Re: [asterisk-users] ADSL Load Balancing
On 3.11.2010 ?. 02:29 ?., Dan Journo wrote: Hi, I've got a client with two ADSL connections for redundancy. Is it possible to set up asterisk to connect to one SIP provider using both adsl connections and load balance between the two connections? Or to use one connection as the main one, and automatically fail over if the first connection drops? Or does this kind of thing need a serious network switch? Thanks Dan Hello, If you are using Cisco gear (891 SOHO routers for example), take a look at Policy Routing, you can set next-hop routers based on any information available to an access-list. Chris -- _ -- 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] GSM Gateway
Another vote for the Voiceblues. Rock solid equipment. Peter den Hartog wrote: http://www.2n.cz/products/gsm_gateways/voip_gsm_gateway/voiceblue_voip_gsm_gateway.html We use this one, and it works great.. easy to setup and it works with a normal network connection :) On Mon, Feb 8, 2010 at 1:52 PM, Peter peterp...@aboutsupport.com mailto:peterp...@aboutsupport.com wrote: Hello, I am looking for a gsm gateway that is SIP based i.e no need of FXO/FXS analogue connection. I searched the email archives and found messages from 2008 but not sure how accurate these are. What do you use and how well it works ? The only sensible one I found is one made by portech and one that is made by Eurodesign. The one from portech is like a trunk while the one from eurodesign relies on USB and project GSMOPEN. what would you recommend - trunk or usb ? Or there are other possibilities ? Thanks, Peter -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Groet // Kind regards, Peter den Hartog -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] spandsp as T.38 termination?
You can also take a look at the T.38 product from Attractel. http://attractel.com/faxterisk.php Disclaimer, I work for these guys. Chris On Sun, November 25, 2007 4:11 pm, Robert Moskowitz wrote: Olivier wrote: Robert, Do you mean T.38 passthrough ou T.38 to T.30 gateway ? The former is said to work with Asterisk 1.4 but the latter is not ... I know about what Asterisk 1.4 can do. And Asterisk 1.2 only does T.30 passthrough :) You need 'stuff' to handle fax. Stuff like spandsp, IAXmodem and T38modem. So using spandsp and rxtax and txfax you **SHOULD** be able to set up a efax with T.38 systems. Also T38modem will make the connection to Hylafax for the same. Cheers ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- 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-- 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-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Astricon Meetup
oohs no! Whats up, haven't heard much out of you lately. Chris Brian West wrote: Everyone, I will be attending Astricon in Phoenix and would like to have a little get together to discuss Open Source Telephony and the challenges we as developers and system integrators face. Exchange ideas and go over some use cases and see how we can all work together to improve our understanding of the dynamics of how everything works together. * Scaleability * Reusability of code * Standards (VoiceXML, MRCP and more) If anyone is interested please email me off list and we'll plan on having a meeting of minds. Thanks, Brian West FreeSWITCH.org ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- 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-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Any plans for proper faxing support
You can also give the our T.38 stack a try. http://www.attractel.com/t38.html Chris Childress AsteriskGuru.com Andrew Joakimsen wrote: I have already tried to contact to persons from Digium and I did not receive a response. I was wondering if there is any plan to support fully faxing in Asterisk, I.E.: A T38 Gateway of sorts. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- 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-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Feasibility Request
Hello Jeremy, We have implemented HA systems in the past for numerous clients, call centers, etc, where reliability matters. We can definitely offer you a bid on this, but I would like to speak with you a bit first to nail down the requirements. What would be a convenient time and number to reach you? Chris Jeremy Mann wrote: I have a ton of Nortel MICS/CICS phone systems and am looking for an easy way to integrate them. Two questions arise: 1.Is it feasible to use asterisk as a Man in the Middle for a T1 PRI system? The idea is to intercept outbound calls from the Nortel PBX and redirect them via VoIP to another asterisk box at another branch transparently(thus saving the LD cost). Otherwise I'd pass the call on to the T1 for outbound processing. Our Nortel is already PRI equipped, the PRI would just come from the Asterisk box instead of the Telco directly. 2. Is it feasible to use asterisk as a Man in the Middle for Analog lines? I'd be using anywhere from 4-12 lines depending on location size. I'd like to do the same feature as above(intercept outbound calls and redirect them using VoIP if they are inter-office calls. a. I'd also like the VoIP trunks to be used for outbound calls in the case of PSTN downtime or busy. For example, all 4 outgoing lines are in use, person 5 wants to make an outbound call and it gets redirected to one of my T1 offices. I'd attach their outbound caller ID to make it appear as the call came from that location. My inevitable hope is to reduce my analog presense in smaller communities to 1 primary Line for 911/emergency calling, and to get a published presense in the community. I'd then beef up my T1 locations to handle more VoIP based calls. Currently we're using on the order of 30k minutes a month of LD just intercompany, about 10k external (IntraLATA). I'd also like any insight or suggestions on uptime. We're a healthcare organization so 5-9's is what we'll require. Any suggestions on hardware configs(or better yet, Bids!) would be appreciated as well. I don't need VoIP capable phones yet, but if the system works well enough we'd probably startup our next location(averaging 3-6 per quarter) with a pure VoIP system with Nortel fallback(again, 5-9's is critical). I'm located in Dallas, TX for any bids that might include installation. We have a presense up to about 400 miles west of here. This e-mail, facsimile, or letter and any files or attachments transmitted with it contains information that is confidential and privileged. This information is intended only for the use of the individual(s) and entity(ies) to whom it is addressed. If you are the intended recipient, further disclosures are prohibited without proper authorization. If you are not the intended recipient, any disclosure, copying, printing, or use of this information is strictly prohibited and possibly a violation of federal or state law and regulations. If you have received this information in error, please notify Texas Health Management Group immediately at 1-817-310-4999. Texas Health Management Group, its subsidiaries, and affiliates hereby claim all applicable privileges related to this information. -- This message has been scanned for viruses and dangerous content by *MailScanner* http://www.mailscanner.info/, and is believed to be clean. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
