Re: [asterisk-users] DNS broken for www.voip-info.org ??
DNS for www.Voip-info.org should be back online shortly. In the meantime here are mirrors: a.. SimpleVoip.info - Location: California, USA, Bandwidth: 100M, Updated Nightly b.. Malico Inc. - Location: Tao-Yuang, Taiwan, Bandwidth: 1Mbps, Updated Daily c.. Totalip - Location: Oslo, Norway. Bandwidth: 100Mbit, Updated Daily d.. http://www.telephreak.org/voip-info - Location: Florida, USA. Bandwidth: Dual DS1, Updated hourly. e.. AFOYI Mirror - Location: Adelaide, Australia, Bandwidth: 12Mbps, Updated 3 hourly f.. LinuxSystems - Location: Florida, Bandwidth: 10M, Updated Nightly Thanks for using voip-info.org! [EMAIL PROTECTED] - Original Message - From: Steve Johnson [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Saturday, December 15, 2007 7:57 AM Subject: [asterisk-users] DNS broken for www.voip-info.org ?? The DNS for www.voip-info.org seems to be non-responsive. Is there a mirror of this invaluable resource site? Tx, Steve dig www.voip-info.org ;; Got SERVFAIL reply from xxx.xxx.xxx.xxx, trying next server ; DiG 9.4.1-P1 www.voip-info.org ;; global options: printcmd ;; Got answer: ;; -HEADER- opcode: QUERY, status: SERVFAIL, id: 61402 ;; flags: qr rd ra; QUERY: 1, ANSWER: 0, AUTHORITY: 0, ADDITIONAL: 0 ;; QUESTION SECTION: ;www.voip-info.org. IN A ;; Query time: 4724 msec ;; SERVER: 127.0.0.1#53(127.0.0.1) ;; WHEN: Sat Dec 15 11:54:57 2007 ;; MSG SIZE rcvd: 35 ___ --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] Voip-info.org
The colo where voip-info.org is hosted is suffering a DOS attack. They hope to recover soon. In the meantime the voip-info.org mirrors are available http://72.14.253.104/search?q=cache:8E6ozIeVoSkJ:www.voip-info.org/wiki/view/Voip-Info%2BMirrors+voip-info+mirrorshl=enlr=lang_enstrip=1 [EMAIL PROTECTED] - Original Message - From: Ed Nuñez To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Sent: Wednesday, June 06, 2007 3:46 AM Subject: [asterisk-users] Voip-info.org Is anyone else having trouble going into voip-info.org today? ___ --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
[asterisk-users] voip-info.org status update
A short status update: Yesterday 3 of the 4 disk drives in the RAID array on the server that hosts voip-info.org failed. The coloprovider is currently working to replace the drives and I'm hoping that the site returns to service soon. Tomorrow is looking most likely. I'd like to thank all those that have called or emailed to offer help and/or encouragement. I will definately be looking for an easy way to create a mirror site once voip-info.org is back up. This is made difficult by the dynamic nature of the site, but its been on my list of things to do for a while now. Thanks for using voip-info.org! [EMAIL PROTECTED] ___ --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
Re: [Asterisk-Users] Define variable in sip.conf
- Original Message - From: Olle E. Johansson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, October 05, 2005 7:55 PM Subject: Re: [Asterisk-Users] Define variable in sip.conf Benjamin Lawetz wrote: I'm looking for a way to transmit a user specific variable to my dialplan If we use the example of the hair color, I was thinking of having something like: [bob] context=users host=dynamic secret=password type=friend username=bob hair=brown Use setvar=hair=brown in 1.2beta or CVS head. Works for both IAX2 and SIP configuration files. /Olle ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DID on analog line
Multitech makes ATAs and Gateways that support EM signaling: http://www.voip-info.org/wiki/view/Multitech - Original Message - From: Kevin P. Fleming [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Thursday, October 13, 2005 8:23 PM Subject: Re: [Asterisk-Users] DID on analog line Peder @ NetworkOblivion wrote: And it's wink-start on an EM analog circuit, not on a standard analog phone line from your telco. You would need a card that supports EM to do it even if the telco provided it (not sure if the Digium cards support it, but I tend to doubt it). We do not have any four-wire analog cards, so we cannot handle analog EM signaling. We do support EM over digital links, though. Analog DID can be done using ground start as well, so it's possible than an FXS port could be convinced to do it, but nobody has implemented it, since analog DID is not something that carriers really want to continue selling. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] wifi phones - desk
At the Boston VON last month Linksys was showing an add-on for their SPA-941 to make it wireless. See: http://www.voip-info.org/tiki-index.php?page=Linksys - Original Message - From: Will Glass-Husain To: asterisk-users@lists.digium.com Sent: Friday, October 07, 2005 6:14 AM Subject: [Asterisk-Users] wifi phones - desk Hi, I'm provisioning an office with limited cabling. I'm looking for a desk based wifi phone. Most of the ones I've seen are handsets. Any ideas? Thanks, WILL ___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] WiFi Phones
Anyone have good words to say about any of the WiFi handsets currently available? Thanks. Jim ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] voip-info - is it alive
If its NOT working for you, please send a traceroute to: [EMAIL PROTECTED] Thanks. Jim [EMAIL PROTECTED] - Original Message - From: Bob Goddard [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, August 27, 2005 3:15 AM Subject: Re: [Asterisk-Users] voip-info - is it alive On Friday 26 Aug 2005 14:54, Julian Lyndon-Smith wrote: I cannot reach voip-info - is it just me or is the site not available ? There is a bad route being propogated. B ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] http://www.voip-info.org/ front page taken outby spammer
History page is back. I think it got too big for the software to deal with. I changed it to show only the last 100 versions. There is an 'undo' option on the history page, but its never worked correctly, and so I have not enabled it. I'm working on a software upgrade that will hopefully address some of these issues. Jim James H. Thompson[EMAIL PROTECTED] - Original Message - From: Remco Barende To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Monday, August 08, 2005 10:36 AM Subject: Re: [Asterisk-Users] http://www.voip-info.org/ front page taken outby spammer Today the front page of http://www.voip-info.org/ was taken out by a spammer. It also seem the history page for http://www.voip-info.org/ was also nuked. I've restored the best I could using google cache, but still missing some information. Who is an admin on http://www.voip-info.org/ and can fix it? Google cache is a hard way to fix wiki-busting -- the easiest way is to click on "history" at the top of the page, go right to the version before the spam, copy it, then paste it into an edit of the page.. Of course, now, it's harder, because since the page was restored, people have since modified it.. (also, for some reason, when I click on "history", nothing seems to happen)..Highly offtopic but weird that the wiki software doesn't have an option to undo all the changes that one user or one ip address made.Wouldn't be too hard to implement IMHO, just keep a copy of all the stuff that was changed / deleted / added for an x number of days and build an option to automagically undo the changes.___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sip/rtp performance monitoring
If the customers are using an ATA or other VOIP device that supports RTCP, then you can often get packet loss and jitter stats by extracting the RTCP packets and analyzing them. This will actually give you the packet loss and jitter that the customer is seeing in the received RTP stream from you. A combination of Tetheral and grep or perl can get you along way in capturing and analyzing this data. Jim James H. Thompson[EMAIL PROTECTED] - Original Message - From: Forrest Christian To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Saturday, August 06, 2005 9:43 AM Subject: [Asterisk-Users] sip/rtp performance monitoring I'm currently running asterisk to provide VoIP services to clients of the ISP I work for.I would like to be able to tell if I am loosing packets and/or are having other issues with any of the voice streams, so I can address them proactively.I'm not particularly interested in spending oodles of money buying one of the commercial analysis tools. Is there some open source tool (or something I can monitor in asterisk) which will tell me if I'm missing packets or similar? I realize this will likely be only from the customer towards me since I can't really monitor at the customer end.-forrest___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: [Asterisk -Users] modprobe wcfxo fails.
I don't remember, what was the problem? Jim James H. Thompson[EMAIL PROTECTED] - Original Message - From: Tim King To: [EMAIL PROTECTED] Sent: Sunday, July 17, 2005 9:09 AM Subject: [Asterisk-Users] modprobe wcfxo fails. I was reading a thread where you were helping someone out and noticed it ended without resolve. Was this issue ever taken care of?I seem to be having the exact same problem. Thanks Tim King Network Engineer Computer Network Solutions LLC 1331 Plainfield Ave Grand Rapids MI 49505 Phone: 800-669-3290 ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] System Jsut hangs Up
Hard to say without seeing all config files. [EMAIL PROTECTED] is an easy way to get a running system with AMP. Jim James H. Thompson[EMAIL PROTECTED] - Original Message - From: Tim King To: [EMAIL PROTECTED] Sent: Sunday, July 17, 2005 3:52 PM Subject: [Asterisk-Users] System Jsut hangs Up I took care of my earlier problem. But now if I call in it just says goodbye, And on my extension no matter what I do it seems to just hang up on me immediately. Its a slackware 10.1 box with Digium 22b card. I am running AMP so its mysql driven. Im not seeing any errors. It just hangs up. Tim King Network Engineer Computer Network Solutions LLC 1331 Plainfield Ave Grand Rapids MI 49505 Phone: 800-669-3290 ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VOIP-INFO
Voip-info is back up --in-spite of Murphy's law. This was phase I(installlatest version of O/S)of an upgrade to improve performance and functionality. Hopefully with Phase II we will see much better performance and new functions. For those that asked, theprimary voip-info-org sponsor: www.commpartners.us provides a dedicated server, bandwidthand hosting in theirLas Vegas data center. Its slow not for any lack of resoruces, but because the software used is rather resource intensive. Jim James H. Thompson[EMAIL PROTECTED] - Original Message - From: Chris Coulthurst To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Thursday, June 09, 2005 11:33 AM Subject: [Asterisk-Users] VOIP-INFO Anyone else unable to get to www.voip-info.org? Site is returning'connection refused' here.Chris Coulthurst[EMAIL PROTECTED]___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] General voip mailing list
Not a mailing list but VOIP forums on DSL Reports are large and active: www.dslreports.com Jim James H. Thompson[EMAIL PROTECTED] - Original Message - From: Gerard Marcel To: asterisk-users@lists.digium.com Sent: Wednesday, April 20, 2005 3:47 AM Subject: [Asterisk-Users] General voip mailing list Does anyone here know of any general, good voip mailing list? I amhaving a hard time with broadvoice and the company is not answeringits phone.TIA,GM___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Router with QoS recommendations
Any FreeBSD/OpenBSD solutions we should add to the list at the bottom of this page? http://www.voip-info.org/tiki-index.php?page=VOIP+Routers Jim James H. Thompson[EMAIL PROTECTED] - Original Message - From: Arnaud PIGNARD To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Monday, April 04, 2005 3:57 AM Subject: Re: [Asterisk-Users] Router with QoS recommendations At 15:36 04/04/2005, you wrote:On 03-Apr-2005, Tim Pushor wrote: I prefer PF's approach to security first, convenience second, and I *really* like the fact that PF has a real parser. As the requements get more complex, having everything in one file, and very readable and structured is a huge plus. Also, the integration with ALTQ is nice, especially for these types of applications.I agree with everything Tim wrote above, and I'll add that the biggestfactor that influenced me in my move to OpenBSD for my firewall was thatit was the only free unix I found that could do bidirectional filteringin bridged mode. As in, when you're in a bridged configuration you canfilter in and out on an interface. Neither Linux nor FreeBSD could dothis. It's certainly an edge case, but if you need that feature it'sinvaluable.I'm using ALTQ since FreeBSD 4.6 and it's also exist ALTQ+PF that's near the same as OpenBSD version.And i confirm that's shapping with ALTQ work great ! Even with 32 Kbps.You can easely shape around 1000 rules and have a full Fast Ethernet port on a dual PIII (FreeBSD ALTQ port without PF)ALTQ have many shape algo, maybe the only one with such diversity.You have some CD distribution with ALTQ enable.I posted my asterisk altq experiments here: http://slacker.com/~nugget/asterisk4.php--David McNett [EMAIL PROTECTED]http://slacker.com/~nugget/___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Arnaud Pignard ([EMAIL PROTECTED])Frontier Online - Opérateur Internet___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Are there online forums instead of this emailforum??
