On 8/3/07, Douglas Garstang <[EMAIL PROTECTED]> wrote:
>
> What techniques do people use for checking ITSP (IP->SIP) carrier
> reliability?
>
> There's different categories of realiability.
>
> The first would be call completion rates. It's fairly easy to check for
> timeouts, bad SIP response cod
On 8/4/07, Douglas Garstang <[EMAIL PROTECTED]> wrote:
>
>
> I need something that can LCR not based just on cost, but also on jitter
> and latency. :)
Try ASTPP, also in Perl. There is work underway in making it modular and
classes based by next release. Moreover there is talk to use the same
un
Steve Totaro can you tell me if if you used Asterisk to do this? what do you
recommend?
On 8/3/07, Steve Totaro <[EMAIL PROTECTED]> wrote:
>
> Why do you say this? I am just curious. I have done countless installs
> where a server was used strictly as ISDN<-> media gateway. Never had a
> proble
>-Original Message-
>From: [EMAIL PROTECTED] on behalf of Jean-Michel Hiver
>Sent: Fri 8/3/2007 4:45 PM
>To: Commercial and Business-Oriented Asterisk Discussion
>Subject: Re: [asterisk-biz] Least Cost Routing Rates
>
>Le Sat, 04 Aug 2007 02:42:33 +0400, Douglas Garstang
><[EMAIL PROTECT
>-Original Message-
>From: [EMAIL PROTECTED] on behalf of Alex Balashov
>Sent: Fri 8/3/2007 4:44 PM
>To: Commercial and Business-Oriented Asterisk Discussion
>Subject: Re: [asterisk-biz] Least Cost Routing Rates
>
>On Fri, 3 Aug 2007, Douglas Garstang wrote:
>
>> Assuming you used several
Why do you say this? I am just curious. I have done countless installs
where a server was used strictly as ISDN<-> media gateway. Never had a
problem with a single one of them. They are the strongest part of the
large scale systems I have built.
You do realize that the Cisco box is essentia
Because SIP termination service is unreliable unless you and your
provider control all of the variables.
I guess you could use call waiting and three way but that would be the
only way to have multiple calls on a single line. If you go with BRI, I
do not believe that Digium makes those cards.
Thanks for your reply
so there is no reliable way of doing this with OpenSER and Asterisk?
On 8/3/07, Alex Balashov <[EMAIL PROTECTED]> wrote:
>
>
> If carrier-grade reliability is such a central concern, chances are
> you don't want to be terminating T1s into PC hardware anyway, and should
> con
If carrier-grade reliability is such a central concern, chances are
you don't want to be terminating T1s into PC hardware anyway, and should
consider investing in a Cisco or similar media gateway that can handle the
ISDN<->SIP conversion for you.
--
Alex Balashov
Evariste Systems
Web: http:/
Well. With call waiting... :-)
On Fri, 3 Aug 2007, Arya wrote:
> Hello
>
> Before I ask my question about the Digium cards, I would like to know if
> there is any way of having more than 1 concurrent call on a PSTN line, Is
> that possible? or would I need a BRI and Digium card is needed to do
Le Sat, 04 Aug 2007 04:53:21 +0400, Arya <[EMAIL PROTECTED]> a écrit:
> Hello
>
> Before I ask my question about the Digium cards, I would like to know if
> there is any way of having more than 1 concurrent call on a PSTN line, Is
> that possible? or would I need a BRI and Digium card is needed to
why dont u just buy a sip termination service?
u can then have unlimited out going channels
Date sent: Fri, 3 Aug 2007 20:53:21 -0400
From: Arya <[EMAIL PROTECTED]>
To: "Commercial and Business-Oriented Asterisk Discussion"
Subject:[asterisk-biz] SIP to PSTN Hardware
S
Hello
Before I ask my question about the Digium cards, I would like to know if
there is any way of having more than 1 concurrent call on a PSTN line, Is
that possible? or would I need a BRI and Digium card is needed to do that?
--
Thank You
___
--Bandw
Le Sat, 04 Aug 2007 02:42:33 +0400, Douglas Garstang
<[EMAIL PROTECTED]> a écrit:
> Lots of questions from me today.
