Re: [asterisk-users] can anybody tell me how Magic jack can be so cheap ????
Or more accurately, they believe they can follow the NetZero or Juno model (Free in exchange for ads being pushed to you). -forrest C F wrote: They believe they have advertisement revenues. On Sat, Feb 7, 2009 at 5:45 PM, Ignacio Ortega A. nachom...@gmail.com wrote: How Magic Jack can only charge $20 per year? do they have a call limit? do they have a call duration limit or limit of minutes per day?, Thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI/T1 data rate...
Steve Totaro wrote: B-chans should be 64k. That is a strange question indeed. For PRI, agreed. This is, however, a common question when provisioning channelized T1 services, since the B channels on robbed-bit T1's are really only 56K since the lowest bit is robbed for signalling. -forrest ___ --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] PRI/T1 data rate...
Steve Totaro wrote: I knew someone would have an explanation that makes sense. I have NEVER done anything but PRI from the Telco. Wouldn't the question of signaling and switchtype negate the need to ask for data rate? Yes. But these are probably telco ordering droids, meaning that all they know is that they have to fill in the blanks. I recently ordered a LD PRI from a carrier. I wanted PRI, switchtype either 5ESS or preferrably National. The order got kicked because I didn't specify whether or not I wanted EM and which type of em (immediate, wink, etc) I wanted. I seem to recall a couple of other totally non-relevant questions that I had to specify as well... Or, more specifically, convince the droid which was checking the order for completeness that they weren't needed. -forrest ___ --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] PRI/T1 data rate...
Andrew Joakimsen wrote: PLEASE tell me who that carrier is. I work with an inept company that doesn't even know what ANI and CPN mean. Well our ANI and CPN are one and the same. A bunch of inbred hicks somewhere in Alabama. The underlying carrier is actually really clueful (Qwest the LD carrier, not Qwest the ILEC). I was really impressed with the provisioning tech who did a really nice job of running over the parameters which are tweakable, but often you don't get to tweak, like ANI vs CID delivery (ANI please), etc.. Mainly it was strictly an issue with undertrained sales people who probably aren't paid well enough to stick around long enough to get a clue. This circuit was actually purchased through one of the website low-price brokers, which then get prices from LD resellers which actually then buy in bulk from carriers. The broker fills out the form and submits it to the LD reseller who then reviews it (and in this case complains because all of the required questions not relevant to a PRI were skipped). This is actually pretty common. Generally once you can get through the nightmare of the order, things go well. In the weird, non-relevant question category, my favorite is the whole discussion I've had every time I've ordered a PRI from my local ILEC regarding the number of presented digits. I do realize that back in old pbx days, you wanted the telco to send you say 4 digits which corresponded to your extension number, and the question is still valid with PRI - athough why anyone would want less than the full 10 digit NANPA number is beyond me. Obviously it isn't as common as I think because my ordering process normally goes something like this: Q:How many presentation digits? Me: 10 Q:Really? Me: Yes, 10 digits. Q: Are you sure your switch can handle 10 digits? Me: Yes. It routes them to the correct extension and that way I don't have to worry about number conflicts. Q: Ohhhkay, (with that tone of voice of I'm not going to protect you from your own stupidity.). (pause) Q: Now what type of start would you like on these trunks? * *ok, the last question doesn't usually get asked by the ILEC, but I just *had* to add it for humor purposes... -forrest ___ --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] Intel Desktop MotherBoards *NOT* Unsuitable for Digium Boards