The Asterisk-users mailing list is available on Gmane which has a forum like interface as an option: http://dir.gmane.org/gmane.comp.telephony.pbx.asterisk.user Jim James H. Thompson[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Physically Small Box Asterisk Systems
Looking for reccomendations for a physically small box configurationthat will do: Run Asterisk One T1 Card One LAN port Enough CPU power to handle encoding/decoding all 24 T1 channels to/from G.729a Someone mentioned the mini-ITX systems, but there seemed to be a concern about adequate CPU power for doing transcoding of more than a few channels. Thanks. Jim James H. Thompson[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] cordless/wireless system with a ip base station?
Vtech and Uniden http://www.voip-info.org/tiki-index.php?page=VOIP+Phones#id416800 Jim James H. Thompson[EMAIL PROTECTED] - Original Message - From: Chuck To: asterisk-users@lists.digium.com Sent: Sunday, March 13, 2005 1:28 PM Subject: [Asterisk-Users] cordless/wireless system with a ip base station? does anyone know of a 2.4 or 5 ghz cordless phone system that has an ip base station?thanks___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Digium : no lead time!
At the Spring VON show that just finished there were two vendors showing Asterisk compatibleT1 cards with on-board DSPs. See: http://www.voip-info.org/tiki-index.php?page=Asterisk+hardware Jim James H. Thompson[EMAIL PROTECTED] - Original Message - From: Matthew Boehm To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Wednesday, March 09, 2005 11:18 AM Subject: Re: [Asterisk-Users] Digium : no lead time! Again, may be off topic but are there any cards out there supported byasterisk that have on-board DSPs to do better 729-711 or 729-PRIconversion?-MatthewTC wrote: This maybe the wrong place to ask this question but... why did you switch to the Sangoma? preliminary testing show Sangoma card/driver are better unload a full load not such an issue with a single 4 span cards but 2+ cards and the Digium T4xx cards start to drop calls, missed interupts etc ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] wiki down?
Wiki is back up. Between comment SPAM storms, over eager robots ignoring robots.txt, and mysql issues, it has been an interesting week. Jim James H. Thompson[EMAIL PROTECTED] [EMAIL PROTECTED] - Original Message - From: Roy Sigurd Karlsbakk To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Saturday, February 19, 2005 8:13 AM Subject: [Asterisk-Users] wiki down? hiis the wiki down again?roy___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ATA that actually work with T.38
Sipura 2100 is supposed to implement T.38 real-soon-now. I've got a Multi-tech ATA withT.38 supporton order on the theory that Multitech has been making well regarded FAX modems for years and might know how to actually do FAX reasonably well. Jim James H. Thompson[EMAIL PROTECTED] - Original Message - From: Steve Underwood To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Monday, February 14, 2005 5:24 AM Subject: [Asterisk-Users] ATA that actually work with T.38 Hi,I am implementing T.38, and finding a problem getting boxes that work with T.38 for testing. A lot (maybe most) ATAs now claim to support T.38, but I'm finding a lot of these lie. I have one box here that just crashes when it hears a fax tone. :-)I'm looking for boxes known to implement T.38 properly, and which really work in the real world.Regards,Steve___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] wikki problem
Surround the script with ~pp~ line 1 of script line 2 of script etc. ~/pp~ See example on this page: http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out+deliver+message Jim James H. Thompson[EMAIL PROTECTED] - Original Message - From: dean collins To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Friday, February 18, 2005 12:47 PM Subject: [Asterisk-Users] wikki problem Im trying to post a script on the wikki but it keeps screwing up the text because it interprets the text as commands that cause graphical errors. Is there some trick to make the wiki think that the text is just text? Tia, Dean ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom 1.4.1 firmware for IP500/IP600
I'm looking for a copy too. Jim James H. Thompson [EMAIL PROTECTED] - Original Message - From: Robert Augustyn [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Tuesday, January 25, 2005 5:00 PM Subject: RE: [Asterisk-Users] Polycom 1.4.1 firmware for IP500/IP600 If you have it, can I get a copy please, or possibly can you send it to the keeper of http://www.freedomphones.net/polycom/files/ I am looking for the latest boot image too. Thanks. robert -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael Devenijn Sent: Tuesday, January 25, 2005 5:30 PM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] Polycom 1.4.1 firmware for IP500/IP600 Does somebody have this new firmware from/for Polycom ? Thanks Michael ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Looking for a prepaid calling card platform
I'm looking for a prepaid calling card platform that: * easily scales to multiple servers with a common database for: redundancy, capacity, and performance Looking to start with capacity to handle100 simultaneous calls andbe able to easilyscale to 1000+ simultaneous calls. * in addition to the normal anti-fraud measures, supports an API for easily adding new anti-fraud tests along the lines of the following: For each newcall being attempted the system wouldinvoke an external authentication program and pass:reseller ID, card ID, time left on card, called # andcalling #; and the history/status for thelast several calls including for each call the called #, calling #, call duration, call timestamp andcall status (in-progress, completed, etc). Progrm would return: call OK, deny call with recording #x, invalidate card with recording #y. * ability to limit calls to a maxium duration and/or to require periodic IVR user response to continue a long call. * contolled, managed andprovisionedwith a web interface * support multiple resellers, each with password protected web access for managing their customers. * ability for customers to call an 800# tohear a recording giving themthe user a local non-800number they need to call to use the card. * credit card recharge support While willing to do minor customizations, would like to find something that is mostlyinstall and go. Open source would be nice, but willing to pay for a well done package. Suggestions welcome. Thanks. Jim James H. Thompson[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FWD-NAT-*
Making asterisk work through NAT is a pain and some of the Wiki stuff is wrong/out dated. This works for me:Please feel free tofix or point out what is wrong/outdated so someone else can fix. Thanks. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Looking for a wireless phone... wifi ortraditional wireless ?
The Vtech istied to Vonage (according to their press release) but I believe that the Uniden is not. The Uniden press release says: Uniden introduces 2005 technology partners Lingo, BroadVoice(R) and SunRocket(SM). Proving interoperability with these Internet phone service providers will allow consumers to use one of these VoIP platforms with Uniden's UIP1868 as part of a complete and cost-effective consumer VoIP offering. So I'm guessingthat the Uniden phoneis not locked to a specific provider. Jim James H. Thompson[EMAIL PROTECTED] - Original Message - From: Jeff R Glassman To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Thursday, January 13, 2005 4:46 AM Subject: RE: [Asterisk-Users] Looking for a wireless phone... wifi ortraditional wireless ? I beleive both are locked into a VOIP carrier (Vontage?) -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of James H. ThompsonSent: Wednesday, January 12, 2005 11:54 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Looking for a wireless phone... wifi ortraditional wireless ? Uniden and Vtech both just announced cordless phones with SIP ATAs built into the base station. You get better range and battery life compared to a WiFi phone. Jim James H. Thompson[EMAIL PROTECTED] - Original Message - From: Kim Lux To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Wednesday, January 12, 2005 5:49 PM Subject: Re: [Asterisk-Users] Looking for a wireless phone... wifi or traditional wireless ? An unflattering zyxel review:http://slacker.com/~nugget/asterisk3.phpI can't help but think my questions are out of place on this list... I'masking questions about SIP phones and everyone else is talking aboutasterisk. Sorry. On Wed, 2005-01-12 at 20:08 -0700, Kim Lux wrote: My wife wants a cordless phone for around the house. We are going to be using VOIP exclusively very shortly. Our current cordless phone is aged and on the verge of replacement. The other phone we are going to use is a SIP Budgetone. Should I buy a SIP to POTS converter and a new cordless phone or a wifi SIP phone ? Is anyone using the Pulver WiSIP phone ? Any comments ? How about the zyxel ? Thanks -- Kim Lux, Diesel Research Inc.___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Looking for a wireless phone... wifi or traditional wireless ?
Uniden and Vtech both just announced cordless phones with SIP ATAs built into the base station. You get better range and battery life compared to a WiFi phone. Jim James H. Thompson[EMAIL PROTECTED] - Original Message - From: Kim Lux To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Wednesday, January 12, 2005 5:49 PM Subject: Re: [Asterisk-Users] Looking for a wireless phone... wifi or traditional wireless ? An unflattering zyxel review:http://slacker.com/~nugget/asterisk3.phpI can't help but think my questions are out of place on this list... I'masking questions about SIP phones and everyone else is talking aboutasterisk. Sorry. On Wed, 2005-01-12 at 20:08 -0700, Kim Lux wrote: My wife wants a cordless phone for around the house. We are going to be using VOIP exclusively very shortly. Our current cordless phone is aged and on the verge of replacement. The other phone we are going to use is a SIP Budgetone. Should I buy a SIP to POTS converter and a new cordless phone or a wifi SIP phone ? Is anyone using the Pulver WiSIP phone ? Any comments ? How about the zyxel ? Thanks -- Kim Lux, Diesel Research Inc.___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: SV: [Asterisk-Users] www.voip-info.org
Commpartners (who provides hosting for voip-info.org) is doing a network upgrade tonight. Jim James H. Thompson[EMAIL PROTECTED] - Original Message - From: Luki To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Wednesday, December 01, 2004 8:53 PM Subject: Re: SV: [Asterisk-Users] www.voip-info.org Dead for me too.. I am in the US..Dead here too and I am in LA, next door to it (last hop commp-2.border17.lax.pnap.net).Maybe there are doing an upgrade... I recall their DB server was spitting out "too many connection" errors yesterday...--Luki ___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] asterisk newsgrup proposal or phpBB forum
Asterisk mailing lists are already setup as NNTP See: http://dir.gmane.org/search.php?match=asterisk Jim James H. Thompson[EMAIL PROTECTED] - Original Message - From: Corvin To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Monday, November 29, 2004 1:02 PM Subject: Re: [Asterisk-Users] asterisk newsgrup proposal or phpBB forum Dnia poniedziaek, 29 listopada 2004 21:32, Steven Critchfield napisa: Instead of starting a new flame over the stupidity of that, I'll point you to the archives via google to see how it is percieved. http://www.google.com/search?q=phpBB+site%3Alists.digium.comNice answer. Really what about *NNTP newsgrup*?It is somehow organized.I know that phpBB forums even those reasonable have bad opinion mostly in older users. But nntp? I don't want to make flames.BR,C.___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] T.38 support
T.38 is often put forward as the solution for reliable FAX over VOIP. Just wondering for anyone using T.38 (with any equipment), how well does it work ascompared to a FAX PSTN call? Thanks. Jim James H. Thompson[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Linksys RT31P2
Bruce Komito [EMAIL PROTECTED] wrote: If anyone finds the generic version of this available (i.e., not locked to Vonage), please advise the list of where. http://www.voip-info.org/tiki-index.php?page=Linksys ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Phones-Receptionist Setup
Gregory Junker [EMAIL PROTECTED] wrote: $400-500 device here. Not very price competitive. I would like to see less than half that. I agree that any touch screen ought to be able to do normal computer graphics. At this point, you are into normal LCD displays with touch capability, which I know retail over US$500 even for smaller ones. And Not all over $500 - a quick search finds: http://www.xenarcdirect.com/search_results.asp?txtsearchParamCat=6txtsearchParamType=ALLtxtsearchParamMan=ALLtxtsearchParamVen=ALLiLevel=1 Product ID: 700TSCategory: 7 LCD Monitor 700TS - 7' USB Touch Screen LCD Monitor with VGA input Description: 4-wire Resistive Touch Screen (USB); VGA Input × 1; Supports 640 x 480 ~ 1600 x 1200 display resolution; For PC, Server, GPS, and Standard VGA Use; On Screen Display Control; Available in Silver or Black Price: $429.00 Jim James H. Thompson [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] quasi-skype channel for Asterisk?
