Re: [asterisk-users] odbc error - server is gone
isql? On Sat, Apr 30, 2011 at 6:18 PM, Pezhman Lali l...@lopl.net wrote: check your odbc connection with isql best On Fri, Apr 29, 2011 at 9:33 PM, Warren Selby wcse...@selbytech.comwrote: You're using 1.4.2. Why not try upgrading to a more recent release of 1.4 (I believe 1.4.41 is current) and see if your issue has been resolved. Thanks, --Warren Selby, dCAP On Apr 29, 2011, at 7:32 AM, Rizwan Hisham rizwanhas...@gmail.com wrote: Yes I have it there, here the content of the file: i think the code is buggy, here is a comment from the function which generated the error (ast_odbc_smart_execute in res_odbc.c line 155 ) /* This is a really bad method of trying to correct a dead connection. It * only ever really worked with MySQL. It will not work with any other * database, since most databases prepare their statements on the server, * and if you disconnect, you invalidate the statement handle. Hence, if * you disconnect, you're going to fail anyway, whether you try to execute * a second time or not. */ This function is used all over. On Fri, Apr 29, 2011 at 12:23 AM, Sherwood McGowan sherwood.mcgo...@gmail.com sherwood.mcgo...@gmail.com wrote: On Thu, Apr 28, 2011 at 11:09 AM, Rizwan Hisham rizwanhas...@gmail.com rizwanhas...@gmail.com wrote: Hi list, yesterday I converted my voicemail.conf to realtime voicemail and also configured to store the voicemessages in a database using odbc as described here http://www.voip-info.org/wiki/view/Asterisk+RealTime+Voicemailand herehttp://www.voip-info.org/wiki/view/Asterisk+Voicemail+ODBC+storage . I am using asterisk 1.4.2 with mysql. I also installed the proper odbc driver for mysql on the server. I successfully completed the conversion of a lot of voicemail users into db yesterday. But today on the CLI thsi error was showing; [Apr 28 11:40:54] WARNING[24676]: res_odbc.c:147 ast_odbc_smart_execute: SQL Execute returned an error -1: 08S01: [MySQL][ODBC 3.51 Driver][mysqld-5.0.68-log]MySQL server has gone away (70) [Apr 28 11:40:54] WARNING[24676]: res_odbc.c:147 ast_odbc_smart_execute: SQL Execute returned an error -1: 08S01: [MySQL][ODBC 3.51 Driver][mysqld-5.0.68-log]MySQL server has gone away (70) [Apr 28 11:40:54] WARNING[24676]: app_voicemail.c:2239 inboxcount: SQL Execute error! [SELECT COUNT(*) FROM voicemessages WHERE dir = '/var/spool/asterisk/voicemail/default/1757XXX/INBOX'] I know that the error is caused due to stale odbc connection with mysql. But i want to find out if there is a cure for it. Why the connection went stale in the first place also. Any ideas? -- Best Ragards Rizwan Qureshi VoIP/Asterisk Engineer Axvoice Inc. V: +92 (0) 6767 26 E: rizwanhas...@gmail.comrizwanhas...@gmail.com W: http://www.axvoice.com/www.axvoice.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users do you have sanitysql = select 1 configured in res_odbc.ini? -- Sherwood McGowan Telecommunications and VOIP Consultant -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Ragards Rizwan Qureshi VoIP/Asterisk Engineer Axvoice Inc. V: +92 (0) 6767 26 E: rizwanhas...@gmail.comrizwanhas...@gmail.com W: http://www.axvoice.com/www.axvoice.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit:
Re: [asterisk-users] HA Asterisk
8 PRI’s? I’d be using something like an AudioCodes Mediant 1000. No messing around with switches and cables an crap. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Satish Patel Sent: Saturday, April 30, 2011 10:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] HA Asterisk Tell me how to do pri failover. I meant we have one pri line but two asterisk in HA. Currently we are doing manually Swapping pri line. -- Sent from my iPhone On Apr 30, 2011, at 2:13 AM, RAJNIKANT VANZA rajniva...@gmail.com wrote: Hi Kaushal, I have done HA for Asterisk servers as well as SIP Server (kamailio). Please write your detail requirement. - how many Asterisk Sever require for HA? - How much down time acceptable during Asterisk Sever failover? - Which type Asterisk Sever Failover u required? Send me your detail requirement and answer of above question ASAP. -- Best Regards, Rajnikant Vanza Software Engineer --- Working On Linux,C/C++,VoIP,Asterisk Technology On Sat, Apr 30, 2011 at 7:59 AM, Kaushal Shriyan kaushalshri...@gmail.com wrote: Hi, I have been looking at Asterisk SCF http://www.asterisk.org/asterisk/scf, but its not yet production ready. Can someone please pitch in about HA feature in Asterisk ? (HA - High Availability.) Also, What would be the pros and cons of using AsteriskNow over Asterisk ? Are the versions same in Asterisk and AsteriskNow ? We have been evaluating Asterisk for our Voice Application and it seems it would fit the requirement. Is Asterisk a CPU Intensive or a Memory Intensive application. Please suggest/guide. Regards, Kaushal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Queue Setup
