Re: [asterisk-users] Linux Command Line Soft Phone - $200+ bonus
--- Tzafrir Cohen [EMAIL PROTECTED] wrote: your requirement don't really make sense. to try again: speex and/or gsm - put into /etc/init.d/___ - phone enabled on boot up Huh? IS that phone a client program? If so: why should it be run as a server? There are plenty of ways to run a program at desktop startup. you are right - I'm after ease of maintenance. - automatically navigate around gnome and kde sound Huh? I have noticed in some soft phone docs that the gnome and kde sound systems need to be turned off for the soft phone to work. the phone needs to work with neither gnome nor kde running. - automatically navigate dhcp (if any) Huh? that the softphone will find its way to the server whether or not the computer is behind a router (dhcp). - gnu has something sort of close(?) - must install through one command, thru apt-get, or thru synaptic On which distribution? initial install on Ubuntu 6.10 Debian already has a host of free phones. The best seem to be Twinkle and Ekiga for SIP and kiax for IAX. A number of others are usable. The only one that does *both* SIP and IAX (if you really need that) is yate-gtk :-p . I'm after is a phone that is working when the computer boots. NO user interface - all control is done through the asterisk server to which it's connected. also, as above, not dependent on gnome or kde - with no graphics this should not be a problem. - must be hosted in free public place - must run on Ubuntu first try Which version? Ubuntu has some of the Debian packages. 6.10 phase 2: - have same run under Puppy Linux Consider giving more information on the limitations of the system (memory? disk-space?) I'm concerned (and ignorant) about installation Do you Yahoo!? Everyone is raving about the all-new Yahoo! Mail beta. http://new.mail.yahoo.com ___ --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] Linux Command Line Soft Phone - $200+ bonus
phase 1 requirements: - sip and/or iax2 using g729 and/or gsm - put into /etc/init.d/___ - phone enabled on boot up - all parameters in /etc/___ - automatically navigate around gnome and kde sound - automatically navigate dhcp (if any) - gnu has something sort of close(?) - must install through one command, thru apt-get, or thru synaptic - must be hosted in free public place - must run on Ubuntu first try phase 2: - have same run under Puppy Linux please leave bids on project here or email me directly. - Don't get soaked. Take a quick peak at the forecast with theYahoo! Search weather shortcut.___ --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] AGI question
in your dialplan: [context] ... h,1,AGI(...) David Ruggles [EMAIL PROTECTED] wrote: I'm working on writing some test IVR code in AGI. I can't get my FXO port to detect a hang-up, but I'm going to deploying this using Digital cards so I decided to just skip that problem for now. However this leaves me with a question. How does AGI detect a hang-up if everything is operating normally? TIA!! Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network Engineer Safe Data, Inc. (910) 285-7200 [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 - Have a burning question? Go to Yahoo! Answers and get answers from real people who know.___ --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] Softphone on Linux
please send me more info thanks! Tim Panton [EMAIL PROTECTED] wrote: On 5 Feb 2007, at 21:46, chester c young wrote: Need to deploy between 50 to 300 lightweight Linux - only browser and softphone. You might want to consider our lightweight java softphone (Corraleta SDK) - it can be embedded in a web page - zero install/config in the client. The UI is in HTML and javascript, so you can get it _exactly_ the way you want it. Any recomendations? Clearly I'm biased :-) Tim Panton www.mexuar.com www.westhawk.co.uk/ ___ --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 - Everyone is raving about the all-new Yahoo! Mail beta.___ --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] Softphone on Linux
Need to deploy between 50 to 300 lightweight Linux - only browser and softphone. Any recomendations? - Expecting? Get great news right away with email Auto-Check. Try the Yahoo! Mail Beta.___ --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] problems with SJPhone (I feel stupid about this)
have a Grandstream and SJPhone SIP phones going to asterisk. with SJPhone (on Linux) getting. any ideas?? SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.2.100;branch=z9hG4bKc0a80264001045c3c2c52331d4920678;received=24.10.123.39;rport=60754 From: sip:[EMAIL PROTECTED];tag=22261807771886928353 To: sip:ca.dummy.net;tag=as45966c6b Call-ID: [EMAIL PROTECTED] CSeq: 41 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Accept: application/sdp Content-Length: 0 - Check out the all-new Yahoo! Mail beta - Fire up a more powerful email and get things done faster.___ --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] Dynamically Adding A Context
In order to do this, I have to add a couple quick extensions to the dial plan dynamically, so I have to be able to add my own context. from API use Command to run the CLI command add extension - The fish are biting. Get more visitors on your site using Yahoo! Search Marketing.___ --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] Response on dialin - no extension
On a SIP phone is it possible to enter the dialplan when the user picks up the phone without having to wait for the user to press an extension? Is is possible to do something like [sip-test] s,1,Answer s,2,Playback(welcome) s,3,WaitExten(30) 1,1,Noop(exten 1) ... t,1,Goto[s,2] - Be a PS3 game guru. Get your game face on with the latest PS3 news and previews at Yahoo! Games.___ --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] Stumped with Dial - $50 for answer - close
its notransfer=yes in iax.conf not transfer=no :) this is getting close! however, it takes about SEVEN seconds after the called party hangs up before the next priority is executed - same as with the T option. as contrast to h option, when called party hits asterisk, the next priority is almost immediate. the seven second delay makes the application very difficult to use. Expecting? Get great news right away with email Auto-Check. Try the Yahoo! Mail Beta. http://advision.webevents.yahoo.com/mailbeta/newmail_tools.html ___ --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] Stumped with Dial - $50 for answer - closed!
