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</o:shapelayout></xml><![endif]--></head><body lang=EN-US link=blue vlink=purple><div class=WordSection1><p class=MsoNormal>Hi,<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>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.<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>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.<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>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.<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>We're using an Asterisk based SIP-T1 trunking gateway and would like to implement failover between two ITSPs.<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal> <o:p></o:p></p><p class=MsoNormal>If we connect a soft phone to the gateway with the following lines in extensions.conf failover works. <o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>If one ITSP is unavailable the call flow cascades to the second ITSP and connects with audio.<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal> [outgoing]<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>exten => _1NXXNXXXXXX,1,NoOp(${CALLERID(all)="" <>}) exten =><o:p></o:p></p><p class=MsoNormal>_1NXXNXXXXXX,2,Dial(SIP/${EXTEN}@ITSP1)<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>exten => _1NXXNXXXXXX,3,Dial(SIP/${EXTEN}@ITSP2)<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal> If we attempt calls from the PBX over the PRI connected to the Astlinux Gateway the calls connects, but there is no audio.<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal> This is what we see:<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal> ITSP1:<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>Accepting call from 'XXXXXX' to 'XXXXXX' on channel 0/22, span 1 Executing [XXXXXX@outgoing:1] NoOp("DAHDI/22-1", """ <XXXXXX>") in new stack Executing [XXXXXX@outgoing:2] Dial("DAHDI/22-1", "SIP/XXXXXX@ITSP1") in new stack Called XXXXXX@ITSP1<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal> SIP/ITSP1-000000c6 is circuit-busy (This result is because the ITSP1<o:p></o:p></p><p class=MsoNormal>account is blocked for testing)<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>Everyone is busy/congested at this time (1:0/1/0)<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal> ITSP2:<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>Executing [XXXXXX@outgoing:3] Dial("DAHDI/22-1", "SIP/XXXXXX@ITSP2") in new stack Called XXXXXX@ITSP2<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>SIP/ITSP2-000000c7 is making progress passing it to DAHDI/22-1<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>SIP/ITSP2-trunk-000000c7 answered DAHDI/22-1<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal> <o:p></o:p></p><p class=MsoNormal>Can someone please make suggestions or point us in the right direction to resolve this no audio issue?<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>Thank you.<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>Message: 13<o:p></o:p></p><p class=MsoNormal>Date: Mon, 20 Jun 2011 11:13:40 +0200<o:p></o:p></p><p class=MsoNormal>From: Olivier <oza_4h07@yahoo.fr><o:p></o:p></p><p class=MsoNormal>Subject: Re: [asterisk-users] ITSP failover for PRI<o:p></o:p></p><p class=MsoNormal>To: Asterisk Users Mailing List - Non-Commercial Discussion<o:p></o:p></p><p class=MsoNormal> <asterisk-users@lists.digium.com><o:p></o:p></p><p class=MsoNormal>Message-ID: <BANLkTimxrxPaA+A1ocdWm8yP-6x+HHEcSg@mail.gmail.com><o:p></o:p></p><p class=MsoNormal>Content-Type: text/plain; charset="iso-8859-1"<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>2011/6/20 Alex Balashov <<a href="mailto:abalashov@evaristesys.com">abalashov@evaristesys.com</a>><o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>> On 06/20/2011 04:20 AM, Olivier wrote:<o:p></o:p></p><p class=MsoNormal>><o:p> </o:p></p><p class=MsoNormal>> What about incoming calls ?<o:p></o:p></p><p class=MsoNormal>>> Do you have a way to have calls that normally comes from ITPS1 to<o:p></o:p></p><p class=MsoNormal>>> comes from ITSP2 ?<o:p></o:p></p><p class=MsoNormal>>><o:p> </o:p></p><p class=MsoNormal>><o:p> </o:p></p><p class=MsoNormal>> No, there is no BGP for the PSTN.