I created a wiki page for Skype gateways. (BSo far the only two I've heard about are the PCphoneonline and the Siemens (BGigaset gateway: (BIf more let me know or add to wiki: (B (Bhttp://www.voip-info.org/wiki-Skype+Gateways (B (B (BThanks. (B (BJim (B (BJames H. Thompson (B[EMAIL PROTECTED] (B (B___ (BAsterisk-Users mailing list ([EMAIL PROTECTED] (Bhttp://lists.digium.com/mailman/listinfo/asterisk-users (BTo UNSUBSCRIBE or update options visit: (B http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] WRT54GP2 (WiFi + ATA)
The generic non-locked version of the WRT54GP2 will be available in a "few weeks" according to Linksys sales. I assume that like the other Linksys unlocked VOIP products, its distribution will be restricted. For more information see: http://www.voip-info.org/tiki-index.php?page=Linksys Jim James H. Thompson[EMAIL PROTECTED] - Original Message - From: Aaron Clauson To: [EMAIL PROTECTED] Sent: Monday, November 08, 2004 10:29 PM Subject: [Asterisk-Users] WRT54GP2 (WiFi + ATA) Hi,If anyone has either:- Found a company which ships these units outside theUS,- Got one of the units and tried to unlock it fromVonage.Please post.(The Linksys WRT54GP2 is the first acceptably priced unit that has a router, WiFi and an ATA, at least thatI know of).Aaron__ Do you Yahoo!? Check out the new Yahoo! Front Page. www.yahoo.com ___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Embedded Asterisk Paper Complete
files mirrored on voip-info.org here: http://www.voip-info.org/tiki-index.php?page=Asterisk+embedded+systems Jim James H. Thompson [EMAIL PROTECTED] - Original Message - From: JR Richardson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, October 31, 2004 4:30 PM Subject: [Asterisk-Users] Embedded Asterisk Paper Complete Hi all, The journey is complete, at least for this project. http://lists.digium.com/pipermail/asterisk-users/2004-October/067289.html I spent the better part of Halloween putting this together, I hope it's useful, enjoy. My ftp server is on the fritz so feel free to post on any other user sites. If you have any difficulties, email me and I'll send the files to you directly. JR ftp://odyssey-tech.net/Embedded_Asterisk.doc ftp://odyssey-tech.net/Embedded_Asterisk.pdf ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Benjk's Question Why FXS
Reid A. Forrest [EMAIL PROTECTED] wrote: I may be wrong, but from what I've seen so far, an FXS port will run you about $100/port anyway, plus the cost of the analog device. At this price, I can't see any reason not to dump the analog and go with a cheap VOIP device. Even the lowest end (i.e. Grandstream) will give you more functionality than most analog phones at the same price. Now if you have a source for cheap or free channel banks, that's another story. ___ Sipura 200 ATA is $40/port Linksys PAP2-NA is $25/port Jim James H. Thompson [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Benjk's Question Why FXS
Reid A. Forrest [EMAIL PROTECTED] wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James H. Thompson Sipura 200 ATA is $40/port Linksys PAP2-NA is $25/port You're correct, I was looking at prices of _new_ channel banks instead of these devices. At $25 or $40 per port it could make sense to use FXS instead of VoIP. I haven't really followed the Linksys products; is the PAP2-NA commercially available unlocked? I thought they only sold these locked to Vonage. LInksys has restricted sales of the unlocked versions to Service Providers. Instructions and more info here: http://www.voip-info.org/tiki-index.php?page=Linksys Jim James H. Thompson [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Direct SIP connection to Vonage service
[EMAIL PROTECTED] wrote: Do you have a list of those providers that use IAX? Check the: Asterisk to/from PSTN services section on the wiki page: http://www.voip-info.org/wiki-VOIP+Service+Providers -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Benjamin on Asterisk Mailing Lists Sent: Friday, October 22, 2004 4:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Direct SIP connection to Vonage service On Fri, 22 Oct 2004 23:39:56 +0200, Stewart Nelson [EMAIL PROTECTED] wrote: I would appreciate your opinions on the feasibility of these techniques, and also about any other methods that have been tried to achieve direct SIP connectivity. If you are that desperate to use Vonage, then why don't you sign up for the secondary soft-client option which is $15 or so IIRC?! That will allow you to connect Asterisk directly to Vonage, although you pay extra for the privilege. I personally wouldn't bother and I wouldn't want to take my money to a company that uses a business model that I despise. So, vote with your wallet. Don't use Vonage. Use a true VoIP service. And while we are at it, support IAX: Use a provider that offers IAX. rgds benjk Jim James H. Thompson [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Bandwidth control on a home office network
Jay Milk wrote: Looks like it's time to add a WIKI page on QOS routing alternatives, listing options such as the Linksys WRT (with OpenWRT or Sveasoft or...), m0n0wall, LEAF, etc. It seems that this would be a bit off-topic, but QOS if very much a concern for VOIP. Any volunteers who'd actually know what they're talking about? I'm currently in the research phase of my next router-solution, since it's good-bye for my trusted 5861 soon. Qos in general http://www.voip-info.org/tiki-index.php?page=QoS Small/Home Routers with QoS http://www.voip-info.org/wiki-VOIP+Routers Please add information! Thanks. Jim ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Sending broadcasts to all phones?
Stan Brinkerhoff wrote: A friend of mine has a real panasonic PBX setup at his house, and is able to pick up the phone, dial an extension, and it broadcasts what he says over every phone in his house without the phones having to be picked up. What is this feature called? See: http://www.voip-info.org/tiki-index.php?page=Asterisk%20Paging%20and%20Intercom Its also possible the new Sipura phone will have this feature, but haven't seen documentation yet. Jim James H. Thompson [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cheap, Highquality IP Phones
Sipura just released their new IP phone - list price is $100 http://www.voip-info.org/tiki-index.php?page=Sipura Jim James H. Thompson [EMAIL PROTECTED] - Original Message - From: Tim Jackson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Friday, October 15, 2004 11:11 AM Subject: RE: [Asterisk-Users] Cheap, Highquality IP Phones Best bang for the buck out there are Polycom SoundPoint IP phones. We use IP500s. Pros: Pricetag (Cheaper than Cisco ~$180/phone) Quality (Built really well) Features (3 lines, XML Directory, DND, MWI, etc etc) Fairly straight-forward provisioning (Once you get the hang of it) Very very very configurable Cons: Confusing XML configurations No direct support from Polycom for Asterisk users No XML minibrowser on the IP500 -Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Boehm Sent: Friday, October 15, 2004 3:50 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Cheap, Highquality IP Phones I know that there is a list of phones on the wiki, but most of them are now out of date by months if not a year. Our whole office is using Cisco 7960s. Nice phones. Works great with asterisk. However, $300 each. If people could send the phone they use with asterisk, a quick pros/cons and its price, it would be appreciated. Basically, I am looking for a high quality $100 2-line SIP phone that supports g729 and works well with asterisk. Much appreciated, Matthew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cheap, Highquality IP Phones
Link to Sipura Press Release http://www.sipura.com/Documents/SipuraPressRelease007.pdf Jim James H. Thompson [EMAIL PROTECTED] - Original Message - From: Jay Milk [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' [EMAIL PROTECTED] Sent: Friday, October 15, 2004 12:15 PM Subject: RE: [Asterisk-Users] Cheap, Highquality IP Phones Where, when and how? $100 for two line appearances (hopefully allowing intercom-operation), and I'll get one for every room of the house. -Original Message- From: James H. Thompson [mailto:[EMAIL PROTECTED] Sent: Friday, October 15, 2004 4:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Cheap, Highquality IP Phones Sipura just released their new IP phone - list price is $100 http://www.voip-info.org/tiki-index.php?page=Sipura Jim James H. Thompson [EMAIL PROTECTED] - Original Message - From: Tim Jackson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Friday, October 15, 2004 11:11 AM Subject: RE: [Asterisk-Users] Cheap, Highquality IP Phones Best bang for the buck out there are Polycom SoundPoint IP phones. We use IP500s. Pros: Pricetag (Cheaper than Cisco ~$180/phone) Quality (Built really well) Features (3 lines, XML Directory, DND, MWI, etc etc) Fairly straight-forward provisioning (Once you get the hang of it) Very very very configurable Cons: Confusing XML configurations No direct support from Polycom for Asterisk users No XML minibrowser on the IP500 -Tim -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Boehm Sent: Friday, October 15, 2004 3:50 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Cheap, Highquality IP Phones I know that there is a list of phones on the wiki, but most of them are now out of date by months if not a year. Our whole office is using Cisco 7960s. Nice phones. Works great with asterisk. However, $300 each. If people could send the phone they use with asterisk, a quick pros/cons and its price, it would be appreciated. Basically, I am looking for a high quality $100 2-line SIP phone that supports g729 and works well with asterisk. Much appreciated, Matthew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/aster isk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cheap, Highquality IP Phones
www.voxilla.com is usually one of the first places to get the new sipura products, at least this has been true in the past. Jim James H. Thompson [EMAIL PROTECTED] - Original Message - From: Stan Brinkerhoff [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Friday, October 15, 2004 3:53 PM Subject: Re: [Asterisk-Users] Cheap, Highquality IP Phones Where do I buy one? Stan Matt Riddell wrote: James H. Thompson wrote: Link to Sipura Press Release http://www.sipura.com/Documents/SipuraPressRelease007.pdf I've put it up on the news page in HTML (just in case it takes anyone else as long as it takes me to open a PDF file!) The URL is: http://www.sineapps.com/news.php?rssid=230 Cheers, Matt Riddell ___ http://www.sineapps.com/news.php (Daily Asterisk News - html) http://www.sineapps.com/rssfeed.php (Daily Asterisk News - rss) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Running Asterisk on Linksys Router
At Astricon Mark mentioned that somone had Asterisk running on a Linksys Router. Anyone have more information on this? Jim James H. Thompson [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] GPL Violations (Was: Advice on OS Choice)
Brian West wrote: The EULA is where the real teeth are -- prohibiting even people who have purchased RHEL from using it in ways that RedHat prohibits. For example, it is not possible to purchase one copy of RHEL and install it on two machines. Nor are you allowed to run RHEL on a machine without having purchased support. I am unclear on how this is not a further restriction on the code (and therefore prohibited by the GPL) but the FSF appears unwilling to pursue the point. I do feel that those are violations of the GPL. They can't place more restrictions on software that is already free via the GPL. This is the exact reason I told RedHat to f$%k off. They used the community to build a brand then said F$%K YOU, but here is Fedora which we will use to test new stuff that might make it into RedHat's high end products. So basically you're a test community for future RedHat products if you run Fedora. Not to aruge one way or the other, but there are a number of free RH Enterprise work-alike distributions http://www.taolinux.org/ http://www.whiteboxlinux.org/ etc. Jim James H. Thompson [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] re: ATA units: anyone have these working with * or SER?