Hi All, I have Asterisk 1.6.2.13, I need to setup a queue of (6) agents, Ring All strategy, I need to set the maximum total time for the caller (Ringing/OR Waiting) on the queue is (2) minutes before going to a fail-over which is a Ring Group of external numbers. How the total max time is being calculated in terms of the number of agents, Ring Strategy, Agent Timeout, Retry, etc.. Can you please explain how it goes. Thank you.-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP, IAX2 and ISDN ISUP data
Hello, Does Asterisk support the history-info header as well? Also, what kind of ISDN mappings are available in 1.4 and 1.6.2 versions? Thanks, Elliot On Thu, Jan 27, 2011 at 1:08 AM, Kevin P. Fleming kpflem...@digium.com wrote: On 01/25/2011 12:44 AM, Phil Lello wrote: Hi all, I'm looking at my options for getting access to ISDN ISUP fields from DDI numbers, when connecting to a 3rd party Asterisk server. This is for a custom voicemail solution, and at this stage I want to avoid renting a PRI. The information I need to capture is: - Calling Number - Called Number (e.g. the DDI handling the call) - Redirecting Number (e.g. the device diverting to the voicemail DDI) - Originally Called Number (e.g. So if Adam phones Bob, Bob is diverted to Charlie, and Charlie is diverted to Voicemail, then Adam probably doesn't want Charlie's Voicemail). Asterisk 1.8 can receive, transmit and transport all this information over ISDN and SIP, including mid-call updates. I believe this information should be in SIP Divert headers, can someone confirm this? There are a number of SIP headers involved. Diversion, P-Asserted-Identity and Remote-Party-Id, if not others. Do I get the same information if I use an IAX2 connection to connect a local Asterisk server to an external one? It is possible that this information will transport properly across IAX2 connections between Asterisk 1.8 servers, but that scenario wasn't tested by the developers that worked on it. Does IAX2 route GSM/ISDN SMS between servers, and if so, would the remote/ISDN connected server need to explicitly support this, or do the remote cards look local? Asterisk does not support native SMS, and doesn't transport it between servers. There is an SMS application, but it is an SMS endpoint, not a router. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HA Asterisk
On Sun, May 1, 2011 at 10:14 PM, Jim Dickenson dicken...@cfmc.com wrote: Xorcom makes a box that connects via USB that can do failover. You connect the box to the two system via a USB cable to each system. When the box detects the primary system fails it switches over the the second one. No need for any extra hardware, except a USB cable. http://www.xorcom.com/catalog/xr0015.html http://www.xorcom.com/optional-extras/twinstar.html Hi Jim, Thanks for sharing the technical details. Still not able to understand the setup. Let me explain what i understand is the 8 PRI line would be connected to the xorcom box and from there USB out would be connected to Primary Asterisk Server and Secondary Asterisk Server. So we do not need any 8 port PRI Card on the Primary Asterisk Server and Secondary Asterisk Server ? Please correct me if i am wrong. Thanks Kaushal -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Apr 30, 2011, at 8:31 PM, Kaushal Shriyan wrote: On Sun, May 1, 2011 at 8:48 AM, Michelle Dupuis mdup...@ocg.ca wrote: Yes that's it - one PRI line in, 2 out (one to the PRI card in each server). If you have lots of PRI lines, you may want to consider a dedicated PRI-to-SIP appliance.. Hi, Thanks a Lot Michelle, Also please let me know the model/make for dedicated PRI-to-SIP appliance. Would appreciate if you can share the details along with the Network Diagram in case of 8 PRI Lines. Much appreciated. Regards, Kaushal From: asterisk-users-boun...@lists.digium.com [ asterisk-users-boun...@lists.digium.com] On Behalf Of Kaushal Shriyan [ kaushalshri...@gmail.com] Sent: Saturday, April 30, 2011 11:03 PM To: Asterisk Users List Subject: Re: [asterisk-users] HA Asterisk On Sun, May 1, 2011 at 2:13 AM, Michelle Dupuis mdup...@ocg.camailto: mdup...@ocg.ca wrote: There are lots out there, but here's the result of a quick search... http://www.zycon.com/News-Press-Releases/Read/RJ45-CAT5e-Line-Off-Line-Switch-Series-Offers-3-Control-Methods-R1357.html and the software to trigger the switch: www.generationd.comhttp://www.generationd.com Hi Michelle So what i understand is that the Single PRI Line from telco is connected to RJ45 (8 wire) A-B switched controllable by serial port and then there will be two patch cord from the A-B switch which will be connected to the 2 Asterisk Box containing PRI Card on each box. Please let me know if i am understanding you correctly or if you can help me with Network Diagram that would be really helpful. Also I have 8 PRI in my setup. How it would fit in this setup. The reason being we need to have atleast 320 Outbound Calls per min if i have 8 PRI Lines for our Voice Application. Regards, Kaushal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Odd error in libpri