the answer sucks, but is apparently correct. imho Andrew Kohlsmith is The Man, although there was someone in Germany who emailed about the T option which actually works about as well - please email me. Andrew Kohlsmith please email me. Will pay paypal if that's ok. --- Andrew Kohlsmith [EMAIL PROTECTED] wrote: On Tuesday 16 January 2007 2:31 pm, chester c young wrote: however, it takes about SEVEN seconds after the called party hangs up before the next priority is executed - same as with the T option. What kind of last leg are these calls? to POTS (even CAS T1) or PRI? as contrast to h option, when called party hits asterisk, the next priority is almost immediate. This is because Asterisk knows you want a hangup. My hunch is that you're terminating to POTS instead of PRI, and that is how long it takes for your telco provider to supply CPD signaling on the analog interface. I know Bell Canada is about that long. -A. ___ --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 Need Mail bonding? Go to the Yahoo! Mail QA for great tips from Yahoo! Answers users. http://answers.yahoo.com/dir/?link=listsid=396546091 ___ --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] Stumped with Dial - $50 for answer - closed!
$25 to openvpn.org - thanks to Anselm Martin Hoffmeister --- Anselm Martin Hoffmeister [EMAIL PROTECTED] wrote: Am Dienstag, den 16.01.2007, 15:04 -0800 schrieb chester c young: the answer sucks, but is apparently correct. If your application involves the caller (e.g. an employee of your company) to rate the call he just did, or to enter any data to a mysql database over the phone right after the call, you could use the H option (neither T nor h, then) and tell your phone personell about it: After the call finished, press * and answer the questions the computer reads out to you. That way, Asterisk would (expectedly) stay in the Audio path and even find out that the call ended if your employee did not *g* - and your employees could cut those 7 second delays. Your IVR for aprés-call interaction should skip the first digit if it happens to be an * though, because it could happen that Asterisk sees the far end hangup just a blink before the user hits the * key. This is for volunteers calling other members of their organization, so need to keep everything low key and polite. A volunteer will call in, either by POT or SIP and will stay connected as Asterisk dials the number of the fellow member whom they've selected on a browser. The seven seconds is bad because that's a bit too long between calls - people tend to loose their concentration. Be a PS3 game guru. Get your game face on with the latest PS3 news and previews at Yahoo! Games. http://videogames.yahoo.com/platform?platform=120121 ___ --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] Stumped with Dial - $50 for answer
g option to Dial only continues the dialplan if the destination (called) leg of the call hangs up. It will NOT cause the dialplan to continue if the source (calling) leg of the call hangs up. When the calling channel hangs up, Asterisk will send the remaining leg of the call to exten = h. this is exactly right and is exactly the problem. when the called leg hangs up the dial plan does not proceed to the next priority. Bored stiff? Loosen up... Download and play hundreds of games for free on Yahoo! Games. http://games.yahoo.com/games/front ___ --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] Stumped with Dial - $50 for answer
Silly question: how are the calls going out? If they're going out through an analog line without the ability to detect hang-ups, then, that's the problem. calls are coming in and out thru an Asterisk server using iax2. have tried two different DID providers and have same problem. Bored stiff? Loosen up... Download and play hundreds of games for free on Yahoo! Games. http://games.yahoo.com/games/front ___ --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] Stumped with Dial - $50 for answer
