-Jai
Buy unmetered VoIP DIDs www.didforsale.com Free Trail
On Mon, Nov 24, 2008 at 11:10 PM, Nhadie [EMAIL PROTECTED] wrote:
Hi,
Need help on mysql cdr, i keep on seeing this log on the console.
but my db is up and i see the calls being logged on
Good work, I am sure this will be endorsed by many and will be useful for
lots of small VoIP user who are ready to expand. Only problem I have seen is
that people who have done (deployed) this type of integration does not share
complete solution mainly because of compititive disadvantage. But
SIP-only accounting is good enough most of the time.
Does not work in production environment. Specially when you are charging per
second or per minute.
Works only if some one is offering unmetered only service or just doing it
for fun. If it metered service like calling cards, termination or
works and
does not work in production environments before implementing for
large-scale billing solutions that are perfectly functional, and indeed,
very much in production.
No Need to be so contemptuous.
On Wed, Oct 29, 2008 at 4:37 PM, Alex Balashov [EMAIL PROTECTED]wrote:
Jai Rangi wrote:
SIP
Hello Dave,
We can offer you. What area DID you are looking for.
Jai
Buy SIP DID, www.didforsale.com
On Thu, Oct 23, 2008 at 2:20 PM, David Troy [EMAIL PROTECTED] wrote:
Hey folks,
I am involved with a group that is going to use Twitter, SMS, iPhone, and
Asterisk to field-monitor the
John,
Client Behind a NAT should not be problem. What are your issues? If you post
your scenario and more details about your problem only then some can help
you better.
Jai
Buy SIP DID at www.didforsale.com
On Wed, Oct 22, 2008 at 12:24 AM, Johanna NIRINA [EMAIL PROTECTED]wrote:
hi there,
I
Check the permissions for the directory.
Jai
http://www.didforsale.com
.
On Sun, Oct 19, 2008 at 1:19 PM, Ahmed Torintino [EMAIL PROTECTED]wrote:
i have done that as follow
[EMAIL PROTECTED] asterisk]# service asterisk start
Starting asterisk: [
All,
I am having audio quality problem in some calls (1-2%) on asterisk. I
captured RTP traffic using ethereal and this is what I found with the
problematic calls. (Worst cases)
Drop by Jitter buff: 25-75%
Out of Seq: 50-100% (100 % means very very poor call quality).
Has anyone had similar
Good
Alex Balashov wrote:
Jai Rangi wrote:
All,
I am having audio quality problem in some calls (1-2%) on asterisk. I
captured RTP traffic using ethereal and this is what I found with the
problematic calls. (Worst cases)
Drop by Jitter buff: 25-75%
Out of Seq: 50-100% (100
Oh yes, how could I forgot about that?
Thank you,
-Jai
On Fri, Oct 3, 2008 at 1:08 PM, Alex Balashov [EMAIL PROTECTED]wrote:
sipp can simulate RTP traffic.
Jai Rangi wrote:
Al and Alex,
Thank you for your input,
Sorry TDM is not the option at this time :( .
Everything has been
) was completely seamless. Did not had any down time, there
was just a pause for just 1 second in the audio. I was very impressed.
-Jai
On Fri, Oct 3, 2008 at 1:21 PM, Jai Rangi [EMAIL PROTECTED] wrote:
Oh yes, how could I forgot about that?
Thank you,
-Jai
On Fri, Oct 3, 2008 at 1:08 PM
For asterisk you can use heartbeat. regarding T1, you will need some thing
out outside Asterisk server.
Any reason you want to go for T1, not true VoIP?
Jai
http://www.didforsale.com/
*Buy SIP DIDs all Over US at low cost, unlimited minutes
http://www.didforsale.com;
On Wed, Oct 1, 2008 at 2:24
connected to a Redfone device (TDMoE).
Thanks,
Steve Totaro
On Wed, Oct 1, 2008 at 5:40 PM, Jai Rangi [EMAIL PROTECTED] wrote:
For asterisk you can use heartbeat. regarding T1, you will need some thing
out outside Asterisk server.
Any reason you want to go for T1, not true VoIP?
Jai
http
Are you looking for inbound or outbound.
