ITSP failover for PRI

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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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