>
>
> Assuming you used several ITSP's, and you wanted to do least cost
> routing, does anyone have any suggestions as to the best way to manage
> all the routes and their costs?
On Fri, 3 Aug 2007, Douglas Garstang wrote:
> Assuming you used several ITSP's, and you wanted to do least cost
> routing, does anyone have any suggestions as to the best way to manage
> all the routes and their costs?
There are a number of home-brewed and enterprise-strength LCR and
genera
On Fri, 3 Aug 2007, Douglas Garstang wrote:
> Lots of questions from me today.
>
>
>
> Assuming you used several ITSP's, and you wanted to do least cost
> routing, does anyone have any suggestions as to the best way to manage
> all the routes and their costs?
>
> It would seem like it could e
You can try them on Phone2net.com
try the FREE TRIAL first
Date sent: Fri, 3 Aug 2007 18:10:31 -0400
From: "Andrew Joakimsen" <[EMAIL PROTECTED]>
To: "Commercial and Business-Oriented Asterisk Discussion"
Subject:Re: [ast
Lots of questions from me today.
Assuming you used several ITSP's, and you wanted to do least cost
routing, does anyone have any suggestions as to the best way to manage
all the routes and their costs?
It would seem like it could easily be a full time job for someone.
Would it be best t
On 8/4/07, Andrew Joakimsen <[EMAIL PROTECTED]> wrote:
> On 8/3/07, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
> > I am looking for a DID carrier that supports passing to me any
> > rdnis/call forwarding details
>
> Likewise.
>
> ___
> --Bandwidth and C
On 8/3/07, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
> I am looking for a DID carrier that supports passing to me any
> rdnis/call forwarding details
Likewise.
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-biz mail
On Fri, 3 Aug 2007, Douglas Garstang wrote:
> Ok, so I haven't spoken to them yet, but it looks like their product
> doesn't allow you to store QoS information realtime with an API to act
> upon it. Pretty graphs etc aren't much use.
I would speak to them. I've seen the whole caboodle demoe
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-biz-
> [EMAIL PROTECTED] On Behalf Of Alex Balashov
> Sent: Friday, August 03, 2007 10:54 AM
> To: Commercial and Business-Oriented Asterisk Discussion
> Subject: Re: [asterisk-biz] Checking Carrier Reliability?
>
> On Fri, 3
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-biz-
> [EMAIL PROTECTED] On Behalf Of Alex Pilosov
> Sent: Friday, August 03, 2007 10:36 AM
> To: Commercial and Business-Oriented Asterisk Discussion
> Subject: Re: [asterisk-biz] Checking Carrier Reliability?
>
> On Fri, 3 A
Indeed!
More than one good & reliable network operator has been shoot down because
they have an HR department that hire morons...
/Danny
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alex Pilosov
Sent: den 3 augusti 2007 19:36
To: Commercial and Busines
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-biz-
> [EMAIL PROTECTED] On Behalf Of Jared Smith
> Sent: Friday, August 03, 2007 10:52 AM
> To: Commercial and Business-Oriented Asterisk Discussion
> Subject: Re: [asterisk-biz] Checking Carrier Reliability?
>
> On Fri, 2007
> On Fri, 3 Aug 2007, Steve Totaro wrote:
> I see a few really annoying things with this:
>
> 1) How do you know what calls are going to what mobile network for
your
> routing?
>
> 2) Is it possible to set outgoing Caller*ID on a GSM card? If not, all
> calls will show up as the phone number for
Are you looking for a specific application or information in general?
There are a number of ways to interconnect depending on your usage.
--
Eric Chamberlain, CISSP
Chief Technical Officer
Voxilla - http://voxilla.com/
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
On Fri, 3 Aug 2007, Steve Totaro wrote:
> Unlimited plans. Family plans with many shared minutes (base charge plus
> $10/mo per extra phone or SIM).
I see a few really annoying things with this:
1) How do you know what calls are going to what mobile network for your
routing?
