[Asterisk-Users] origination providers
hi folks, Has anyone found a good (and, ideally, cheap -- we don't really want any per-minute charges) origination provider which can handle a moderate number of simultaneous incoming calls (to the same, single DID)? Many of the providers I've tried contacting either won't call me back, or want me to sign an NDA just to get a rate quote, or some other bullshit. Most of the providers whose rates are plainly posted on their website have a limit of at most 4 or 6 simultaneous calls, which is not likely to be enough for the application I'm considering. You can reply off-list or on-list, as you prefer. many thanks, mike -- mike castleman network / systems administrator democracy now! mailto:[EMAIL PROTECTED] tel:+1-212-431-9090 (office) tel:+1-646-382-7220 (mobile) signature.asc Description: Digital signature ___ 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] origination providers
I'm not entirely sure what you're asking. The application in question will involve setting up asterisk in a datacenter where we already have a fair amount of bandwidth. As far as the DID provider's portion of the bandwidth, I assume that they would account for this in the rate they quote us. I'm just asking the list if they have any good experiences with origination providers, as my attempts to get them just to return my calls have not been successful. If it's relevant, I imagine gsm or speex codec for this application, but haven't yet made a decision. mike On Tue, May 24, 2005 at 03:42:43PM -0400, Kanuri, Seshu (Company IT) wrote: Assuming that you will need about 12 to 24 simulataneous calls on each DID you want to run, and you are using Ulaw to get these calls, what is the bandwidth that the DID provider has to give you, apart from the DID service? Ulaw needing 64 kbps per line, needs 1.2 mbps for 20 simultaneous calls. Assuming a Data T1 costs about $500 bucks a month and assuming that you need/use the DID for only 8 hours a day at that rate, it costs about $100 per month in data bandwidth alone. Who will pay for this, If it is not Democracynow who is footing the bill? Seshu -- mike castleman network / systems administrator democracy now! mailto:[EMAIL PROTECTED] tel:+1-212-431-9090 (office) tel:+1-646-382-7220 (mobile) ___ 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] Will Asterisk do well in this application?
On Mon, May 09, 2005 at 04:16:18PM -0700, snacktime wrote: We have, and the one comment I would make is that asking for an alphanumeric username and password is too much for most people. A good combination we found is a numeric pin/password and the phone number. It's easier for people to remember, easier to enter, and then you can also do caller id verification. Caller ID is getting easier to spoof every day, so this may be unwise, depending on the security level required. mike -- mike castleman network / systems administrator democracy now! mailto:[EMAIL PROTECTED] tel:+1-212-431-9090 (office) tel:+1-646-382-7220 (cell) signature.asc Description: Digital signature ___ 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] vmail.cgi: -rwsr-sr-x as root *still* won't read the files
On Fri, Apr 29, 2005 at 12:23:48AM -0500, Brian Capouch wrote: Drat. Perl screams bloody murder if you try to just set its SUID bit, which of course is dangerous as hell. The perl-suid is *not* simply a version of perl with the suid bit set but rather a helper binary which allows perl to run suid scripts. Try it. mike -- mike castleman network / systems administrator democracy now! mailto:[EMAIL PROTECTED] tel:+1-212-431-9090 (democracy now) tel:+1-646-382-7220 (cell) ___ 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] vmail.cgi: -rwsr-sr-x as root *still* won't read the files
Try making sure you have installed the suid perl stuff if your OS needs it. Some kernels do not natively obey the setuid flag when executing scripts (On Debian, this involves installing the perl-suid package. Other Linux-based distributions probably need something similar.) On Thu, Apr 28, 2005 at 11:43:57PM -0500, Brian Capouch wrote: I'm running Apache as nobody. Wondering why the SUID vmail.cgi script still can't read my files; it comes with the bits set SUID, which I thought would do the trick. It works just fine if I make the files in the maildir world-readable. Thanks. No clues in the archives no Wiki that appear germane. B. -- mike castleman network / systems administrator democracy now! mailto:[EMAIL PROTECTED] tel:+1-212-431-9090 (office) signature.asc Description: Digital signature ___ 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] SetGroup on dialed calls?
hi folks, I looked through the list archives and the wiki, but couldn't find an answer to this. Apologies if I just missed something obvious. I want to only have call waiting for certain calls (i.e., those that are dialed directly to a user rather than going through a queue). It seems that the way to do this is to call SetGroup() on all incoming calls and CheckGroup() only on non-call-waiting calls, combined with Local/ channels when needed. However, I can't figure out how to do the SetGroup() properly on outgoing calls (i.e., those that the internal user dials). Is there some obvious way that I'm missing to call some commands before proceeding to the rest of the dialplan? Any thoughts -- including alternate ways to achieve the same goals -- would be much appreciated. (I tried using the incominglimit parameter in sip.conf, but it seems not to be very flexible.) Many thanks, mike -- mike castleman network / systems administrator democracy now! mailto:[EMAIL PROTECTED] tel:+1-(212)-431-9090 signature.asc Description: Digital signature ___ 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
