* You are viewing Posts Tagged ‘DIDs’

Common/Reasonable Assumption on DID/Channel over-subscription

Hello All,

just throwing this out there. What are people generally using these days
when designing their services, esp. those that require a user to call a DID
to access their system, similar to calling card services. There was a time
when this used to be 50 to 1 for DIDs, and about 10 to 1 for number of
channels bought in SMB with IP-PBX.

I believe this would have changed today and assuming a service is pretty
popular, the ALOCs are longer due to cheaper rates and convenience of
calling. Does anyone have any real world numbers they can share? Is 10 to 1
a good ratio to ensure a user practically never gets a “circuits are busy”?

Thanks in advance

Set Caller Number in E1 PRI ISDN Lines

Hi,

I am having an E1 PRI ISDN Lines with 30 bearer channels and 1 D Channel
with hundred DIDs (Direct Inward Dialing) numbers attached to a Sangoma PRI
Card on the server,
I am using asterisk 1.8.5 on CentOS 5.6.

How can i configure DIDs so that if i make an outgoing call the DID number
should go to the caller not the pilot number

For example

PRI Numbers Range -> 31303000 – 31303099
Pilot Number -> 31303000

So if i need to set caller number as 31303008 for example and not as
31303000, is there a way to set this in dial plan (extensions.conf)

Please guide and let me know if anyone needs more information and have
questions.

Regards,

Kaushal

looking for free DID 708-839

SipGate.Com used to offer free DIDs in Illinois, but it looks like they
ran out of numbers…

-Vladimir

On 9/23/2011 2:45 PM, Joseph wrote:
> Are there any free DID in Illinois 708-839 or area?
>

DID to display the calling number

Hi All;

The main number is 56725000 and we have DIDs from 5000 to 5999. Now, I need that if five IP Phones make outside calls, then destination should see only 56725111 so I beleive it is related to the DID 5111 but I do not know what I have to do a settings for this DID and where, so I can acheive this.

Any advise?

Regards
Bilal

ITSP failover for PRI

Hi,

I still have the same problem trying to configure ITSP failover in
extensions.conf for a connected PRI. Any comments thoughts or direction
would be greatly appreciated.

I sympathize with wanting inbound DID failover. If we have a client with
multiple DIDs we will spread them across two or three ITSPs so that all
inbound connectivity will not be lost if one of them has an issue.

I have a little experience with using SS7 from when we set up multiple call
centers in Norway for Telenor. Using SS7 we were able to determine incoming
call credentials, then sending the call the proper switch/CSR based upon the
number dialed and where the caller was located. The call was not actually
connected until after it was routed to the proper destination. This still
would not have dealt with the originator not supplying inbound service.

We’re using an Asterisk based SIP-T1 trunking gateway and would like to
implement failover between two ITSPs.

If we connect a soft phone to the gateway with the following lines in
extensions.conf failover works.

If one ITSP is unavailable the call flow cascades to the second ITSP and
connects with audio.

[outgoing]

exten => _1NXXNXXXXXX,1,NoOp(${CALLERID(all)=”” <>}) exten =>

_1NXXNXXXXXX,2,Dial(SIP/${EXTEN}@ITSP1)

exten => _1NXXNXXXXXX,3,Dial(SIP/${EXTEN}@ITSP2)

If we attempt calls from the PBX over the PRI connected to the Astlinux
Gateway the calls connects, but there is no audio.

This is what we see:

ITSP1:

Accepting call from ‘XXXXXX’ to ‘XXXXXX’ on channel 0/22, span 1 Executing
[XXXXXX@outgoing:1] NoOp(“DAHDI/22-1″, “”” “) in new stack Executing
[XXXXXX@outgoing:2] Dial(“DAHDI/22-1″, “SIP/XXXXXX@ITSP1″) in new stack
Called XXXXXX@ITSP1

SIP/ITSP1-000000c6 is circuit-busy (This result is because the ITSP1

account is blocked for testing)

Everyone is busy/congested at this time (1:0/1/0)

ITSP2:

Executing [XXXXXX@outgoing:3] Dial(“DAHDI/22-1″, “SIP/XXXXXX@ITSP2″) in new
stack Called XXXXXX@ITSP2

SIP/ITSP2-000000c7 is making progress passing it to DAHDI/22-1

SIP/ITSP2-trunk-000000c7 answered DAHDI/22-1

Can someone please make suggestions or point us in the right direction to
resolve this no audio issue?

Thank you.

Message: 13

Date: Mon, 20 Jun 2011 11:13:40 +0200

From: Olivier

Subject: Re: [asterisk-users] ITSP failover for PRI

To: Asterisk Users Mailing List – Non-Commercial Discussion

Message-ID:

Content-Type: text/plain; charset=”iso-8859-1″

2011/6/20 Alex Balashov

> On 06/20/2011 04:20 AM, Olivier wrote:

>

> What about incoming calls ?

>> Do you have a way to have calls that normally comes from ITPS1 to

>> comes from ITSP2 ?

>>

>

> No, there is no BGP for the PSTN.

>

Yes, that’s what I thought but you never know ;-)

(Maybe SS7 offers such redundancy but I’ve got no experience of any king in

this domain).

>

> —

> Alex Balashov – Principal

> Evariste Systems LLC

> 260 Peachtree Street NW

> Suite 2200

> Atlanta, GA 30303

> Tel: +1-678-954-0670

> Fax: +1-404-961-1892

> Web: http://www.evaristesys.com/

>

> —

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Faxing with Asterisk 1.8.4 & T.38

On May 20, 2011, at 3:30 PM, Anthony Messina wrote:

> On 05/20/2011 01:20 PM, ewr@erols.com wrote:
>> #1) We would like to use Fax For Asterisk with asterisk 1.8.4 in order to receive faxes via T.38. Sending faxes is not a requirement. Does anyone have a working asterisk 1.8.4 configuration and ITSP provider that they can recommend? We have been trying T.38 DIDs from our current ITSP, but we have been unable to make it work. I am more than happy to purchase new DIDs from a different provider if they will consistently work and are fairly priced.
>
> I use http://www.ipcomms.net/ with a free inbound DID for faxes. I
> always receive T.38.

Thanks for the suggestion Anthony.

I went ahead and signed up for a free DID from http://www.ipcomms.net/, but am still unable to receive faxes sent to that DID into FaxForAsterisk.

I have tried faxing to the DID from 2 different fax machines connected to different POTS lines. One fax machine is a Xerox Workcentre, and the other is a Brother Intellifax. Can you provide some more information about your setup? If you wouldn’t mind sharing your sip.conf settings, and maybe any other FaxForAsterisk related dialplan settings I would be greatly appreciative. I feel like we must have *something* really stupid set incorrectly. The faxes usually attempt to send, and appear to be properly switching to T.38, but usually end up failing with a “receive partial”. We are currently using the Digium fax driver, but have also tried it with spandsp.

I am open to any suggestions on resolving these issues. Thanks for any help you might be able to give!

Eric
ewr@erols.com