--- Paul [EMAIL PROTECTED] wrote: Anselm Martin Hoffmeister wrote: Curious - is this still a $50 thread? yes. Never miss an email again! Yahoo! Toolbar alerts you the instant new Mail arrives. http://tools.search.yahoo.com/toolbar/features/mail/ ___ --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] Stumped with Dial - $50 for answer
cannot make Dial(...,,g) work correctly, although Dial(...,,gh) works just fine. (to make matters worse, it does seem to work sometimes, although once working or not working between changes it either works or doesn't work all the time.) extensions.conf: [general] static=yes writeprotect=no autofallthrough=yes clearglobalvars=no priorityjumping=no [incoming] exten = 505111.,1,Answer exten = 505111.,2,Noop(top) exten = 505111.,3,Dial(IAX2/[EMAIL PROTECTED]/1501212,,g) exten = 505111.,4,Noop(done the dial) exten = 505111.,5,Goto(2) after the called party hangs up nothing happens, the 4,Noop(done the dial) is never executed. if the dial is done with a Dial(...,,gh), and the answering phone hits an asterisk, the next priority is executed as expected. -- Executing [EMAIL PROTECTED]:1] Answer(IAX2/telavoip-2, ) in new stack -- Executing [EMAIL PROTECTED]:2] NoOp(IAX2/telavoip-2, top) in new stack -- Executing [EMAIL PROTECTED]:3] Dial(IAX2/telavoip-2, IAX2/telavoip/1505222||g) in new stack -- Called telavoip/1505222 -- Call accepted by 1.2.3.4 (format ulaw) -- Format for call is ulaw -- IAX2/telavoip-4 is proceeding passing it to IAX2/telavoip-2 -- IAX2/telavoip-4 is ringing -- IAX2/telavoip-4 stopped sounds -- IAX2/telavoip-4 answered IAX2/telavoip-2 -- Channel 'IAX2/telavoip-4' ready to transfer -- Channel 'IAX2/telavoip-2' ready to transfer -- Releasing IAX2/telavoip-2 and IAX2/telavoip-4 -- Hungup 'IAX2/telavoip-4' /* stays here until originating phone is hung up */ == Spawn extension (telavoip-iax-in, 1505111, 3) exited non-zero on 'IAX2/telavoip-2' -- Executing [EMAIL PROTECTED]:1] NoOp(IAX2/telavoip-2, call_loop: hungup) in new stack -- Hungup 'IAX2/telavoip-2' Sucker-punch spam with award-winning protection. Try the free Yahoo! Mail Beta. http://advision.webevents.yahoo.com/mailbeta/features_spam.html ___ --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] Stumped with Dial
--- Anselm Martin Hoffmeister [EMAIL PROTECTED] wrote:
Am Sonntag, den 14.01.2007, 17:13 -0800 schrieb chester c young:
cannot make Dial(...,,g) work correctly, although Dial(...,,gh)
works
just fine. (to make matters worse, it does seem to work sometimes,
although once working or not working between changes it either
works or
doesn't work all the time.)
For me,
exten = 990,1,Dial(SIP/sip502,60,gro)
exten = 990,2,Playback(special/monkeys)
exten = 990,3,Hangup
does what I would expect, and it seems reliable:
Sip phone 502 rings, after taking the call there and hanging up, the
caller gets the monkeys file.
I use the ro here because it seems that makes Asterisk behave as I
want it; ro should not be relevant for the g.
It does not to work though if I do not specify a timeout (60 in my
case)...
You can use the DIALSTATUS variable afterwards to find out wether a
call
had taken place, something like
exten = 990,3,GotoIf($[X${DIALSTATUS} = XANSWER]?10)
exten = 990,10,NoOp(Call had been answered)
and do some dialplan magic :)
BR
Anselm
thanks, but tried gro and even Playback instead of Noop after the Dial
- still no luck. it's placing the call to a pstn thru an aix.
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Re: [asterisk-users] Symbolic Link
--- bilal ghayyad [EMAIL PROTECTED] wrote: Hi List; To create the symbolic link, I read in the documenation that I have to type this command: # ln -s /usr/src/'uname -r' /usr/src/linux-2.4 1) What it means by 'uname -r'? 2) Why I have to create such symbolic link to do pointing for the kernel? For what exctly will be used with asterisk? 3) What is the relation between creating such symbolic link and build directory? Any advise. Regards Bilal 1) those are backticks: `uname -r` (it prints the kernel release) 2) ? 3) don't understand your question what are you trying to do?? Do you Yahoo!? Everyone is raving about the all-new Yahoo! Mail beta. http://new.mail.yahoo.com ___ --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] API: how to bridge originated call?
