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 at outgoing:1] NoOp("DAHDI/22-1", """
<XXXXXX>") in new stack Executing
[XXXXXX at outgoing:2] Dial("DAHDI/22-1", "SIP/XXXXXX at
ITSP1") in new stack
Called XXXXXX at 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 at outgoing:3] Dial("DAHDI/22-1", "SIP/XXXXXX
at ITSP2") in new
stack Called XXXXXX at 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 <oza_4h07 at yahoo.fr>
Subject: Re: [asterisk-users] ITSP failover for PRI
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <BANLkTimxrxPaA+A1ocdWm8yP-6x+HHEcSg at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
2011/6/20 Alex Balashov <abalashov at evaristesys.com>
> 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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Message: 14
Date: Mon, 20 Jun 2011 05:16:05 -0400
From: Alex Balashov <abalashov at evaristesys.com>
Subject: Re: [asterisk-users] ITSP failover for PRI
To: asterisk-users at lists.digium.com
Message-ID: <4DFF0FD5.4070101 at evaristesys.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
On 06/20/2011 05:13 AM, Olivier wrote:
> 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).
SS7 certainly offers link redundancy, but the issue is that your
numbers can't just be flash-ported to a different underlying carrier.
--
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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