[asterisk-users] Maximum E1 Ports on Asterisk ?
Hi All, Just a little over thought. Sorry if someone already asked about this before. Is it possible to put all 16 Ports of E1 in One Asterisk Server ? And if it's not possible is there any suggestion or alternative for me to use more than 320 lines of outgoing calls on One Asterisk Server ? Thanks ZH -- _ -- 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] Maximum E1 Ports on Asterisk ?
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Cary Fitch Sent: Wednesday, December 22, 2010 7:49 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Maximum E1 Ports on Asterisk ? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andrew Latham Sent: Wednesday, December 22, 2010 6:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Maximum E1 Ports on Asterisk ? On Wed, Dec 22, 2010 at 8:50 AM, Zoel Hairi - Yahoo zoelha...@yahoo.co.id wrote: Hi All, Just a little over thought. Sorry if someone already asked about this before. Is it possible to put all 16 Ports of E1 in One Asterisk Server ? And if it's not possible is there any suggestion or alternative for me to use more than 320 lines of outgoing calls on One Asterisk Server ? Thanks ZH Zoel It is possible to do what you are asking. In general the issue is raised about having all your eggs in one basket where one server or hardware failure can drop all of your lines for a period of time. External solutions like Xorcom and Redfone are great ways of abstraction. The concurrent call load on a server relies on the work to be done on each call. If you are using multiple codecs and recording the calls in another file format with other complex dialplan or AGI scripts then one server may not handle the calls well. If everything is ALAW and just dialing though then this would not be a problem for one server. If you search the list for sizing concurrent and load you will find more information. One very nice thing is that testing is very easy with or without the E1 hardware, try running the TDMoE channels between two servers and run a SIPp or other test to see the issues in a lab. ~~~ Andrew lathama Latham lath...@gmail.com ~~~ -- _ In a previous post I also mentioned Xorcom. They do have a unique fail over ability with their Astribank systems. With dual servers, separate chassis and power supplies for the 4 port T1/E1 cards, USB interconnections, and redundant power supplies for the Astribanks, system downtime can be minimized, and if there is a failure, repair would be at worst, no screwdriver needed. If system failure would be idling 200 - 400 people, avoiding system down time would be a major objective. Cary Fitch Thanks Cary and Andrew, This is a great suggestion and alternative for me. I will take a look at Xorcom and the AstriBank. Once again, Thanks guys. ZH -- _ -- 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] Quintum AFT800 on Asterisk 1.4.29
Hi Seighalani, Sorry for the late reply. We used the SIP Trunk Concept, so we create a SIP Trunk to communicate with the Quintum AFT800. Here’s the detail : On sip.conf [] context=from-internal secret= type=user username= fromuser= qualify=no [QUINTUM] context= host=YourQuintumIP insecure=port,invite nat=yes secret= type=peer username= context=from-trunk and on Quintum we used default setting from the Tenor Configuration Manager except for these steps : - Create SIP User : with secret - Gateway Selection : use SIP - Trunk Circuit Routing Group : route to your created SIP User That’s what I do with my config. Thanks ZH From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of alireza sadeh seighalan Sent: Saturday, December 04, 2010 2:40 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Quintum AFT800 on Asterisk 1.4.29 hi Zoel would you tell us how do that? good luck seighalani On Fri, Dec 3, 2010 at 4:57 PM, Zoel Hairi - Yahoo zoelha...@yahoo.co.id wrote: All, This case solved. Thanks … J Regards, Zoel Hairi From: Zoel Hairi - Yahoo [mailto:zoelha...@yahoo.co.id] Sent: Monday, November 22, 2010 12:07 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Quintum AFT800 on Asterisk 1.4.29 Hi All, Is it possible to use Quintum AFT800 on Asterisk 1.4.29 as Trunk for Analog (like Digium Analog Card) ? And if it’s possible, could any one please give me the reference how to configure it on Asterisk 1.4.29. Thanks Regards, Zoel Hairi -- _ -- 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 -- free is to know that you have a different option -- _ -- 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] Quintum AFT800 on Asterisk 1.4.29
All, This case solved. Thanks . J Regards, Zoel Hairi From: Zoel Hairi - Yahoo [mailto:zoelha...@yahoo.co.id] Sent: Monday, November 22, 2010 12:07 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Quintum AFT800 on Asterisk 1.4.29 Hi All, Is it possible to use Quintum AFT800 on Asterisk 1.4.29 as Trunk for Analog (like Digium Analog Card) ? And if it's possible, could any one please give me the reference how to configure it on Asterisk 1.4.29. Thanks Regards, Zoel Hairi -- _ -- 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] Quintum AFT800 on Asterisk 1.4.29
Hi All, Is it possible to use Quintum AFT800 on Asterisk 1.4.29 as Trunk for Analog (like Digium Analog Card) ? And if it's possible, could any one please give me the reference how to configure it on Asterisk 1.4.29. Thanks Regards, Zoel Hairi -- _ -- 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] Fax On Demand - Asterisk 1.4.29
Hi All, Is there anyone who ever implemented successfully Fax On Demand on Asterisk 1.4.29 ? I've tried to look from Google about this issue and could not find any satisfying about this. Thanks in advance for all of you who willing to help And Sorry if there's any mistake in my question, cause this is my first time asking question in this mailing list. Thanks Regards, Zoel Hairi -- _ -- 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] Fax On Demand - Asterisk 1.4.29
Hi Tarek, what do you need exactly from Fax on demand? sending faxes? receiving faxes? In simple explanation is like this, Caller goes through IVR (After having been validated), Then Caller Choose Fax On Demand option and hang up, and then Asterisk Send the Caller a Fax that already been prepared. That's the plan that I had in mind. Thanks Regards, Zoel Hairi -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tarek Sawah Sent: Friday, September 24, 2010 5:42 PM To: Asterisk Users Subject: Re: [asterisk-users] Fax On Demand - Asterisk 1.4.29 i don't see any mistakes in your question.. but i still don't get it. what do you need exactly from Fax on demand? sending faxes? receiving faxes? From: zoelha...@yahoo.co.id To: asterisk-users@lists.digium.com Date: Fri, 24 Sep 2010 17:27:57 +0700 Subject: [asterisk-users] Fax On Demand - Asterisk 1.4.29 .ExternalClass p.ecxMsoNormal, .ExternalClass li.ecxMsoNormal, .ExternalClass div.ecxMsoNormal {margin-bottom:.0001pt;font-size:12.0pt;font-family:'Times New Roman','serif';} .ExternalClass a:link, .ExternalClass span.ecxMsoHyperlink {color:blue;text-decoration:underline;} .ExternalClass a:visited, .ExternalClass span.ecxMsoHyperlinkFollowed {color:purple;text-decoration:underline;} .ExternalClass span.ecxEmailStyle17 {font-family:'Tahoma','sans-serif';color:#1F497D;} .ExternalClass .ecxMsoChpDefault {;} @page WordSection1 {size:8.5in 11.0in;} .ExternalClass div.ecxWordSection1 {page:WordSection1;} Hi All, Is there anyone who ever implemented successfully Fax On Demand on Asterisk 1.4.29 ? Ive tried to look from Google about this issue and could not find any satisfying about this. Thanks in advance for all of you who willing to help And Sorry if theres any mistake in my question, cause this is my first time asking question in this mailing list. Thanks Regards, Zoel Hairi -- _ -- 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