Re: [Asterisk-Users] PHP/AGI Problem
On Thu, 2005-26-05 at 17:59 -0400, Michael Stearne wrote: On 5/26/05, Benjamin West [EMAIL PROTECTED] wrote: Michael, The version, in the context of Jon's problem, was irrelevant. Jon's problem was due to a small bug in his code, and not related to PHPAGI. Yeah. I was just wondering what version people were using since 2.0 hasn't been formally released yet. I spoke to the developer and he suggested 2.0. phpagi-2.14 has now been released, this is the first official release for the 2.x series but should be stable. There is also a compatibility module in CVS to support older 1.x apps. You can snag it from http://phpagi.sourceforge.net/ -- Matthew Asham - the B.C. Wireless Network Society www.bcwireless.net - +1 604 484 5289 x1006 ___ 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] PHP/AGI Problem
On Thu, 2005-26-05 at 21:23 -0400, Michael Stearne wrote: Thanks for the efforts Matt. David Eder is the one who deserves the praise, he's done wonderful work on version 2. Matthew ___ 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] NuFone problems to non-na numbers
Is anyone else having problems with Nufone dialing international (non NA) numbers? Pretty much every intl number dialed comes up with a voice intercept saying the call could not be completed as dialed. Tried it with two separate accounts, and the numbers themselves work from the local telco. The problem appears to have started within the last few days (and yes I have emailed [EMAIL PROTECTED], just wondering if we're the only ones having the problem). Matthew -- Matthew Asham - the B.C. Wireless Network Society www.bcwireless.net - +1 604 484 5289 x1006 ___ 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] Text Messaging or AIM
On Sun, 2005-03-13 at 11:14, Peter Svensson wrote: Whatever gave you that idea? Most operators have an interface allowing reception of sms:es over internet. The protocols may be strange (they are) and the pricing models vary greatly, but there are many receive interface to sms:es. I've been wondering about this for some time, is there a common product name for this service? ___ 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 Hardware in Canada
On Tue, 2004-12-14 at 11:09, Adi Linden wrote: I am looking for a supplier of Digium hardware in Canada. Any suggetions? Other coast, but http://www.netvoice.ca/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
Re: [Asterisk-Users] SysMaster and GPL Violation (lets think before we
What's a top post? oh and, me too! On Sun, 2004-11-14 at 18:15, [EMAIL PROTECTED] wrote: Does that mean I have to put my posts there when I'm PMSing? On Sun, 14 Nov 2004 17:30:23 -0800, Bruce Ferrell wrote: Only if we can move the top-post discussion there too Gary wrote: Hi folks, Might I propose a new mailing list ?? Asterisk-bitch Thus discussions such as the one with this topic could be moved to it rather than clutter up an already very busy list. All those in favour ? Only if we can move the top-post discussion there too Sure, any of those types of debate. In fact, i just wished people would NOT use the list for debates !! . ___ 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] Enhancing list quality?
people complaining about list etiquette... On Tue, 2004-11-02 at 04:45, Patrick wrote: Hi all, Following up on Steve Szmidt's post, have we come to a point where something should be done about the deteriorating signal to noise ratio on the asterisk-users list? Similar to other fast growing successful FOSS projects/communities, the Asterisk community is no exception when it comes to growing pains. Anyone who has been subscribed for quite some time surely has noticed: * the lazy top posting * asking too simple/obvious questions that are found on voip-info.org, the list archives, google and the docs that comes with the source * asking totally unrelated questions (the Sparco example) * asking why it doesn't work without giving any config info * the GPL license, Windoze vs. Linux, You are clueless! No you are! discussions/flame wars that go on and on and on * the lack of proper trimming of posts * the html email sent time after time in spite of requests to turn it off * the huge signatures that imho are a form of spam or just a waste of bandwidth Some suggestions to hopefully improve things: 1) change the Welcome to the asterisk-users list email to contain: - before sending email to more than 8000 people, search voip-info.org, the mailing list archives, google and the docs in the source - do not top post - add as much config info as possible - trim long posts (especially unnecessary sigs) - posts should be related to Asterisk - be nice, polite, courteous. keep flames off list - your sig should not be a billboard for your products/services (spam) if you want to advertise your services, use asterisk-biz - do not send html mail to the list. It will get rejected or bounced or whatever (see item 3) - do not use unnecessary capitals, exclamation marks and question marks (don't be loud) - do not mail questions to asterisk-dev that are not related to the actual development of asterisk - do not crosspost your question to asterisk-users and asterisk-dev - do not resend your email after 4 hours because you have not received an answer. Join #asterisk on Freenode irc if you require a possibly speedier answer 2) post this info periodically to the list to remind people 3) strip all html or bounce it back to the sender telling them to send normal email, not html email. Add the list's netiquette too, just to make sure. Afaik mailman is capable of stripping html from email 4) ask the community to respond to people ignoring these reasonable requests and gently point them in the right direction Feedback welcome. Regards, Patrick ___ 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] common numbers ?
http://www.google.ca/search?hl=enq=vertical+service+codesbtnG=Google+Searchmeta= On Fri, 2004-10-22 at 01:57, [EMAIL PROTECTED] wrote: hi, Can someone point me to a list of a common numbers used for different functions ex. callparking,forwarding etc... I can thin of my own but want to know is there some standard wich is good to follow. tia ___ 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] viewing fax tiffs?
