- "Steve Edwards" wrote:
> > - "Steve Edwards" wrote:
> >>
> >> 42[:1]
> >>
> >> (The fact that you ask such a generic question implies you have a
> high
> >> probability of failure. You should hire somebody with a bit more
> clue
> >> and learn from them.)
>
> On Thu, 15 Oct 2009, Tim
>> On Thu, 15 Oct 2009, Shahnawaz Mir wrote:
>>
>>> I am planning to deploy an Asterisk PBX for 100-200 users. I am not
>>> sure about PSTN incoming/outgoing line ratio for SIP users. I mean if
>>> you recall dial up internet the common line ratio is 1:10 (one line
>>> for 10 users on access ser
Thanks Tim,
Your response is really helpful. Its not going to be very busy. I was
expecting 10:1 but I will start some where between 4-10. Thank you
very much.
Regards
Shahnawaz Mir
On 15-Oct-09, at 11:11 AM, Tim Nelson wrote:
> - "Steve Edwards" wrote:
>> On Thu, 15 Oct 2009, Shahnaw
Shahnawaz Mir wrote:
> Hi,
>
> I am planning to deploy an Asterisk PBX for 100-200 users. I am not
> sure about PSTN incoming/outgoing line ratio for SIP users. I mean if
> you recall dial up internet the common line ratio is 1:10 (one line
> for 10 users on access server or an E1 for 300 use
- "Steve Edwards" wrote:
> On Thu, 15 Oct 2009, Shahnawaz Mir wrote:
>
> > I am planning to deploy an Asterisk PBX for 100-200 users. I am not
> sure
> > about PSTN incoming/outgoing line ratio for SIP users. I mean if you
>
> > recall dial up internet the common line ratio is 1:10 (one lin
On Thu, 15 Oct 2009, Shahnawaz Mir wrote:
> I am planning to deploy an Asterisk PBX for 100-200 users. I am not sure
> about PSTN incoming/outgoing line ratio for SIP users. I mean if you
> recall dial up internet the common line ratio is 1:10 (one line for 10
> users on access server or an E1
Shahnawaz Mir wrote:
> I am planning to deploy an Asterisk PBX for 100-200 users. I am not
> sure about PSTN incoming/outgoing line ratio for SIP users. I mean if
> you recall dial up internet the common line ratio is 1:10 (one line
> for 10 users on access server or an E1 for 300 users). Can
Hi,
I am planning to deploy an Asterisk PBX for 100-200 users. I am not
sure about PSTN incoming/outgoing line ratio for SIP users. I mean if
you recall dial up internet the common line ratio is 1:10 (one line
for 10 users on access server or an E1 for 300 users). Can somebody
tell me what
The Cisco's also support T.38 gateway functions whereas Asterisk can
only do pass thru. Either way you'll still need another server,
typically hylafax, to receive the faxes to get them somewhere useful.
In my experience the Cisco switches are definitely the way to go for
the ISDN/SIP gateway and t
Hi,
Our requirement is just to be able to do voice and fax at a quality manner.
What is the difference between using a physical server vs a PCI card that
plugs in
to the Asterisk server? Is there a big difference in terms of scalability?
We are looking at a solution that can be easy-to-deploy ou
Rhino or audiocode PSTN gateway
- Original Message -
From: "mark morreny"
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Subject: [asterisk-users] PSTN to SIP
Date: Thu, 17 Apr 2008 01:25:49 +0800
Dear all, A quick question on deploying Ast
If your requirements are simple and you only have a small number if E1s,
you can also use a Cisco 36xx with a T1/PRI card. 3600's have limited
capacity but we run 4 PRIs on a 3640 no problem and it's been very stable
for several years. The nice thing about 3600's is they are almost free,
although
Dear all,
A quick question on deploying Asterisk over E1. I am looking for a low-cost
solution for bridging my E1 line and Asterisk with reasonable stability
suppoing both voice and fax. Will a Digium T100 be good for that or I
really need a Cisco AS 5400 for this task? What is the difference b
Hi all,
I've tried everything over past month but no success. When somebody
calls in from a PSTN line or cell phone to my asterisk box, which is a
connected to a SIP provider, MoH doesn't work good, it is very choppy.
I've tried native format, silence suppression, ulaw and gsm music files,
changin
On Fri, Jul 15, 2005 at 08:23:14AM +0200, Florian Overkamp wrote:
> Try:
>
> exten => _X.,1,Dial(SIP/[EMAIL PROTECTED])
>
> The '.' is a wildcard match of unknown length. With your pattern you only
> accept extensions of 1 digit long.
Perfect! Thank you!
Hmm...it appears it's not receiving AN
Hi,
> -Original Message-
> So far I've gotten Asterisk to say:
> -- Extension 'XX' in context 'pstn' from '' does not
> exist. Rejecting call on channel 0/23, span 1
> (where XX is the phone number I dialed)
> So, that's a start, I guess ;)
> extensio
On Fri, Jul 15, 2005 at 01:28:45AM -0400, Jose Raborg wrote:
> Do you want to route the calls depending on the caller id? Or Do you
> want to assign a DID to a SIP?
The remote SIP device will route the calls appropriately based on the
information sent to them (the "*ANI*DNIS*" sent as an "extensio
: [Asterisk-Users] PSTN to SIP gateway
I've been looking through the examples and docs, but haven't yet quite
figured out how to do what I want. We've got a T1 coming in carrying a
block of telephone numbers, terminating on an Asterisk box. Any call
that comes in needs to get sent
I've been looking through the examples and docs, but haven't yet quite
figured out how to do what I want.
We've got a T1 coming in carrying a block of telephone numbers,
terminating on an Asterisk box. Any call that comes in needs to get
sent to a SIP proxy, with a particular extension format:
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