I just updated libpri 1.4 on my system to the latest from that branch and my QSIG connection to an NEC SV8300 stopped working. The trace showing the problem is below: q931.c:5640 q931_connect: Call 7168 enters state 10 (Active). Hold state: Idle DL-DATA request Protocol Discriminator: Q.931 (8) len=21 TEI=0 Call Ref: len= 2 (reference 7168/0x1C00) (Sent to originator) Message Type: CONNECT (7) TEI=0 Transmitting N(S)=29, window is open V(A)=29 K=7 Protocol Discriminator: Q.931 (8) len=21 TEI=0 Call Ref: len= 2 (reference 7168/0x1C00) (Sent to originator) Message Type: CONNECT (7) [18 03 a9 83 81] Channel ID (len= 5) [ Ext: 1 IntID: Implicit Other(PRI) Spare: 0 Exclusive Dchan: 0 ChanSel: As indicated in following octets Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 Type: NET] [1e 02 81 82] Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Progress Description: Called equipment is non-ISDN. (2) ] [29 05 0b 05 01 0e 03] Time Date (len= 7) [ 11-05-01 14:03 ] Protocol Discriminator: Q.931 (8) len=13 TEI=0 Call Ref: len= 2 (reference 7168/0x1C00) (Sent from originator) Message Type: STATUS (125) [08 03 81 e0 29] Cause (len= 5) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Mandatory information element is missing (96), class = Protocol Error (e.g. unknown message) (6) ] Cause data 1: 29 (41) [14 01 04] Call State (len= 3) [ Ext: 0 Coding: CCITT (ITU) standard (0) Call state: Call Delivered (4) Received message for call 0x2aaab81d15c0 on link 0x1b0db440 TEI/SAPI 0/0 -- Processing IE 8 (cs0, Cause) -- Processing IE 20 (cs0, Call State) As I'm reading this, libpri thinks that the SV8300 is complaining that a mandatory IE is missing, in this case time/date. However, the field is THERE. But when I go back to a working libpri (r1878), I see that the time/date is NOT sent on the CONNECT. If I'm reading Q.931 correctly, 5.1.8 (page 118) says that the Date/time IE may be included as a network option. I see this was added to libpri at revision 2187, in response to issue number 18047. I played around a bit. Since the spec includes seconds, I added seconds to see if that made it work, but it didn't. I DID work when I deleted Q931_IE_TIME_DATE from connect_net_ies. Whether or not it's a bug for the SV8300 to reject that IE, it's likely that NEC won't fix it. This likely means that a new config option is needed, but I think that means it'd also have to be done in chan_dahdi.c in Asterisk in addition to libpri. Is that right? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HA Asterisk
On May 1, 2011, at 10:09 AM, Kaushal Shriyan wrote: On Sun, May 1, 2011 at 10:14 PM, Jim Dickenson dicken...@cfmc.com wrote: Xorcom makes a box that connects via USB that can do failover. You connect the box to the two system via a USB cable to each system. When the box detects the primary system fails it switches over the the second one. No need for any extra hardware, except a USB cable. http://www.xorcom.com/catalog/xr0015.html http://www.xorcom.com/optional-extras/twinstar.html Hi Jim, Thanks for sharing the technical details. Still not able to understand the setup. Let me explain what i understand is the 8 PRI line would be connected to the xorcom box and from there USB out would be connected to Primary Asterisk Server and Secondary Asterisk Server. So we do not need any 8 port PRI Card on the Primary Asterisk Server and Secondary Asterisk Server ? Please correct me if i am wrong. Thanks Kaushal Correct, there are no cards inside any system. You have an external box that can have a combination of PRI, FXO and FXS ports; depending on need. The external box is connected via USB to the two systems. The twinstar option allows you to connect the external box to two systems via USB and provides fall over from primary to secondary on failure of the primary.-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] HA Asterisk