<o:p></o:p></p><p class=MsoNormal>><o:p> </o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>Yes, that's what I thought but you never know ;-)<o:p></o:p></p><p class=MsoNormal>(Maybe SS7 offers such redundancy but I've got no experience of any king in<o:p></o:p></p><p class=MsoNormal>this domain).<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>><o:p> </o:p></p><p class=MsoNormal>> --<o:p></o:p></p><p class=MsoNormal>> Alex Balashov - Principal<o:p></o:p></p><p class=MsoNormal>> Evariste Systems LLC<o:p></o:p></p><p class=MsoNormal>> 260 Peachtree Street NW<o:p></o:p></p><p class=MsoNormal>> Suite 2200<o:p></o:p></p><p class=MsoNormal>> Atlanta, GA 30303<o:p></o:p></p><p class=MsoNormal>> Tel: +1-678-954-0670<o:p></o:p></p><p class=MsoNormal>> Fax: +1-404-961-1892<o:p></o:p></p><p class=MsoNormal>> Web: <a href="http://www.evaristesys.com/">http://www.evaristesys.com/</a><o:p></o:p></p><p class=MsoNormal>><o:p> </o:p></p><p class=MsoNormal>> --<o:p></o:p></p><p class=MsoNormal>> ______________________________**______________________________**_________<o:p></o:p></p><p class=MsoNormal>> -- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com">http://www.api-digital.com</a> --<o:p></o:p></p><p class=MsoNormal>> New to Asterisk? Join us for a live introductory webinar every Thurs:<o:p></o:p></p><p class=MsoNormal>> <a href="http://www.asterisk.org/hello">http://www.asterisk.org/hello</a><o:p></o:p></p><p class=MsoNormal>><o:p> </o:p></p><p class=MsoNormal>> asterisk-users mailing list<o:p></o:p></p><p class=MsoNormal>> To UNSUBSCRIBE or update options visit:<o:p></o:p></p><p class=MsoNormal>> <a href="http://lists.digium.com/**mailman/listinfo/asterisk-**users%3chttp:/lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/**mailman/listinfo/asterisk-**users<http://lists.digium.com/mailman/listinfo/asterisk-users</a>><o:p></o:p></p><p class=MsoNormal>><o:p> </o:p></p><p class=MsoNormal>-------------- next part --------------<o:p></o:p></p><p class=MsoNormal>An HTML attachment was scrubbed...<o:p></o:p></p><p class=MsoNormal>URL: <<a href="http://lists.digium.com/pipermail/asterisk-users/attachments/20110620/79a1a8f5/attachment-0001.htm">http://lists.digium.com/pipermail/asterisk-users/attachments/20110620/79a1a8f5/attachment-0001.htm</a>><o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>------------------------------<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>Message: 14<o:p></o:p></p><p class=MsoNormal>Date: Mon, 20 Jun 2011 05:16:05 -0400<o:p></o:p></p><p class=MsoNormal>From: Alex Balashov <abalashov@evaristesys.com><o:p></o:p></p><p class=MsoNormal>Subject: Re: [asterisk-users] ITSP failover for PRI<o:p></o:p></p><p class=MsoNormal>To: asterisk-users@lists.digium.com<o:p></o:p></p><p class=MsoNormal>Message-ID: <4DFF0FD5.4070101@evaristesys.com><o:p></o:p></p><p class=MsoNormal>Content-Type: text/plain; charset=ISO-8859-1; format=flowed<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>On 06/20/2011 05:13 AM, Olivier wrote:<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>> Yes, that's what I thought but you never know ;-)<o:p></o:p></p><p class=MsoNormal>> (Maybe SS7 offers such redundancy but I've got no experience of any<o:p></o:p></p><p class=MsoNormal>> king in this domain).<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>SS7 certainly offers link redundancy, but the issue is that your <o:p></o:p></p><p class=MsoNormal>numbers can't just be flash-ported to a different underlying carrier.<o:p></o:p></p><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>-- <o:p></o:p></p><p class=MsoNormal>Alex Balashov - Principal<o:p></o:p></p><p class=MsoNormal>Evariste Systems LLC<o:p></o:p></p><p class=MsoNormal>260 Peachtree Street NW<o:p></o:p></p><p class=MsoNormal>Suite 2200<o:p></o:p></p><p class=MsoNormal>Atlanta, GA 30303<o:p></o:p></p><p class=MsoNormal>Tel: +1-678-954-0670<o:p></o:p></p><p class=MsoNormal>Fax: +1-404-961-1892<o:p></o:p></p><p class=MsoNormal>Web: <a href="http://www.evaristesys.com/">http://www.evaristesys.com/</a><o:p></o:p></p><p class=MsoNormal><span style='font-family:"Tahoma","sans-serif";color:#0037E6'><o:p> </o:p></span></p></div></body></html>