I can get you free inbound test DID. LMK
Jai
www.didforesale.com
On 9/27/08, Sam Tam [EMAIL PROTECTED] wrote:
Hello everyone
I am trying to look for a free test call generator that will get me some
stats like PDD, ASR and call quality etc on each
We are using few dell 1950, it been two year and never had any issue,
Jai
www.didforsale.com
*Buy SIP DIDs all Over US at low cost, unlimited minutes
http://www.didforsale.com;
On Thu, Sep 25, 2008 at 3:19 PM, Alex Balashov [EMAIL PROTECTED]wrote:
Philipp Kempgen wrote:
Jon Weisman schrieb:
Hitesh,
If you dont have experience with Linux I would recommend you to use Trixbox,
that will come with all the required packages and will do everythign for
you.
Re: FXO and FXS, you don't need to buy any card for True VoIP. Now you can
buy DIDs that can come to your asterisk over the internet.
enable me to call
out using the PSTN line at my home in India from Canada?
Thanks.
Best REgards,
Hitesh
On Fri, Sep 19, 2008 at 10:33 AM, Jai Rangi [EMAIL PROTECTED] wrote:
Hitesh,
If you dont have experience with Linux I would recommend you to use
Trixbox,
that will come with all
Another idea can be have the customers to opt-in for auto-refill if they
want to use multiple call feature. Usually this does not have be a high
number, just autorefill the account if the balance goes down $1.
Jai
www.didforsale.com
*Buy DID at low cost http://www.didforsale.com;
On Thu, Sep 18,
We also have the similar setup, 2 ser server with heartbeat doing the load
balance and 4 asterisk servers handling the media. Of course the data is on
MySQL Cluster.
Jai Rangi
www.bingotelecom.com
On Mon, Jul 21, 2008 at 5:13 PM, Edgar Guadamuz [EMAIL PROTECTED] wrote:
I have used
Has anyone used or thought of using Asterisk server farm behind LVS.
-Jai
___
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
Vivek,
What do you need, DID or Termination?
BTW We are in California. Send me you Contact info and we can discuss more
about your needs.
-Jai
On Thu, Mar 6, 2008 at 10:25 AM, Vivek Shrivastava [EMAIL PROTECTED]
wrote:
Hi,
I would like to seek an opinion or list of providers in USA or
smooth as silk for me.
*From:* Jai Rangi [mailto:[EMAIL PROTECTED]
*Sent:* Tuesday, December 11, 2007 2:15 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] Most Stable version of Asterisk
Hello,
I tried to install the asterisk 1.4.15 and I am
Anyone,
could you please suggest the latest stable release for asterisk.
-Jai
On Dec 10, 2007 9:08 PM, Jai Rangi [EMAIL PROTECTED] wrote:
I am planning to upgrade my asterisk to
Asterisk
1.4.15http://www.digium.com/elqNow/elqRedir.htm?ref=http://downloads.digium.com/pub/asterisk/releases
Hello,
I tried to install the asterisk 1.4.15 and I am not able to start it. I get
the segmentation fault error. What might be wrong, where I can look for a
clue.
Also could some one PLEASE suggest the most stable version of asterisk.
-Jai
___
And that version name/number is ???
:)
-Jai
On Dec 11, 2007 4:17 PM, C F [EMAIL PROTECTED] wrote:
In my experience the most stable asterisk is the one that runs and
runs and never crashes.
On 12/11/07, Jai Rangi [EMAIL PROTECTED] wrote:
Hello,
I tried to install the asterisk 1.4.15
Hello,
Since last few days I have noticed some people complaining that their call
gets disconnected while they are in the middle of the conversations. Looking
in the log I found these error messages,
Dec 10 11:18:56 DEBUG[8833] channel.c: Bridge stops bridging channels
SIP/5060-b7a03560 and
-0800, Jai Rangi wrote:
Is this the right place to post this error message and expect for the
solution.
I am using asterisk-1.2.12 on FC5. I will appreciate if someone can give
me some hints to get rid of this problem.
I doubt you'll get much response, unless you try again with a newer
have customized few components and don't want to do that again).
My Current asterisk is configured with MySql. Is there any change in the
asterisk tables and databases structures.
I will appreciate any feedback.
Thank you,
-Jai
On Dec 10, 2007 4:27 PM, Jai Rangi [EMAIL PROTECTED] wrote:
Thank
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