That's right. By failover in this context just means making a connection to another box. There is no detection of Asterisk hanging, missing registrations, no synchronization of mailboxes etc. (So the word HA is a bit misleading)... From: asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.com] On Behalf Of Kaushal Shriyan [kaushalshri...@gmail.com] Sent: Sunday, May 01, 2011 1:09 PM To: Asterisk Users List Subject: Re: [asterisk-users] HA Asterisk On Sun, May 1, 2011 at 10:14 PM, Jim Dickenson dicken...@cfmc.commailto:dicken...@cfmc.com wrote: Xorcom makes a box that connects via USB that can do failover. You connect the box to the two system via a USB cable to each system. When the box detects the primary system fails it switches over the the second one. No need for any extra hardware, except a USB cable. http://www.xorcom.com/catalog/xr0015.html http://www.xorcom.com/optional-extras/twinstar.html Hi Jim, Thanks for sharing the technical details. Still not able to understand the setup. Let me explain what i understand is the 8 PRI line would be connected to the xorcom box and from there USB out would be connected to Primary Asterisk Server and Secondary Asterisk Server. So we do not need any 8 port PRI Card on the Primary Asterisk Server and Secondary Asterisk Server ? Please correct me if i am wrong. Thanks Kaushal -- Jim Dickenson mailto:dicken...@cfmc.commailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ On Apr 30, 2011, at 8:31 PM, Kaushal Shriyan wrote: On Sun, May 1, 2011 at 8:48 AM, Michelle Dupuis mdup...@ocg.camailto:mdup...@ocg.ca wrote: Yes that's it - one PRI line in, 2 out (one to the PRI card in each server). If you have lots of PRI lines, you may want to consider a dedicated PRI-to-SIP appliance.. Hi, Thanks a Lot Michelle, Also please let me know the model/make for dedicated PRI-to-SIP appliance. Would appreciate if you can share the details along with the Network Diagram in case of 8 PRI Lines. Much appreciated. Regards, Kaushal From: asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com [asterisk-users-boun...@lists.digium.commailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kaushal Shriyan [kaushalshri...@gmail.commailto:kaushalshri...@gmail.com] Sent: Saturday, April 30, 2011 11:03 PM To: Asterisk Users List Subject: Re: [asterisk-users] HA Asterisk On Sun, May 1, 2011 at 2:13 AM, Michelle Dupuis mdup...@ocg.camailto:mdup...@ocg.camailto:mdup...@ocg.camailto:mdup...@ocg.ca wrote: There are lots out there, but here's the result of a quick search... http://www.zycon.com/News-Press-Releases/Read/RJ45-CAT5e-Line-Off-Line-Switch-Series-Offers-3-Control-Methods-R1357.html and the software to trigger the switch: www.generationd.comhttp://www.generationd.com/http://www.generationd.comhttp://www.generationd.com/ Hi Michelle So what i understand is that the Single PRI Line from telco is connected to RJ45 (8 wire) A-B switched controllable by serial port and then there will be two patch cord from the A-B switch which will be connected to the 2 Asterisk Box containing PRI Card on each box. Please let me know if i am understanding you correctly or if you can help me with Network Diagram that would be really helpful. Also I have 8 PRI in my setup. How it would fit in this setup. The reason being we need to have atleast 320 Outbound Calls per min if i have 8 PRI Lines for our Voice Application. Regards, Kaushal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.comhttp://www.api-digital.com/ -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every
Re: [asterisk-users] HA Asterisk
On Mon, May 2, 2011 at 1:46 AM, Jim Dickenson dicken...@cfmc.com wrote: On May 1, 2011, at 10:09 AM, Kaushal Shriyan wrote: On Sun, May 1, 2011 at 10:14 PM, Jim Dickenson dicken...@cfmc.com wrote: Xorcom makes a box that connects via USB that can do failover. You connect the box to the two system via a USB cable to each system. When the box detects the primary system fails it switches over the the second one. No need for any extra hardware, except a USB cable. http://www.xorcom.com/catalog/xr0015.html http://www.xorcom.com/optional-extras/twinstar.html Hi Jim, Thanks for sharing the technical details. Still not able to understand the setup. Let me explain what i understand is the 8 PRI line would be connected to the xorcom box and from there USB out would be connected to Primary Asterisk Server and Secondary Asterisk Server. So we do not need any 8 port PRI Card on the Primary Asterisk Server and Secondary Asterisk Server ? Please correct me if i am wrong. Thanks Kaushal Correct, there are no cards inside any system. You have an external box that can have a combination of PRI, FXO and FXS ports; depending on need. The external box is connected via USB to the two systems. The twinstar option allows you to connect the external box to two systems via USB and provides fall over from primary to secondary on failure of the primary. Hi Jim, Thanks for the explanation, I have couple of questions here. 1) Does the xorcom box support *8 Port PRI E1 Interface*. ? 2) Also the Primary and Secondary Asterisk Server can be any server which will run Asterisk or AsteriskNow (http://www.asterisk.org/asterisknow) Application and customizable or do i also need to buy this from Xorcom ? Not sure i understand that. 3) How does the xorcom box communicate with the Asterisk Server which do not contain any PRI Card inside the system. Much Appreciated. Thanks and Regards, Kaushal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