Hello list, please take a look at these units: http://www.tigerdirect.com/applications/SearchTools/item-details.asp?EdpNo=1048701CatId=1596 are they locked? does anyone have one working with asterisk or SER? Are these rebadged units from a different manufacturer? anyone have any experience good or bad with these? Here is some additonal information: http://home.businesswire.com/portal/site/google/index.jsp?ndmViewId=news_viewnewsId=20040914005648newsLang=en It appears that the boxes are intended for use with the VOIP2 service. There are fairly detailed manuals here: http://www.voip2.net/callbox.html Jim ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Database of world area codes
See this wiki page: http://www.voip-info.org/wiki-Numbering+plans Jim James H. Thompson [EMAIL PROTECTED] - Original Message - From: Storm D. J. Petersen [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Monday, October 11, 2004 7:28 AM Subject: [Asterisk-Users] Database of world area codes Hi, I'm looking for a database with all the world's country codes and area codes. Can anyone point me into the right direction? Cheers, S. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Intel Modem vs Digium Cards
Grandstreams are availabe for $65 quanity one, so its not hard to believe that you could get them for $55 for larger quantities http://froogle.google.com/froogle?q=grandstreamhl=enlr=tab=wfscoring=p Jim James H. Thompson [EMAIL PROTECTED] - Original Message - From: Steve Totaro [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Sunday, October 10, 2004 8:11 AM Subject: Re: [Asterisk-Users] Intel Modem vs Digium Cards i am still looking for the elusive $55 grandstreams. - Original Message - From: dean collins [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion [EMAIL PROTECTED] Sent: Sunday, October 10, 2004 1:46 PM Subject: RE: [Asterisk-Users] Intel Modem vs Digium Cards Lol, you're kidding right, go and look at what it costs to buy an alternative ip-pabx in comparison, and sorry but no corporate budget here, this is just a system for my home $100 on an old P3-700, and about the same on a card, and 2 $55 grandstream handsets along with some free sip softphone software. Hardly a fortune. On the other hand I think we are very fortunate that asterisk exists and cant help but get excited about where they will grow to. Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Wolf Paul Sent: Sunday, October 10, 2004 1:09 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Intel Modem vs Digium Cards dean collins [EMAIL PROTECTED] wrote: Rajeev, supporting Digium enables Asterisk to exist in the first place. Don't come asking for support here should you not be able to get these work alikes to operate correctly. I don't know Rajeev's situation, but here is mine: I am all for supporting Digium, and when I get ready to set up my production PBX I will buy their cards. However, those of us not working with hefty corporate budgets may not have the option of spending $100 for a test machine when there's a more cost effective option available. When I build my production machine, I will need multiple E1 ports; the FXO from the test machine will then land on my pile of no-longer-needed hardware. I'd rather use a $7 card for that than spend $100 which I will not be able to recover. (By the time I get two such Intel cards over here to Austria, I may well have spent $100 on shipping and customs charges, anyway). If that warrants don't come asking for support then you guys are not much of a community but a sales machine for Digium. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] IP Intercom's
There are several ways to approach this: * modify an existing SIP phone with Auto-answer (Grandstream for example) to interface with a loud speaker * use a SIP client (Asterisk for example) on a small PC and interface the sound card to a loudspeaker * use a traditional overhead paging/intercom hardware and interface to it via the sound card or via an FXS port. * use an analog auto answer door phone with an FXS interface Check these wiki pages for starting points: http://www.voip-info.org/wiki-Asterisk+Paging+and+Intercom http://www.voip-info.org/wiki-Asterisk+phone+door Jim James H. Thompson [EMAIL PROTECTED] - Original Message - From: Steve Maroney [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, September 18, 2004 8:41 AM Subject: [Asterisk-Users] IP Intercom's Im looking for an Intercom solution thats interoperable wit Asterisk. Ive read several posts about people using the 2nd lines on some SIP phones w/speaker phone. Unfortunatley I dont that is going to cut it in a large warehouse enviroment. Does anyone have a solution that uses a loudspeaker ? Thank you, Steve Maroney ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP/IAX2 phones with builtin magnetic stripe reader
Dinesh Nair wrote: hey * folk, need to tap the collective wisdom of this list for any details or pointers to vendors who manufacture/sell SIP or IAX phones with builtin magnetic stripe readers. these phones will be used in combination with * in a prepaid application. it would be advantageous if the mag stripe data was sent as DTMF over the SIP connection. any clues as to who manufactures/produces these phones ? Years ago it was common to integrate a credit card magnetic strip reader into phones for doing credit card authorizations. The industry moved to stand-alone authorization terminals years ago. The pay phone industry has been using phones with integrated magnetic strip readers, and you may be able to find some of these and use them with a SIP ATA. For example: http://www.payphone.com/shop/customer/product.php?productid=16148cat=252page=1 Here is a payphone with an ethernet interface: http://www.vending-usa.com/einpa.html You can also find magnetic card readers that have a DTMF output, you would need to integrate such a device with a phone of your choice. For example this reader has optional DTMF output: http://www.internationalbarcode.com/smagj.htm Another possibility would be to use a magnetic stripe reader that attached separately to an ethernet connection: http://www.computerwise.com/ethernet/ep210.html http://www.internationalbarcode.com/seriesj.htm If you used a soft client on a PC then you would have many different options for attaching a magnetic stripe reader: USB, RS-232, Keyboard wedge, built-in to keyboard, etc. Jim [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Lower cost router suitable for VOIP ?
Wiki page: http://www.voip-info.org/tiki-index.php?page=VOIP+Routers Feel free to create add to/update/create new pages. Jim James H. Thompson[EMAIL PROTECTED] - Original Message - From: Chris Shaw To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Friday, September 03, 2004 11:19 AM Subject: Re: [Asterisk-Users] Lower cost router suitable for VOIP ? How about the Wiki? :-)I think I'm gonna have to because it would be too long to e-mail! I can giveyou guys the short version though... -Chris___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] OT: Headset for Cisco 7960?
Started a Wiki page here: http://www.voip-info.org/wiki-Cisco+Phone+Headsets Jim James H. Thompson[EMAIL PROTECTED] - Original Message - From: Edward Eastman To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Sent: Tuesday, August 31, 2004 10:28 AM Subject: RE: [Asterisk-Users] OT: Headset for Cisco 7960? Cisco headset pinout is different from normal ones (grr)If it's just for you, (ie nothing too professional ;) you can snip the leadof an existing plantronics type headset and do some reordering - this willgive you the necessary info (sorry - can't remember exactly how I did it):http://www.mml.uni-hannover.de/einhorn/headset/index_e.htmlIf you're after something more professional then obviously one of theleads/adapters will be a better approach.HTHEd-Original Message-From: [EMAIL PROTECTED][mailto:[EMAIL PROTECTED] On Behalf Of Nate CarlsonSent: 31 August 2004 21:05To: [EMAIL PROTECTED]Subject: [Asterisk-Users] OT: Headset for Cisco 7960?Sorry, I know it's OT, but does anyone know of a relatively inexpensiveheadset that is compatible with the Cisco 7960?I've tried the headset off Norstar phones, doesn't seem to work with orwithout the amp.| nate carlson | [EMAIL PROTECTED] | http://www.natecarlson.com || depriving some poor village of its idiot since 1981 |___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___Asterisk-Users mailing list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Hardware for PBX with 4 incoming/outgoing lines and 20 phones
Scott Laird wrote: On Aug 25, 2004, at 9:57 AM, spectro wrote: IMHO, If you plan to use analog phones the cheapest is to buy a bunch of sipuras instead of TDM40B. (TDM40B = 4 FXS for $300, $75 each; sipura SPA2000 = 2 FXS for $100, $50 each) It might be worth looking at the new Linksys PAP2 -- it's only $50, and it's essentially a Sipura 2000. I haven't heard any first-hand revews, though. The Linksys PAP2 now available may be locked to Vonage. Anyone actually gotten one yet? Jim James H. Thompson [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] spamdsp capacity
Anybenchmarks or guesses onhow many simultaneous incoming FAXes spandsp could handle on a reasonable size server? Thanks. Jim James H. Thompson[EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP unphones
Jay Milk wrote: Does anyone know if there are additional SIP devices out there which aren't phones? I'm basically looking for a fully-automatic SIP speakerphone. I'd like to be able to dial a sip-extension and make an announcement (PA) and/or simply listen in to a room (baby-monitor). Yes, I know, some of the more advanced phones can be configured to behave like that, but it seems to a waste of money to have all those fancy displays and keys tucked away behind a speakergrille and drywall. BTW, I'm not dead-set on SIP, but it seems to be the most logical protocol for this app (NOTIFY msg can carry directions on mike/speaker/two-way, etc) Grandstream phone has this feature. Would be hard to find anything cheaper. Alternative would be to use an analog auto-answer phone attached to a SIP ATA. Would cost more than a Grandstream though. For more information see: http://www.voip-info.org/wiki-Asterisk+paging+and+intercom http://www.voip-info.org/wiki-Asterisk+phone+door http://www.voip-info.org/wiki-Analog+Telephone+Information Jim James H. Thompson [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP Phone recommendation for Receptionist
el Flynn wrote: Hi there, I've got an installation where there's 12 POTS line incoming into *, and am trying to get some insight as to which VoIP hard phone would be most suitable for this scenario. Other than the incoming lines, the receptionist would need the normal keyphone type stuff -- call pickup, park, hold, forward etc. What would you guys recommend? How about a touch screen LCD display running the Asterisk Flash Operator Panel? Or mabe a Tablet PC running Asterisk Flash Operator Panel? Jim James H. Thompson [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sipura partners with Linksys for new combo router/SIP ATA
Voxilla news story: http://voxilla.com/voxstory84-nested-order0-threshold0.html Two new products * A Sipura 2000 in a linksys box: Linksys PAP2 Phone Adapter * A combination NAT router with 2 FXS ports: Linksys RT31P2 Broadband Router Jim James H. Thompson [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Searchable Archive
Muiz Motani wrote: This brings up a good point that has had me scratching my head for a long time. Is there a good searchable archive of the asterisk mailing lists? I don't particularly want to download and keep updated the full 206 MB of the asterisk-users .mbx file on my laptop. The current format is just not searchable by keyword and a Google search does not work very well. For example, a search of the keywords opencall.org and asterisk-users on Google turned up nothing useful. See here: http://www.voip-info.org/tiki-index.php?page=Asterisk+FAQ Jim James H. Thompson [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] where to start asterisk sourcecode
Look at the "Developer Resources" section here: http://www.voip-info.org/wiki-Asterisk In particular this may help: http://www.voip-info.org/wiki-Asterisk+Understanding+the+source+code Jim James H. Thompson[EMAIL PROTECTED] - Original Message - From: Shanmuganathan Kumaravel To: [EMAIL PROTECTED] Sent: Thursday, July 29, 2004 3:10 AM Subject: [Asterisk-Users] where to start asterisk sourcecode Hi all,I would like to study the asterisk source code(Program). I dont' know from which file i've to start. can anyone helpme.RegardsShan.