George Pajari wrote: To make a long story short, according to Intel Dealer Technical Support (we became Intel dealers in order to get answers to our questions) there is no Intel motherboard that permits the IRQs to be configured uniquely. They are all hardwired and shared. This information applies to both the Intel Desktop Board and Server Board product lines. I have been using a D945GNT with great success, even with shared interrupts. But read on for a solution I just found.. I shared your frustration with not being able to get the interrupts to move to not being shared. What is more frustrating is that I knew with almost certainty that every device had a distinct interrupt line wired into the APIC, and that linux wasn't moving the interrupts off of a single interrupt Or stated differently, I knew, with reasonable certainty, that the hardware of the machine was capable of moving any device to almost any interrupt, but the software wasn't asking the hardware to do so. This functionality is available in almost every reasonably modern intel chipset. My interrupts looked something like this: # cat /proc/interrupts CPU0 0: 1507033453 XT-PIC timer 1:730 XT-PIC i8042 2: 0 XT-PIC cascade 8: 1 XT-PIC rtc 9: 0 XT-PIC acpi, ehci_hcd, uhci_hcd 10:1169132 XT-PIC libata, uhci_hcd 11: 1593809534 XT-PIC eth0, uhci_hcd, uhci_hcd, wct4xxp 12: 66 XT-PIC i8042 NMI: 0 ERR: 0 Didn't matter if I moved the card to a different slot, etc. etc. etc. etc.. Always on interrupt 11. In short, the motherboard was putting everything on interrupt 11 in XT-PIC mode. This was a *software* issue. Someone mentioned IO-APIC in this thread, and it lit up a different part of my brain for me to be able to search around the net and find that at least under CentOS, you have to be running a SMP kernel (even on a UP machine) to be able to get the IO-APIC functionality. Now, with a SMP kernel I get: # cat /proc/interrupts CPU0 0: 55935IO-APIC-edge timer 1: 8IO-APIC-edge i8042 8: 1IO-APIC-edge rtc 9: 0 IO-APIC-level acpi 12: 66IO-APIC-edge i8042 169: 0 IO-APIC-level uhci_hcd 185: 0 IO-APIC-level uhci_hcd 193: 0 IO-APIC-level uhci_hcd 201: 0 IO-APIC-level ehci_hcd, uhci_hcd 209: 17700 IO-APIC-level wct4xxp 217:201 IO-APIC-level eth0 233: 5187 PCI-MSI libata NMI: 0 LOC: 55761 ERR: 0 MIS: 0 Much Better. For reference: Linux 2.6.9-22.0.1.ELsmp #1 SMP Thu Oct 27 13:14:25 CDT 2005 i686 i686 i386 GNU/Linux -forrest ___ --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] Pricing for DS3000P
On Sat, 4 Jun 2005, Tom Fanning wrote: What's so special about Digium cards that makes them this expensive? $4000 for a PCB is extortion IMO! $4K for a channelized DS3 card isn't all that bad. We've been paying ~2K for a free-framed DS3 card. Component-wise yov'e got upwards of $1K if not $2K on-board. Factor in RD time and some reasonable profit, $3K or even $4K isn't that bad. Now if you want to discuss whether or not the prices for the IC's and other components are extortion or not, then I might be willing to agree with you. -forrest ___ 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 UA behind NAT
On Fri, 3 Jun 2005, Eric Yu-Wei Sung wrote: I am trying to make 1 soft SIP UA behind NAT connect to a public hard CISCO UA via a public asterisk server. The CISCO UA can hear the voice from the SIP UA but not vice versa. I do set nat to yes for the soft phone. Any help would be greatly appreciated. Turn off reinvite or whatever it's called in sip (can never remember if reinvite is the sip or iax2 term). This will force the audio through asterisk. -forrest ___ 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] Portable USB headset for VoIP
I'm trying to find a voip-suitable USB headset (I.E. headphones + microphone) which I can use with my laptop while I'm traveling and using Firefly or another softphone. I'm currently using a Logitech headset which works well (except the echo it generates toward the other caller when I turn up the gains too high), but it just doesn't carry well - in fact, I can't carry it in my laptop case any more just becuase it doesn't fit and it was getting very beat up. I'd like to find something which folds up and is designed for travel. It has to be USB sicne I don't have a MIC in (just line) on my laptop. Any ideas? -forrest ___ 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] Known Working Motherboard/CPU for TE410P
I've been running into some issues with a TE410P and lost audio and similar which I have tracked down to some apparent incompatibilities between the version of Linux I'm running and my hardware - which causes zttest to not get 100% of the samples every time. Instead of fighting this on the machine it's running on (which I don't want to run it on long-term anyways), I'm wanting to put together a new server for this. For a lot of reasons I don't want to go into here, we tend to prefer to build or own servers as opposed to buying a preconfigured server. With that in mind, I'm trying to select an appropriate motherboard. I want to ensure that the CPU/MB combo I pick is known to work reliably with Asterisk and a TE410P (3.3V), and passes zttest with flying colors. What motherboards are known to work well? -forrest ___ 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] Sipura 401 Unauthorized.