2) Is it possible
Unlimited plans. Family plans with many shared minutes (base charge plus
$10/mo per extra phone or SIM).
When I was in war torn West Africa, you could not get a land line in any
way shape or form. I met with the Minister of Telecom for Liberia and he
had a phone on his desk which ironically did
Hi,
> Does anyone have a good price for this card ? We are based in the UK.
We can do 1652,26 EUR plus 26,20 EUR shipping costs, for 3-4 day delivery.
We can also do 1652,26 EUR plus 43,88 EUR shipping costs, for 2 day
delivery.
Regards,
--
Sérgio Araújo
3GNTW - Tecnologias de Informação, Lda
On Fri, Aug 03, 2007 at 01:52:52PM -0400, Alex Balashov wrote:
> If you send enough traffic to a mobile carrier to warrant it, you might be
> able to privately interconnect with them via TDM. Of course, they can be
> very picky about this and generally will thumb their nose at you and tell
> you
On Fri, 2007-08-03 at 10:22 -0700, Douglas Garstang wrote:
> What about dead air? How do you check for dead air?
>
> What about quality?
If we're talking SIP traffic, one suggestion would be to look at the
RTCP traffic coming back to you from the carrier, and see what type of
jitter/packet loss/
If you send enough traffic to a mobile carrier to warrant it, you might be
able to privately interconnect with them via TDM. Of course, they can be
very picky about this and generally will thumb their nose at you and tell
you to go through some ILEC's tandem unless you're Big and Important
enoug
On Fri, 3 Aug 2007, Douglas Garstang wrote:
> What about dead air? How do you check for dead air?
>
> What about quality?
>
> What tools exist to proactively monitor call quality in real time?
To the extent that any of this is possible to do on a media level,
talk to Brix Networks (http://www.
On Fri, 3 Aug 2007, Douglas Garstang wrote:
> What techniques do people use for checking ITSP (IP->SIP) carrier
> reliability?
I use retards test.
If carrier's sales person sounds like a retard, I don't have to evaluate
their network reliability. Similarly, if I ask them a technical question
an
What techniques do people use for checking ITSP (IP->SIP) carrier
reliability?
There's different categories of realiability.
The first would be call completion rates. It's fairly easy to check for
timeouts, bad SIP response codes etc.
What about dead air? How do you check for dead air?
I was reading that if you want to terminate to cell phones, that sending
the calls through a GSM (or whatever technology is locally available)
gateway, is cheaper than sending the calls to the PSTN via T1/E1,
Analog.
How is that?
I'm not sure how these devices work anyway.
Does the gateway
Le Fri, 03 Aug 2007 09:59:42 +0400, Mitul Limbani <[EMAIL PROTECTED]> a
écrit:
> Jean,
>
> Quoting Jean-Michel Hiver <[EMAIL PROTECTED]>:
>
>> Hi List,
>>
>> I have a special on Reunion Island Proper (prefix +262 262) at 0.01 EUR
>> /
>> minute, 1/1 billing, capacity 2 E1, 30/30 billing terms.
I just tried to call in after creating an account.
After the call connects, enter the show id: 22622# and your_PIN#
I dial in and am asked for the podcast id, I enter 22622# and am told
that my passcode is not correct. I also tried just entering my passcode
but got the same error message.
What
Hi folks,
The August 3 edition of our Friday conference call and podcast kicks
off in just over and hour. I know the list isn't delivering properly
but if a few people get this it'll be better than none.
We are going to be talking today about TDM inside and outside the box.
I own some antiiquate
I am looking for a DID carrier that supports passing to me any
rdnis/call forwarding details passed from my Centrex DMS-100 to them
back to me via standard sip headers.
This is needed to be able to properly implement a sip trunked voice mail
solution for my centrex dms-100. Thanks!
Thanks!
_
Priceless.
C F wrote:
Oh, thanks for clearifying, BTW, since it's a business list, he should
get someone with a better Enlish to post his proposals.
smime.p7s
Description: S/MIME Cryptographic Signature
___
--Bandwidth and Colocation Provided by
42 matches
Mail list logo