Re: [Asterisk-Users] Re: Upgrade from Altigen
Thanks Jim. Does anyone think that the Altigen has this feature built-in or might they have a device similar to the Power Fail Bypass installed? The Altigen web site has technical manuals online. http://www.altigen.com/customer_tech-manuals.html It would appear that the power failure transfer feature is part of their hardware. I believe that in some places there are legal requirements that some ability to call emergency services remain during a power failure. Jim James H. Thompson [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Upgrade from Altigen
Currently with Altigen if their PBX goes down then calls will automatically route to some dedicated phones. Is there a way to configure this with Asterisk and the channel bank? You can use power fail transfer switches to do this. For example see: http://www.vikingelectronics.com/ Look at product: PF-6A Power Fail Bypass Jim James H. Thompson [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] * INSTRUCTIONS FOR THE ASTERISK COMMUNITY - PLEASE READ NOW *
I am on many mailing lists and lots of them have similar problems with people posting messages they could better answer themselves. Since many of these messages are from people posting for the first time, I think to some degree this is a failing of the mailing list structure itself. I've wondered if a mechanism like this would help: For the first N messages you post to the mailing list, your post does not automatically get posted. Instead you get a message similar to Olle's below, ending with something like: If you still want to send your message to the mailing list, just reply to this message Jim James H. Thompson [EMAIL PROTECTED] - Original Message - From: Olle E. Johansson [EMAIL PROTECTED] To: Users Asterisk [EMAIL PROTECTED] Sent: Tuesday, July 20, 2004 11:40 PM Subject: [Asterisk-Users] * INSTRUCTIONS FOR THE ASTERISK COMMUNITY - PLEASE READ NOW * Welcome to the Asterisk users community! Asterisk.org is a fast moving project. New code is added every day. Asterisk is the leading Open Source Telephony platform, with support both for classical telephony and IP telephony. Our community is also growing fast and we're having a lot of interaction, on the IRC and on the mailing lists. It's great to have you participating in this Open Source project - building an Open Source PBX. Here are a few things to know and remember while working with the project. ** The mailing list is growing The lead programmer of Asterisk, Mark Spencer at Digium, inc, recently wrote: The Asterisk community is growing at a remarkable pace. I know there are thousands of you out there -- in fact there are over eight *thousand* subscribers to asterisk-users alone, and almost one *thousand* registered users on the bug tracker. This means that everything anyone write to this mailing list, is sent to over 8.000 mailboxes that is already flowing over with messages. ** Think before sending a message, think twice I would like to stress the fact that you have to think before you send a message to such a big list. Do *not* send out personal replies on the list. If you offer services to someone, do *not* CC: or reply to the list, it will annoy more potential customers than get you new customers. If you send out a message by mistake, you don't have to apologize to all of us, we understand you're embarassed. We will get more annoyed by your apology than over your first message. ** Looking for or offering a commercial service? Use the asterisk-biz list for discussions on who offers what and for offering your business services. ** Try finding the answer first, then ask the list The Asterisk Wiki at http://www.voip-info.org project is an important knowledge base for the project. Go there to find your answer first, then search the mailing list archives (Google or http://search.voip-forum.com) and then go to the IRC channel. The IRC channel is populated with Asterisk gurus around the clock (literally) and they'll help you move forward. * IRC info: http://www.asterisk.org/index.php?menu=support#irc * There's many links to Asterisk web pages on the documentation page at http://www.asterisk.org * The Asterisk FAQ is found on the wiki http://www.voip-info.org/wiki-Asterisk+FAQ * The Asterisk documentation project (which needs your help) is at http://www.asteriskdocs.org Their handbook The hitchhiker's guide to Asterisk is already well worth reading. Finally, if you don't find the answer elsewhere, try the list. ** Mailing lists For developers, there is a developer's list, asterisk-dev. For BSD users (FreeBSD, NetBSD, OpenBSD and OS/X) there's a list called asterisk-bsd. There is also a business list for those that want to ask for commercial services and inform their community about new services. You'll find all lists on http://lists.digium.com, which is the site where you manage your subscription to this list as well. Please, do not crosspost the same message to multiple mailing lists. It will not help you, it will only add to the mail flow and get people that read both lists irritated. ** Reporting bugs If you think you have found a bug, report it. We need bug reports. Read this document http://www.digium.com/bugtracker.html and then go to the bugtracker http://bugs.digium.com to file a report. If you are unsure, find a bug marshal on the IRC channel to help you. They're appointed to support you with how to handle bugs. Please check the bugtracker thoroughly before posting a new bug; often, your bug or feature already exists but is simply slowly making it's way through the system. Duplicate reports slow things down for everyone, so please spend a few minutes searching first. The bug tracker is also a place where you add your contribution to Asterisk. If you have coded extra functionality, make sure you give it back to the project so it can be added to the code base
[Asterisk-Users] Looking for WiFi phone recommendations
Looking for WLAN - WiSIP - WiFi phone recommendations and experiences. What works, what doesn't. Thanks. Jim James H. Thompson[EMAIL PROTECTED]
Re: [Asterisk-Users] SIP phones recommendations
Consider the Uniden UIP200, I believe it meets all your criteria. http://www.voip-info.org/wiki-Uniden Jim James H. Thompson [EMAIL PROTECTED] - Original Message - From: Jean-Yves Avenard [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, July 14, 2004 9:47 PM Subject: [Asterisk-Users] SIP phones recommendations -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Dear all. We are currently using either Grandstream BT100 phones or SNOM 200. The BT100 comes with a 10mbit ethernet port and the snom with 2x100mbit port Problem with the SNOM is that they are expensive and I don't really like their design: often the handset slightly move on its base and it makes the whole thing unreliable: poor mechanical design in my opinion. Problem with the BT is that you need either a little switch to connect it or a spare ethernet port: very annoying and it needs far too many cables. Could you recommend a nice SIP phones (which works with * obviously) cheap, well-featured and reliable that comes with 2 x 100mbit port (so no need for a switch or an additional ethernet port). Can be powered over the Ethernet port so it's really cables-free Thank you Jean-Yves -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (Darwin) iD8DBQFA9jaLXeDVKqIr3GURApC8AJ0XiA1IyIiZlHCrCwtLriUwF9Kb2wCfXS08 0nafjBjZDIB/S8FybYv6YlE= =UsoD -END PGP SIGNATURE- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] spa-3000 review?
Sorry, no detailed HOW-TO's yet. This thing can obviously be made to do what I want of it, but it will be a while figuring it all out. This thing really needs a wiki devoted to it. ;-) Feel free to add as many pages on this topic as you wish to the wiki. Jim [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Starting up considerations.....
Is call quality affected by starting it differently? My belief is no. Regardless of how you start it, quality will be the same... Correct? Correct. Most interfaces are digital and unless a filter was introduced it would sound exactly the same. Also while running as root, there isn't any niceness problems either. I've found this not to be the case. On slow machines, turning on lots of debug output can seriously affect call quality. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Rotary phones? (No, I'm serious)
- Original Message - From: Steven Critchfield [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, July 13, 2004 12:25 PM Subject: Re: [Asterisk-Users] Rotary phones? (No, I'm serious) You will not likely find any device that converts pulse to tone though. Although it might be possible if it went through a channel that doesn't use pulse dialing like sip. Pulse to DTMF converter: See: http://www.voip-info.org/wiki-Analog+Telephone+Information ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Three (quick?) questions...
An ordinary T1 (non-ISDN) doen't have a separate channel for signalling. See: http://www.voip-info.org/wiki-T1 ISDN T1's have a separate signalling channel. Jim James H. Thompson [EMAIL PROTECTED] - Original Message - From: Dean Collins [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, July 10, 2004 6:52 PM Subject: RE: [Asterisk-Users] Three (quick?) questions... Hi Paul, you would know better than I would but I always thought a T1 was 24 channels of voice with the signalling additional like we have in Australia a Pri or E1 is 30 channels voice channels plus signalling. Can anyone else clarify? Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Mahler Sent: Sunday, 11 July 2004 2:39 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Three (quick?) questions... Hi, T1 is the carrier. T1 provides 24 D channels of 64Kbps each. Telephone companies provide ISDN (integrated services data network) on top of T-carrier. Two common flavors are BRI (basic rate interface) and PRI (Primary rate interface.) BRI provides two 64 kps channels, PRI provides 23 usable channels, the 24th is used for signalling. So--you can get phone calls over a T1 or over a T1 that is provisioned as a PRI. You can get 24 calls on a T1 and 23 on a PRI. A T1 has 24 channels. You can split, that is partialize, the channels between data and voice. You can do this with hardware outside the * server. Higher end Cisco routers, for example, support this. You can also use * and linux to partialize the T1. You better plan on spending a lot of time on making it work if you do it this way. You have to install the Linux packages to split the line. NON trival. Works great, though. Much less expensive, too. Paul Paul Mahler [EMAIL PROTECTED] Signate, LLC 665 Third Street Suite 100 San Francisco, CA 94107-1901 Asterisk Services and Training -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ken D'Ambrosio Sent: Saturday, July 10, 2004 8:33 AM To: [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Subject: [Asterisk-Users] Three (quick?) questions... [Please excuse if this is a repeat; I initially tried to send it from a different account, and it's been held up for a couple of days awaiting moderation.] 1) What's the absolute minimum required (hardware-wise) in order to get one in-bound POTS line into Asterisk, and then have IP phones inside? [In other words, I obviously need a NIC -- but what would be the bare-bones telco POTS interface?] 2) What phones would be recommended for inexpensive (doesn't even need LCD), and yet functional? 3) In order to share data and voice over a T1, does it have to be PRI? [I've got a T1 I could probably play with, but I'd like to be sure it'll... well, you know: work.] Thanks, Ken D'Ambrosio Sr. SysAdmin, Xanoptix, Inc. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk receives TMC Labs Internet Telephony Innovation Award
Asterisk receives TMC Labs Internet Telephony Innovation Award http://www.tmcnet.com/it/0704/tmclabs.htm Jim James H. Thompson[EMAIL PROTECTED]
Re: [Asterisk-Users] FINALLY! a good book about Asterisk.