I'm having ongoing registration fits with some SPA-2000's. Right now I have one which, based on the debugging output repeatedly fails with 401 unauthorized: - -- SIP read from 206.127.114.240:5060: REGISTER sip:voip-proxy.mt.net SIP/2.0 Via: SIP/2.0/UDP 206.127.114.240:50291;branch=z9hG4bK-f4e3f6bc From: MIC Sipura User sip:[EMAIL PROTECTED];tag=abf0b0fbea0a3e68o0 To: MIC Sipura User sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 52 REGISTER Max-Forwards: 70 Contact: MIC Sipura User sip:[EMAIL PROTECTED]:50291;expires=60 User-Agent: Sipura/Sipura/SPA2000-2.0.13(g) Content-Length: 0 Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER Supported: x-sipura --- (12 headers 0 lines)--- Using latest request as basis request Sending to 206.127.114.240 : 50291 (NAT) Transmitting (NAT) to 206.127.114.240:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 206.127.114.240:50291;branch=z9hG4bK-f4e3f6bc;received=206.127.114.240;rport=5060 From: MIC Sipura User sip:[EMAIL PROTECTED];tag=abf0b0fbea0a3e68o0 To: MIC Sipura User sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 52 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] Content-Length: 0 --- Transmitting (NAT) to 206.127.114.240:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 206.127.114.240:50291;branch=z9hG4bK-f4e3f6bc;received=206.127.114.240;rport=5060 From: MIC Sipura User sip:[EMAIL PROTECTED];tag=abf0b0fbea0a3e68o0 To: MIC Sipura User sip:[EMAIL PROTECTED];tag=as3bb6fc64 Call-ID: [EMAIL PROTECTED] CSeq: 52 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: sip:[EMAIL PROTECTED] WWW-Authenticate: Digest realm=asterisk, nonce=36609d12 Content-Length: 0 - This box will register and then be ok for several days and then will go into this mode. I've seen other sipuras doing something similar but with 407. In fact, calls were made with this box yesterday with absolutely no configuration changes on either end, until I started to try to figure out why it isn't registering. With only 5-6 SPA-2000's in test and several of them acting flaky registration-wise I'm feeling that I'm missing something which causes this flakiness. Both the SPA and asterisk have been rebooted. Asterisk has actually been updated to the latest CVS version today in case there was an already-in-cvs fix. The config I have in asterisk for this sipura box is: [A0974L1] type=friend host=dynamic context=cosinternational secret=REMOVED callerid=MIC Sipura User 406### dtmfmode=rfc2833 reinvite=no canreinvite=no nat=yes qualify=no Ideas? Other places I should look? -forrest ___ 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] Fax and Voice
On Thu, 24 Mar 2005, Guy Decarpentrie wrote: Le jeudi 24 Mars 2005 10:56, Altus Snyman a écrit : google asterisk fax Well, i know how to receive and mail a fax, now i want to know how to detect if the call is a fax or a voice call, and reroute the call if it's a voicecall, and mail the fax if it's one. I think you need to follow the original directions: go to google, search for asterisk fax The very first hit tells you exactly what you want: Fax Detection with IAX and SIP If you are trying to detect faxes over IAX, SIP, or for that matter any type of channels, Newman Telecom has created NVFaxDetect and updated BackgroundDetect? as NVBackgroundDetect for that purpose. We have had near perfect results on decent IAX connections using ULAW/ALAW. Fax detection utilizes Asterisk DSP and works in the same way once detected, faxes are sent to the fax extension. See Asterisk fax for example fax detection scripts. and has links to another part of the wiki where examples are given. -forrest ___ 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] OT: Best DB
On Wed, 9 Mar 2005, [EMAIL PROTECTED] wrote: For some reason I didn't think PostgreSQL was for mission critical apps. I don't think I have any reasoning behind it, just didn't think it was hardcore...sounds like i might be wrong...i'll have to look into it more. For your app, probably either MySQL or PostreSQL will work. I'm a happy MySQL user ... others are just as happy with PostgreSQL. I think it's almost what you're familiar with at this point. The differences between the two are getting smaller. MySQL traditionally was considered a very high speed database server lacking some advanced features such as transactions and triggers and some query types. Postgres was considered a slower, feature complete SQL implementation. Today, MySQL has more features that it lacked earlier - i.e. it's got transactions and additional queries, and so on. I understand that PostgreSql has also gotten faster than it used to be. So, at this point it's almost devolved into a holy war as opposed to there being any real difference. Personally I use MySQL because I find it easier to admin and configure on my FreeBSD systems than PostgreSQL, which I tend to have ongoing problems with in the spots I have to run it. I don't miss the couple of PostgreSQL features that mysql still doesn't have (but will in the near future). I'd really recommend that you look at developing the app so it is database independent - at least between MySQL and PostgreSQL. That way, you can swap from one to the other if you decide you don't like the one you pick initially. -forrest ___ 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] Re: No ringback over IAX - LiveVoip
Seems kind of starnge that they are the only ones having this problem. I am pulling an account from Voicepulse using IAX and not have a problem at all. Maybe they need to call Digium, or some other contractor, and pay someone to set it up for them correctly since it is obviously they cannot accomplish this. I had some issues with VoicePulse as well with IAX. Don't remember exactly what they were... but I believe it may had been an IAX trunking issue. -forrest ___ 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-bsd] Asterisk not accepting multiple SIP phone logins
On Fri, 11 Feb 2005, Brian Buhrow wrote: Hello. You can't have two phones login with the same extension. You need to assign one phone to 101, and the other to 102. Set the user to 101 on one and 102 on the other. Actually, that isn't quite 100% accurate. The more accurate statement is that you can't have two phones log in as the same username/etc in sip.conf. You can, however have extensons.conf ring numerous phones all at the same time for a given extension. What you can do is set up two separate phone configurations in sip.conf, one per phone. I.E: [101-phone1] ...sip config... [101-phone2] ...sip config... and then modify your dial command in extensions.conf to look something like: exten = 101,1,Dial(SIP/101-phone1SIP/101-phone2,20,tr) ___ 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