Hello, I have spandsp setup to accept incoming faxes and receiving tif files via Email. Using tiff2pdf, or tiff2ps -a2 or even tiffsplit, the last page of the fax is cut off and the quality of the text looks squished. I figure it's a tiff parsing thing, as opposed to a problem with my spandsp installation (heh). Has anyone experienced the same thing, or can anyone recommend a proper way to convert fax'd tiffs to another format or even a multi-page tiff viewer for Linux? Thanks 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
Re: [Asterisk-Users] e164.org
On Sat, 2004-05-22 at 18:08, Tony Hoyle wrote: Simon Dorfman wrote: I wonder if someone can help me understand this. Let's say I configure my asterisk box to use e164 and then I try to call a phone number in Germany. I'm in the U.S.A. So if the number I'm calling in Germany is registered in e164's dns, would my call be routed directly via their voip provider? Or directly to their asterisk box? And would it be free? From the looks of it, they're just a directory... it looks like their not running asterisk themselves. It's a DNS root, that Asterisk (via the EnumLookup application) can use. The EnumLookup() application will resolve the number to a dial() channel. ie: ; north america enum exten = _1NX,1,Playback(doing-enum-lookup) exten = _1NX,2,EnumLookup(${EXTEN}) exten = _1NX,3,BackGround(enum-lookup-successful) exten = _1NX,4,Dial(${ENUM},30,tr) exten = _1NX,5,Hangup exten = _1NX,6,Playback(enum-lookup-failed) exten = _1NX,7,Hangup To get * to resolve against e164.org, add: search = e164.org to /etc/asterisk/enum.conf. So yes Simon, if you called someone in Germany and it was the zone, your call would be switched over the 'net. If not, you could drop it to NuFone or some other carrier. They use something called EnumLookup which I guess is some kind of plugin/script. If the number you're calling is in their database, it calls the VOIP number directly, otherwise it calls the POTS number Or whatever else your dial plan wants to do. 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
Re: [Asterisk-Users] e164.org
You know, sleep deprivation cause people to do dumb things. The example I pasted was hastily pasted and renumbered, exten = _1NX,6,Playback(enum-lookup-failed) exten = _1NX,7,Hangup are actually: exten = _1NX,103,Playback(enum-lookup-failed) exten = _1NX,104,Hangup Duane wrote up some more detailed examples at http://www.e164.org/config.php. Sorry for not proofing that when I posted it. I'll go sleep now. On Sat, 2004-05-22 at 18:46, Tony Hoyle wrote: Matthew Asham wrote: ; north america enum exten = _1NX,1,Playback(doing-enum-lookup) exten = _1NX,2,EnumLookup(${EXTEN}) exten = _1NX,3,BackGround(enum-lookup-successful) exten = _1NX,4,Dial(${ENUM},30,tr) exten = _1NX,5,Hangup exten = _1NX,6,Playback(enum-lookup-failed) exten = _1NX,7,Hangup Interesting.. how does it know to go to '6', or does it just jump +4 on failure? That reminds me I seriously need to restructure my extensions.conf... there's no way currently I could add anything like that without major surgery (only discovered the 'local' target this afternoon so I have everything copied/pasted). Tony ___ 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] IAX/SIP in 604?
Hello, I hate to ask here, but.. Does anyone know of an IAX/SIP DID provider in Vancouver, British Columbia? I'm looking for a voicepulse isk service, one DID with standard calling features and some sort of long distance package. I've looked around on voip-info.org's list of VoIP providers but so far I haven't found one that offers a 604/778 number. Thanks 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] How does Asterisk use CPU?
Hello, I'm trying to figure out what portions of Asterisk need a lot of CPU time. I thought I read somewhere that a Dual P4 2.something will support approximately 80 calls. Is this based on calls that Asterisk is actively doing voice processing for (say, Zap channels and voicemail)? Would a SIP client going through Asterisk and out an IAX channel be CPU intensive if I kept the codec the same throughout the path? I'm probably not asking very clearly, it's awefully late (err, early) but any pointers would be greatly appreciated. Thanks Matthew ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [Asterisk-Users] How does Asterisk use CPU?
Thanks Steve, that pretty much answers what I wanted to know. I was asking more out of general than for a specific deployment, but if I do have further questions I'll be sure to elaborate. :) Matthew Friday, November 28, 2003, 6:41:23 AM, you wrote: On Fri, 2003-11-28 at 06:51, Matthew Asham wrote: Hello, I'm trying to figure out what portions of Asterisk need a lot of CPU time. Mostly codec translations. I thought I read somewhere that a Dual P4 2.something will support approximately 80 calls. Is this based on calls that Asterisk is actively doing voice processing for (say, Zap channels and voicemail)? I think that thread would have been more specific to a task, or just blanket overkill. Would a SIP client going through Asterisk and out an IAX channel be CPU intensive if I kept the codec the same throughout the path? If the codec is the same, then all that is being done is reformating the control protocol. If there is any codec translations, then you would run into some extra overhead. I'm probably not asking very clearly, it's awefully late (err, early) but any pointers would be greatly appreciated. Maybe you would do better to ask more pointed what YOU need help with. If you need help with specing out a machine for your installation, then start specing your installation so we can help. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users