From: William Boehlke [EMAIL PROTECTED] We encourage anyone who is not an Asterisk beginner to send $50 to the Wiki instead of buying the book, Nice suggestion. The Wiki is not really in need of donations at the moment. It is being fully sponsored by www.commpartners.us So I would encourage you instead of sending money, to spend some time adding or updating content on the Wiki (www.voip-info.org) I'd also like to acknowledge all of the many contributors that have made the Wiki into a useful resource. Thanks. Jim James H. Thompson [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Special Delivery from China
There are lots of GPL 8051 tools Try a google search: 8051 gpl For example: http://sdcc.sourceforge.net/ SDCC is a Freeware, retargettable, optimizing ANSI - C compiler that targets the Intel 8051, Maxim 80DS390 and the Zilog Z80 based MCUs. Work is in progress on supporting the Motorola 68HC08 as well as Microchip PIC16 and PIC18 series. The entire source code for the compiler is distributed under GPL. Jim James H. Thompson [EMAIL PROTECTED] - Original Message - From: Holger Schurig [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, June 30, 2004 9:09 PM Subject: Re: [Asterisk-Users] Special Delivery from China That would be a great alternative. For what it's worth, the phone is based on a PA1688 single-chip VOIP terminal, which in turn contains a 50MHz 8051-compatible and a ADSP2181 DSP running at 33MHz. Okay, open sourcers, that does not include Linux. Even uLinux (that runs on CPUs without a MMU) should be far to fat for this environment. Hey, that thing has even still Banks to access memory, very much like the Lotus EMS that we once used years ago on 8086 and 80186. Or in the Language Card for the Apple II ... For what it's worth, I was able to determine that they're using VC6 and KeilC51 (?) to cross-compile. Keil is a company that develops and sells cross-compilers for a host of embedded type CPUs. The compiler usually runs on Windows and generates binary files that you either flash into Flash chips, EEPROM or via JTAG. It's well known in the commercial community. The KeilC51 costs here 1600 Euro, and that's just the CA51 Compiler+Assembler. No debugger. I think that the No Linux and Windows words in my statemement above greatly reduces the chance that people really will jump onto this opensource bandwagon. The price tag as well (althought me might be able to create a 8051 cross compilation environment on Linux). ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Causing Cisco 7200 Router to Crash?
I've seen a similar problem caused by the Ethernet card in the server. Everytime there was any load, it would crash the Cisco. Changing to a different brand Ethernet card resolved the problem. Jim James H. Thompson [EMAIL PROTECTED] - Original Message - From: Brian Wilkins [EMAIL PROTECTED] To: Asterisk-users [EMAIL PROTECTED] Sent: Wednesday, June 30, 2004 11:05 AM Subject: [Asterisk-Users] Asterisk Causing Cisco 7200 Router to Crash? Hi, We are having an issue here. It seems that whenever we initialize Asterisk on our network, the router that the Asterisk server is connected to (Cisco 7200) crashes and loses it configuration. This has happended five times and each time we have tested it, it is always when Asterisk starts up. Has anyone else seen this problem? It is very odd because this is a very good router and we had the Asterisk server on an exact same router but different network before and it did not cause a crash. We have gone through two different Cisco 7200 series routers and both exhibited the same problems. Any clues? Thanks - -- -- Heritage Communications Corporation Melbourne, FL USA 32935 http://www.hcc.net ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Special Delivery from China
The one thing not available in any low cost SIP phone is auto-answer controllable via SIP. i.e. enable microphone only, enable speaker only, or enable both. Grandstream has an option to enable auto answer in its configuration screen, but since once enabled, it always auto-answers and it only has one line appearance, its not much use. You need this feature to do paging and intercom with handsfree talkback. See: http://www.voip-info.org/wiki-Asterisk+Paging+and+Intercom for more information. Jim James H. Thompson [EMAIL PROTECTED] - Original Message - From: Jay Milk [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, June 30, 2004 11:57 AM Subject: [Asterisk-Users] Special Delivery from China I received a sample IP/Speakerphone from my friends in China today. Asterisk setup was fairly uncomplicated and I had it running as an extension on my server within a few minutes. Sounds quality of both the receiver and the speakerphone are fine (wife's opinion). Are there any tests I should run with this phone? Following are the specs: - Single line appearance - Alpha display, 2x16 chars - Configurable by telnet and http; password protected - upgradeable by tftp - Protocols: sip, mgcp, h323 (only tested SIP) - DHCP or static address; support for NAT traversal - g729,g711u,g711a,g723 - configurable ringtones, user-downloadable ringtones - hold button works with asterisk - inband, rfc2833 dtmf-modes - second RJ45 PC port Physical: - 8.5 x 8.0 x 2.0 (WxDxH) - flat-black plastic - LCD can be tilted up for visibility up to about 80 degrees - Large number buttons, 13 function buttons (redial, volume, etc) Shortfalls: - Caller-ID Name doesn't seem to work - MWI doesn't seem to exist - Dialplan isn't very flexible (pretty much requires #/send for all numbers) - PC port *may* be legacy (10mbps), not FastEthernet (100mbps); not yet benchmarked Requested improvements: - Second line appearance - configurable soft-keys - distinctive ring w/ auto-answer I should be able to resell these for $75 in quantity, $80-$85 for samples/endusers. Any takers? Once the CID Name MWI are fixed, I'll set up a pre-order list, and if there's enough interest I'll import a batch. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Special Delivery from China
Another approach would be to sell the hardware without firmware and start and opensource project to build firmware for it. It would seem like this could be a good niche for a small manufacturing company. Jim James H. Thompson [EMAIL PROTECTED] - Original Message - From: Jay Milk [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, June 30, 2004 11:57 AM Subject: [Asterisk-Users] Special Delivery from China I received a sample IP/Speakerphone from my friends in China today. Asterisk setup was fairly uncomplicated and I had it running as an extension on my server within a few minutes. Sounds quality of both the receiver and the speakerphone are fine (wife's opinion). Are there any tests I should run with this phone? Following are the specs: - Single line appearance - Alpha display, 2x16 chars - Configurable by telnet and http; password protected - upgradeable by tftp - Protocols: sip, mgcp, h323 (only tested SIP) - DHCP or static address; support for NAT traversal - g729,g711u,g711a,g723 - configurable ringtones, user-downloadable ringtones - hold button works with asterisk - inband, rfc2833 dtmf-modes - second RJ45 PC port Physical: - 8.5 x 8.0 x 2.0 (WxDxH) - flat-black plastic - LCD can be tilted up for visibility up to about 80 degrees - Large number buttons, 13 function buttons (redial, volume, etc) Shortfalls: - Caller-ID Name doesn't seem to work - MWI doesn't seem to exist - Dialplan isn't very flexible (pretty much requires #/send for all numbers) - PC port *may* be legacy (10mbps), not FastEthernet (100mbps); not yet benchmarked Requested improvements: - Second line appearance - configurable soft-keys - distinctive ring w/ auto-answer I should be able to resell these for $75 in quantity, $80-$85 for samples/endusers. Any takers? Once the CID Name MWI are fixed, I'll set up a pre-order list, and if there's enough interest I'll import a batch. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Ruggedised IP Phone
- Original Message - From: Matt [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, June 29, 2004 12:42 AM Subject: [Asterisk-Users] Ruggedised IP Phone Hi all, I want to use my * box to control entry to a building. I was wondering who else has done this and what phones they might recommend. The phone itself needs to be externally mounted so will have to be durable. Functionally I would like it to just dial and extension when picked up. See: http://www.voip-info.org/wiki-Asterisk+phone+door Please add any new info you find. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Cheap (US$120 or less) SIP Phones
- Original Message - From: Chris Hirsch To: [EMAIL PROTECTED] Sent: Friday, June 25, 2004 6:56 AM Subject: Re: [Asterisk-Users] Cheap (US$120 or less) SIP Phones James H. Thompson wrote: Are there any online retailers that carry the Uniden UIP series phones? I did a quick Froogle search to no avail. See: http://www.voip-info.org/wiki-Uniden So you *must* sign up as a reseller to purchase one? What are your opinions/problems on the UIP-200? It looks like a pretty good phone for a reasonable price.Todd at Teledynamics (see wiki page mentioned above) has been very responsive to email, and we did not need to sign up as a reseller to purchase the Uniden phones.
Re: [Asterisk-Users] Cheap (US$120 or less) SIP Phones
Are there any online retailers that carry the Uniden UIP series phones? I did a quick Froogle search to no avail. See: http://www.voip-info.org/wiki-Uniden Jim James H. Thompson [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] T1 Failover switch?
http://www.voip-info.org/tiki-index.php?page=Failover%20switches Jim James H. Thompson [EMAIL PROTECTED] - Original Message - From: Chris A. Icide [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, May 28, 2004 11:48 AM Subject: [Asterisk-Users] T1 Failover switch? Somewhere in the last 4-5 moths someone on this list posted something about a 'failover' switch for T1's. Basically, a box that would receive the T1 from the network provider and have the ability to output that T1 to a primary device and when that primary device goes into alarm, immediately switch over to a secondary device. I even believe they may have posted a link to a site that sells the things. However, I've not been able to google the right keywords to force the post to pop up in a search. So if there is anyone out there who can point me in the right direction (towards one or more of these devices), I would greatly appreciate it. -Chris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Wiki down
Sorry for the downtime. There was a configuration problem on the server that went unnoticed for a few hours. I have no objection to mirrors, but all of the pages are dynamically generated and the software the wiki is currently running on doesn't provide any easy way to create and/or update mirrors. Jim James H. Thompson [EMAIL PROTECTED] - Original Message - From: Gregory Junker [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, May 27, 2004 2:44 AM Subject: [Asterisk-Users] Wiki down http://www.voip-info.org gives: Warning: mysql error: No Database Selected in query: select `name` ,`value` from `tiki_preferences` in /var/www/html/tikiwiki-1.8.2/lib/tikidblib.php on line 133 Values: Array ( ) $result is false $result is empty Was going to grab a link to give to Florent regarding his CTI thread and question about how to program against the Asterisk API... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Premisys Slimline CB
Seems like an array of Sipura 2000's is price competitive, especially if you take into account the cost of the T1 card. Jim James H. Thompson [EMAIL PROTECTED] - Original Message - From: Michael Welter [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, May 20, 2004 5:57 AM Subject: [Asterisk-Users] Premisys Slimline CB I need to connect a bunch of analog telephone sets. Does anyone have any comments about this channel bank? Disconnect supervision? Echo? ADSI problems? The price is right @ $995 new and $695 refurbished. Thanks, -- Michael Welter Introspect Telephony Corp. Denver, Colorado +1 303 674 2575 [EMAIL PROTECTED] www.introspect.com ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Ok, Im confused
Look here: http://www.voip-info.org/wiki-Asterisk+How+to+connect+to+FWD Jim James H. Thompson [EMAIL PROTECTED] - Original Message - From: tmpm [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, April 21, 2004 11:50 AM Subject: [Asterisk-Users] Ok, Im confused Im totally a newbee at * Im confused. Ive got a FWD account, and it works on the winboxen. Ive got * up and can do the echotest etc, so its working. I want to get FWD working, and all the pages ive seen on setup are most confusing. Is FWD setup like IAXTEL? Do i plug in my FWD info in the same places as the IAXTEL stuff? Ive been trying for a week now, and Im more lost than before. Ive got a Internet phonejack card in the penguin, phone0, and all I want to do at this point is make and receive calls thru FWD using that jackIll plug the house in later...Ill learn the other stuff later. No voicemail, no BS, no dial thru least cost routing, or nightlines just make it work as a phone. Im either more stupid than I think, or Im missing something major here. Ive got to the point the CLI shows me connected to FWD fine.(I think) Sip show users Username Secret Authen Def. Context a/c fwd.pulver.com secret md5,plaintext default no Need some basic, stupidly simple scripts here...I dont need or want to dial 1-700 or *9 or any other crap, just make it work like the stupid winbox phone for now...Ill read the wiki for a couple years, and then maybe I can do voicemail or whatever... frustrated...and I know its showing...sri ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Voicemail storage in DB
No one solution is going to be best for everyone. It would be nice if there was a clean interface for voicemail storage so it was easy to plugin whichever scheme best fit your requirements. Jim James H. Thompson [EMAIL PROTECTED] - Original Message - From: Brian Capouch [EMAIL PROTECTED] To: [EMAIL PROTECTED] Cc: Shumard Kenneth Charles [EMAIL PROTECTED]; Tony Braner [EMAIL PROTECTED] Sent: Tuesday, April 13, 2004 7:17 PM Subject: Re: [Asterisk-Users] Re: Voicemail storage in DB Matt White wrote: James H. Thompson wrote: Would it make any sense to store the voice mail formatted as a email msg in a Maildir directory structure. Then you could also retreive them with an email client. As an extension of this thought, how about going one step further and storing the voicemail on an imap server directly? It would remove the whole storage question and allow storing on a remote system. The tools in the WU-IMAP c-client package would be pretty useful... Two of my students are working on this very project right now. Some thoughts/caveats: One design constraint that is mostly enforced in the asterisk code is that it run standalone (I think the mpg123 dependency is the sole exception) and there was very strong sentiment that anything we do NOT require the installation of a whole IMAP suite. So that complicates things somewhere between a little and a lot. Basically the task is to design a maildir type entity that can be completely manipulated within the asterisk application itself. We're still not coding it heavily, but my sense is that the real gotcha is going to lie in the IVR access routines. They'll have to be mapped into IMAP-space, as I see things right now, and in looking at the code that's already there, that isn't going to be a trivial thing. There is also the split between the UW orientation (keep the files in the maildir owned by the user they're sent to) and the cyrus orientation (lock down the IMAP store and require all access to pass through a server agent). I think the maildir approach is the Correct One, but the path thither appears to be a least minorly studded with complexity. My HO, of course. Thx. B. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Voicemail storage in DB
WipeOut == WipeOut [EMAIL PROTECTED] writes: WipeOut Have you thought of mounting the spool directolr on an NFS WipeOut file server ... [I am] not sure if there would be any file WipeOut locking issues.. Yes, there would be. This is the same issue as using nfs mail spools with maildir style storage. W/o locking there is no way to guarantee that two servers do not create the same vm file on top of one another. You could use NFS with the Maildir alogrithm or something similar to avoid the need for locking. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Voicemail storage in DB
Would it make any sense to store the voice mail formatted as a email msg in a Maildir
directory
structure.
Then you could also retreive them with an email client.
Jim
James H. Thompson
[EMAIL PROTECTED]
- Original Message -
From: James H. Cloos Jr. [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, April 12, 2004 11:30 AM
Subject: [Asterisk-Users] Re: Voicemail storage in DB
James == James H Thompson [EMAIL PROTECTED] writes:
James You could use NFS with the Maildir alogrithm or
James something similar to avoid the need for locking.
Here is an(other) idea if anyone is looking for a project:
When using a db for the meta data, there is no need for the filenames
to have the message number in them. As such, one can use a mix of the
${UNIQUEID}, (the hostname or ip if a networked fs), and any other call
details of interest as the filename. Or, for even nearer certainty of
collision-proof-ness you can use the base64 of the sha hash of some
entropy and the call's ${UNIQUEID} for the filename.
The message order can be kept in the db table with the rest of the
meta data.
-JimC
(don't you love neologisms)
be sure to use a filesystem-friendly version of base64
--
James H. Cloos, Jr. [EMAIL PROTECTED] http://jhcloos.com
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[Asterisk-Users] VON show report
I wrote up report on products that caught my interest at the VON show going on this week in Santa Clara. http://www.voip-info.org/wiki-VON+Spring+2004+Report Jim James H. Thompson[EMAIL PROTECTED]
Re: [Asterisk-Users] IAX2 as an IETF Standard?
No guarantee then when public IPs match that clients are both on same NAT LAN. Client A 192.168.0.1 - NAT Router A - NAT Router X with Public IP 123.123.123.123 --- Internet Client B 192.168.0.1 - NAT Router B -| Jim James H. Thompson [EMAIL PROTECTED] - Original Message - From: Adam Hart [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Wednesday, March 24, 2004 2:26 PM Subject: Re: [Asterisk-Users] IAX2 as an IETF Standard? Robert Hajime Lanning wrote: quote who=Adam Hart I also like to see two people behind the same nat being able to communicate directly (without requiring pin-wheeling). Ie The client attaches their private ip to the register packet, which is used when client A B's public ips match. 192.168.1.0/24 -- NAT-BOX -- Internet -- NAT-BOX -- 192.168.1.0/24 | | | IAX phoneAsterisk-Box IAX phone umm... I would suggest the default setting to be off, as the above topology would be very common. from my post: which is used when client A B's public ips match. meaning in this situation both clients would have different public IPs and it wouldn't be used. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Simple Front Gate Intercom
In a nutshell: Can I use Asterisk to hook up an intercom at my front gate? My wife would like to have one of those simple speaker/microphone intercoms. People show up at our front gate, press the doorbell, it rings in the house. We pick up a phone on my Asterisk system and dial (example) 105 to connect to the intercom and say, Who's there? The dipweed at the gate leans forward, and speaks into the same speaker he heard the voice come from and says, Joe Sixpack. http://www.voip-info.org/wiki-Asterisk+phone+door Jim James H. Thompson [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Simple Front Gate Intercom
Some of the door phone systems are designed to share an already existing line. For example: http://www.sandman.com/pdf/Page21.pdf I believe some of the Viking configurations can do this too. For example see the diagram here: http://www.vikingelectronics.com/products/pdf/c-2000(dbb).pdf Jim James H. Thompson [EMAIL PROTECTED] - Original Message - From: Greg Kedrovsky [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Thursday, February 26, 2004 10:48 AM Subject: Re: [Asterisk-Users] Simple Front Gate Intercom On Thu, Feb 26, 2004 at 12:16:06PM -0800, TC wrote: http://www.vikingelectronics.com/products/app-notes/doorboxes.html The W-1000, W-2000A and W-3000 doorboxes are designed to be installed on the unused telephone line input of nearly any phone system or... Key word: input. My telephone line input is my x100p fxo card, and it is a ONE-port card. I have no unused line input on my phone system. Therefore, I'm hosed with these models??? Help me... I've fallen, and I can't get up... -gk -- Mutt 1.4.1i on Slackware 9.1 Linux Curridabat, San Jose, Costa Rica http://www.greg-and-sue.com/screenshot.jpg Yahoo Instant Messenger ID: gregkedro ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] VideoPhone
See the video phone section here: http://www.voip-info.org/wiki-VOIP+Phones Jim James H. Thompson [EMAIL PROTECTED] - Original Message - From: Isamar Maia [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Saturday, February 14, 2004 1:31 AM Subject: [Asterisk-Users] VideoPhone Hi folks, Anybody knows a Grandstream-linux VideoPhone... I mean, proportionaly the same price and quality. Anybody knows? Thanks, Isamar ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Intercom system (not paging system)
There is an auto-answer speakerphone that might do what you described: http://www.vikingelectronics.com/products/apartmententry/k-1700-3(rd).html Jim James H. Thompson[EMAIL PROTECTED] - Original Message - From: David Schumann To: [EMAIL PROTECTED] Sent: Monday, February 09, 2004 9:57 AM Subject: [Asterisk-Users] Intercom system (not paging system) I've looked around and found previous discussion about this, but so far I have not seen any answers that really solve this problem.I'd like to integrate an intercom system into Asterisk so that users could dial an extension, the phone on the other end would emit a beep, and then the speakerphone would activate letting two people have a conversation without the personat the extension picking up the phone. This is a huge benefit for an office/warehouse environment.I know that this can be programmed with certain Cisco phones (set to autoanswer), but my problem is that the phones that I want to do this with are down in our warehouse and I think the Cisco phones would be stolen quickly. I've also thought about a writing a software solution with microphone and a set of speakers, but, again, the computers would probably end up getting stolen.A $20 analog speaker phone would work great if I could wire the speaker phone to pick up the line automatically on ring and then hang up the line on disconnect. Anyone know how to change the wiring to get it to work? David Schumann [EMAIL PROTECTED]Filter Products CompanyP.O. Box 13068Richmond, VA 23225804-231-4646, phone804-233-3912, faxhttp://fpcfilters.com___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Intercom system (not paging system)
This may be away to do what you described with a $20 speaker phone and no phone modifications: Using the speaker phone dial-in to a conference room on the asterisk Then whenever anyone wants to call this extension, you can route their call to the conf room and they can have a two-way conversation. Possible Downsides: * ifcall from the conference room gets terminated for any reason, then it will have to be re-established manually. * resources consumed to keep conference room open and call always active Possible Advantages: * phone only in auto-answer when you put it in that mode * phone usable to make and receive normal calls Jim James H. Thompson[EMAIL PROTECTED] - Original Message - From: James H. Thompson To: [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Sent: Monday, February 09, 2004 1:55 PM Subject: Re: [Asterisk-Users] Intercom system (not paging system) There is an auto-answer speakerphone that might do what you described: http://www.vikingelectronics.com/products/apartmententry/k-1700-3(rd).html Jim James H. Thompson[EMAIL PROTECTED] - Original Message - From: David Schumann To: [EMAIL PROTECTED] Sent: Monday, February 09, 2004 9:57 AM Subject: [Asterisk-Users] Intercom system (not paging system) I've looked around and found previous discussion about this, but so far I have not seen any answers that really solve this problem.I'd like to integrate an intercom system into Asterisk so that users could dial an extension, the phone on the other end would emit a beep, and then the speakerphone would activate letting two people have a conversation without the personat the extension picking up the phone. This is a huge benefit for an office/warehouse environment.I know that this can be programmed with certain Cisco phones (set to autoanswer), but my problem is that the phones that I want to do this with are down in our warehouse and I think the Cisco phones would be stolen quickly. I've also thought about a writing a software solution with microphone and a set of speakers, but, again, the computers would probably end up getting stolen.A $20 analog speaker phone would work great if I could wire the speaker phone to pick up the line automatically on ring and then hang up the line on disconnect. Anyone know how to change the wiring to get it to work? David Schumann [EMAIL PROTECTED]Filter Products CompanyP.O. Box 13068Richmond, VA 23225804-231-4646, phone804-233-3912, faxhttp://fpcfilters.com___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] http://www.oneunified.net
Should we create a area in the WIKI for all of the VOIP providers so we can leave comments about them someplace, and not take up mailling list time? Many of the providers already have a page on the Wiki. (You can create one if not) Please feel free to add comments to these pages about your expericences using them. http://www.voip-info.org/wiki-VOIP+Service+Providers Jim James H. Thompson [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Nortel and Asterisk interconnection
OK its on the wiki: http://www.voip-info.org/tiki-index.php?page=Asterisk+Interop+Nortel+Norstar+MICS Thanks. Jim James H. Thompson [EMAIL PROTECTED] - Original Message - From: David Gomillion [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, February 03, 2004 6:23 AM Subject: [Asterisk-Users] Nortel and Asterisk interconnection I have created a pdf document about my experience in integrating a Nortel Norstar MICS with *. This is not a cookbook, but it does describe the process I followed and gave a copy of the relevant configuration files. If anybody is interested, please feel free to download a copy at http://www.eyecarenow.com/asterisk. Please be patient, as the Internet connection here is, well, lacking. If anybody finds this useful and would like to mirror it, please let me know. Thanks, David Gomillion ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] The Smallest Asterisk Server Ever?
Check here for list of small Asterisk implementations mentioned on the mailing list. http://www.voip-info.org/wiki-Asterisk+setup+minimum Jim James H. Thompson [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] The Smallest Asterisk Server Ever?
http://nlug.org/mail/nlug__2003_12/0094.html Jim James H. Thompson [EMAIL PROTECTED] - Original Message - From: Panny Malialis [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, February 03, 2004 10:58 AM Subject: Re: [Asterisk-Users] The Smallest Asterisk Server Ever? I cant wait to see the asterisk on an xbox page!!, but the link seems broken http://nlug.org/mail/nlugb2003_12/0094.html I've tried removing the b with no luck Anyone know what the link should be ? Thanks Panny - Original Message - From: David J Carter [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, February 03, 2004 8:31 PM Subject: RE: [Asterisk-Users] The Smallest Asterisk Server Ever? Hey I don't know, I paid ?100 ($170) for my XBox, couldn't get a PC for that. The Linux bit is all free, and only a couple of PCB work to disenable the protection. Dave -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Chris Albertson Sent: 03 February 2004 18:01 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] The Smallest Asterisk Server Ever? I read a report of Asterisk running on a Microsoft X-Box. That's kind of a stunt as you could buy a decent PC for the price of a Linux-capable XBox. Id's still like to see Asterisk run on very low-end hardware The Snom IP phone runs Linux inside? I assume as Linux is GPL'd Snom will supply the source code? It would be fun to install an Asterisk server in a phone. --- Panny Malialis [EMAIL PROTECTED] wrote: Does anyone have it running on a Cyclades T100 ? same as used for ntop/nbox. I was thinking of using that as an IAX-sip translator for offices with NAT. CPU MPC855T (PowerPC Dual-CPU) Memory 32MB RAM / 4MB Flash (TS100) Interfaces1 Ethernet 10/100BT on RJ45 1 RS232 Console on RJ45 RS232 Serial Ports on RJ45 Looks like fun! Although a little lacking on memory. Any comments? Panny ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! SiteBuilder - Free web site building tool. Try it! http://webhosting.yahoo.com/ps/sb/ ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Incoming SIP matching
I ran some tests and reviewed the source code. It appears that for incoming INVITE messages, Asterisk first checks for [name] entries that match the user portion of the SIP URI in the From: header of the INVITE message.. i.e. if you are calling From sip:[EMAIL PROTECTED] it looks for [123] in the sip.conf file. If this fails then it checks for an IP match. If the IP match fails then it looks in the extensions.conf file (in the context set as default in sip.conf) for a matching extension. If I've intereperted it correctly, it seems a strange way for it to operate. Adding some debug log messages about which sip.conf entry is being selected would make figuring out what is happening a lot easier. Jim James H. Thompson [EMAIL PROTECTED] - Original Message - From: Thomas B. Clark [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Sunday, January 25, 2004 3:01 AM Subject: [Asterisk-Users] Incoming SIP matching Incoming FWD calls from other FWD users, iaxtel, or via ipkall, need to have dtmfmode=rfc2833. However, incoming FWD calls from the dialup access numbers (such as libretel) need to have dtmfmode=inband. To solve this problem, I created a second FWD account and configured sip.conf as follows, in order to match the incoming number to the proper dtmfmode: [fwd-rfc] type=friend secret=* host=dynamic dtmfmode=rfc2833 username=76153 callerid=CLARK THOMAS B 76153 [fwd-inband] type=friend secret=* host=dynamic dtmfmode=inband username=244006 callerid=CLARK THOMAS B 244006 What I find is that, no matter what I change (for example, host-dynamic in order to prevent matching by IP address), I cannot make the incoming SIP calls match successfully. With the configuration above, all incoming calls use dtmfmode=rfc2833, but that could be because it's the default. Either entry works correctly alone (with the other commented out.) I found some discussion in the archives about incoming sip matching, but no patches. Is there a better way to handle the two types of incoming FWD calls? If not, is there something else I could change in order to make them match the correct section? Any ideas would be appreciated. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Mediatrix 1204 sip experience?
See: the ATA section of: http://www.voip-info.org/wiki-VOIP+Phones and: http://www.voip-info.org/wiki-VoIP+Gateways for a list of what is available. Jim James H. Thompson [EMAIL PROTECTED] - Original Message - From: Rich Adamson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, January 23, 2004 12:34 PM Subject: Re: [Asterisk-Users] Mediatrix 1204 sip experience? I'm not sure I understand your english here. I have two x100p's working just fine, but I've got a couple more pstn lines I'd like to connect up. I probably could put another one in the system, but I'd rather use a 4-port external gateway that works well if such a thing exits at a reasonable price. (No, I don't want channel banks and T1 cards for such a simple environment.) I'm just starting to do the research on what is actually available. Is it so hard to put X100P as a ethernet device? I have been trying FXO devices, but gets me luck. Kannaiyan - Original Message - Anyone had any good/bad/otherwise experience with the Mediatrix 1204 Sip FXO 4-port gateway? The archives tend to suggest the box is not very straight forward, and possibly lacks some basic pstn interaction features. Thinking about trying one in place of a pair of x100p's (functioning fine now). CallerId, etc, supported on this gateway? Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ---End of Original Message- ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ultra-cheap asterisk box
FAQ for Dell 400SC: http://www.aaltonen.us/forums/viewtopic.php?t=8 Jim James H. Thompson [EMAIL PROTECTED] - Original Message - From: calvis [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Friday, January 16, 2004 10:33 AM Subject: RE: [Asterisk-Users] ultra-cheap asterisk box I got in on the same Dell deal I think. You must hang out on the bargain boards just like I do? I hang out mainly at fatwallet.com. This is the thread that I got in on the Dell machines that I just recently purchased. http://www.fatwallet.com/forums/messageview.php?start=920catid=24threadid= 264777 I found out by another 400SC user and you can not control assign interrupts on the PCI slots on this machine. Does that point bother you if you are going to run this unit with *? I want to put 3 X100P cards and 1 TDM400P in my up coming 400SC, but not sure if I will have conflict if I use up all the PCI slots in the machine. Charles Alvis Internet Technology Group Redmond, WA -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Mahler Sent: Thursday, January 15, 2004 4:54 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] ultra-cheap asterisk box I have a Dell 400sc sever on order. It will be shipped on the 27th. It is a 2.4GHz P4 with a 533 MHz front side bus, a 40GB disk, 128MB of memory, sound card, ethernet, and year of on-site next day maintenance. It is $318 delivered after rebates. Yes, $318. This is a real server, by the way, not a desktop machine. It also makes NO noise. I can't hear a thing with my ear right next to it. Why would you even THINK about getting anything else? Paul Paul Mahler mail:[EMAIL PROTECTED] phone: 650.207.9855 fax: 877.408.0105 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Chris Albertson Sent: Thursday, January 15, 2004 9:32 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] ultra-cheap asterisk box I'm looking to do about the same thing, build very low cost systems. (I'm looking at putting Asterisk at some non-profit organizations.) but one thing you can't make a compromise on is reliabilty. It has to work and keep working for years to come. I was able to keep the price of a new PC to about $300 ad still use an ASUS mainboard and an AMD XP2600+ The trick is to add absolutly nothing not needed. No floppy, no CDROM so you can run off a 200W P/S. Next I'll experiment with a notebook sized IDE disk drives and to see if _underclocking_ the CPU reduces it's power comsumption enough that we can save one fan. Ideally Asterisk will be ported one day to Linux/ARM or some other very low cost platform. for VOIP you do not need the PCI slots. In theory Asterisk could run on a Lynksys router box with re-flashed EEPROM. After all Lynksys' latest wireless router runs Linux inside Low cost to me means low total cost of ownership To get this I don't think buying the lowest priced parts is the way to go. I want quality mainboard, and a quality power supply and, this is importernt: A low internal case temperature. for this reason I'll spend the extra $50 to go with Antec cases and ASUS mainboards over the generic ones. What I'm finding is that the PCs are so cheap that the cost of electric power to run them is now a large part of the cost. (assume 0.20/kwh times 200W times 365 days = $350. So you pay for the PC again every year in electric power to run it. Worse. In an office with airconditioning _all_ of that PC's 200W goes to heat and your A/C unit will use about 220W of power to remove that 200W of heat.) and at a small office they will not have a server room so noise from the fan is an issue. --- Roy Sigurd Karlsbakk [EMAIL PROTECTED] wrote: hi all what about this... I just put together a box on a web shop (komplett.no) that will cost me NOK ~1850 (¤ 216) plus a small ¤50 drive and cables, so say ¤300. This consists of a cheap MB with a duron 1400, 256MB SDRAM and two HFC-PCI cards (if capijod will finish off the zaptel-driver soon). This is all in a cheap PC case. What do you think? Should this be doable? as a product? With only IP phones and potentially a fax solution? any ideas? thanks roy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users = Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __ Do you Yahoo!? Yahoo! Hotjobs: Enter the Signing Bonus Sweepstakes http://hotjobs.sweepstakes.yahoo.com/signingbonus ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com
Re: [Asterisk-Users] Doorbells Door Intercoms
The combination of a cheap ATA and standard doorphone would seem easier (and maybe cheaper). For door phone hardware check out: http://www.vikingelectronics.com/ Jim James H. Thompson [EMAIL PROTECTED] - Original Message - From: Adthrawn [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Tuesday, January 06, 2004 11:45 AM Subject: [Asterisk-Users] Doorbells Door Intercoms Hi, Does anybody know of a VoIP compatible doorbell or door intercom unit? I've contemplated buying a cheap SIP phone, ripping it apart, and putting it inside an IP66 sealed unit... It would need: - At least one speed-dial key, or some way to make every button dial the same extension number - PoE (power over ethernet), so I can power it off the central switch - cheap enough to rip apart Any ideas? Also, is there anyway in the Asterisk config, to ensure anything dialed from the phone in turn just dials a specific extension? What I'm thinking: Door bell button rings extension , which rings all phones. Caller ID is Front Door Entry or whatever, and the Cisco's Dial Plan has a doorbell chime as the ring tone for this specific extension filter (so has it's own ring tone). There is a time out of 5 to 10 seconds, which represents the time it takes for a normal doorbell to chime and stop. Now, the intercom bit... If I want to find out who it is before running downstairs, I simply pickup my receiver and ask them. Otherwise, I just leave it and the ringing stops, and the call is terminated. Also, if I want to find out who is outside, I can just dial a special number, say 8889, and I connect to the phone outside. Now, can the phone be set to automatically pickup a call?! Or perhaps, If I find a Cisco 7940 with a damaged case on eBay, I can rip it apart, and then use the SIP/SCCP Intercom feature on that phone only?! Does this sound sensible/achievable?! Best, Ad. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] RE: Inexpensive Analog Ports
2 sipuras SPA2000, sold at 100USD each they have 2 FXS ports its like cisco ata While this is certainly a step in the right direction, it needs to get cheaper to compete with existing solutions. For example if you were to configure a 6x24 (6 FXO ports, 24 FXS ports) Panasonic PBX with voice mail, it comes out to approx $48 per port including everything except the phones. The FXS ports support both ordinary analog phones and Panasonic system phones. Reference: http://www.twacomm.com/Catalog/Model_KX-TA624-5.htm Panasonic KX-TA624 PBX KX-TA624 System Unit $420 with 3 FXO and 8 FXS ports KX-TA62477-3 3x8 expansion $300with 3 FXO and 8 FXS ports KX-TA62470-2 0x8 expansion $230 with 8 FXS ports KX-TVS50 2 port Voicemail $500 Total$1,450 with 6 FXO ports24 FXS ports Total Ports 30 * Cost per port$48 * voice mail interconnect may use two of the ports Jim [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