Moises what you've done here looks great, but some examples or a little doc would be really helpful - following the bug report, although informative, is a very difficult way to extract specs. in my case I want a user to be on-line all the time - the system will dial and connect them and, when they're done, they select the next one. what I'm doing now is putting them into a loop with a g-option on the dial. the number it dials is set thru the api. if the number's not set it waits one second and loops again. from my limited knowledge using a Bridge() function is much more elegant, but I am in the dark as to what the context of the user, what happens on hangup (can it fall thru?), etc. maybe after this demo is done I'll solve this correctly using Bridge, but alas little time now for experimentations. thanks cy --- Moises Silva [EMAIL PROTECTED] wrote: I have uploaded a working patch for version 1.2.12.1, and other that seems to work in Trunk, but few people is reporting results, you can help to get this into Asterisk, go here: http://bugs.digium.com/view.php?id=5841 The patch I ported to 1.2.12.1 is working fine, I have tested in my servers, is the one called bridge-1.2.12.1.patch, there are other ones that say trunk, obviously only work with the trunk version of Asterisk. Kind Regards On 1/3/07, chester c young [EMAIL PROTECTED] wrote: (my pstn calls are coming in thru an upstream asterisk server, so the called and calling phone number is passed as an extension.) when caller comes in on 555, he will go to extension 1234 where he will wait for the API to make a call to 999 for him. how do I bridge the two calls? extensions.conf: ;context where caller comes in [caller] 555,s,1 Answer() 555,s,n UserEvent(Init) ;this lets me know the connection for 555 555,1234,1 Noop(caller waits to be bridged) 555,1234,2 Background(soothingmusic) ;context for connection - is this needed? [connect] from the API: (do I need to create a new context/extension first?) Action: Originate Channel: IAX2/upstream/999 -- calls 999222 thru upsteam IAX Context: ?? Exten: ?? Priority: ?? __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --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 -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --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 Yahoo! Music Unlimited Access over 1 million songs. http://music.yahoo.com/unlimited ___ --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] getting tones during conversation
after the Dial has connected, I want the caller (on a SIP phone) to be able to press keys in order to record call status. is this possible? __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --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] postgres and asterisk
use a simple agi - php is easy to do. --- O.Kamal [EMAIL PROTECTED] wrote: I just need to retrieve a value from a field in a postgres database, and playback this value when someone dial a specific extension. On 1/4/07, Thomas Kenyon [EMAIL PROTECTED] wrote: O.Kamal wrote: I need to retrieve my asterisk to retrieve a values from postgresql, i am looking for some sort of application like *mysql*() app, I found one but it is only available on Suse, is there any way for doing this? Regards, O.Youssef What do you need to do? To get an SQL console with postgres you need to: psql -d database name to start in -U username to connect as ie: psql -d asterisk -U asterisk The location of psql is different depensing upon distribution but usually it's in either /usr/bin/psql or /usr/local/pgsql/bin/psql. I'm not sure if this is what you want, if you want a pretty GUI front-end then you could look at Pgadmin III (www.pgadmin.org) which will run on Windows 2000/XP/2003 or unix/linux running X and requires wxWindows and a pile of common libraries. ___ --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 __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --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] API: how to bridge originated call?
(my pstn calls are coming in thru an upstream asterisk server, so the called and calling phone number is passed as an extension.) when caller comes in on 555, he will go to extension 1234 where he will wait for the API to make a call to 999 for him. how do I bridge the two calls? extensions.conf: ;context where caller comes in [caller] 555,s,1 Answer() 555,s,n UserEvent(Init) ;this lets me know the connection for 555 555,1234,1 Noop(caller waits to be bridged) 555,1234,2 Background(soothingmusic) ;context for connection - is this needed? [connect] from the API: (do I need to create a new context/extension first?) Action: Originate Channel: IAX2/upstream/999 -- calls 999222 thru upsteam IAX Context: ?? Exten: ?? Priority: ?? __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --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] API: how to bridge originated call?
Moises this sounds great! three questions if you don't mind: 1. how is this fitting into 1.6? 2. are there some directions I can follow for downloading the right source and applying your patches? 3. is there a central place for doc on your patches? (if not I would be more than happy to write it) thanks cy --- Moises Silva [EMAIL PROTECTED] wrote: By the way, Chester, please report results to the bug I sent you, is very imortant the users feedback to get this into Asterisk Regards On 1/3/07, Moises Silva [EMAIL PROTECTED] wrote: I have uploaded a working patch for version 1.2.12.1, and other that seems to work in Trunk, but few people is reporting results, you can help to get this into Asterisk, go here: http://bugs.digium.com/view.php?id=5841 The patch I ported to 1.2.12.1 is working fine, I have tested in my servers, is the one called bridge-1.2.12.1.patch, there are other ones that say trunk, obviously only work with the trunk version of Asterisk. Kind Regards On 1/3/07, chester c young [EMAIL PROTECTED] wrote: (my pstn calls are coming in thru an upstream asterisk server, so the called and calling phone number is passed as an extension.) when caller comes in on 555, he will go to extension 1234 where he will wait for the API to make a call to 999 for him. how do I bridge the two calls? extensions.conf: ;context where caller comes in [caller] 555,s,1 Answer() 555,s,n UserEvent(Init) ;this lets me know the connection for 555 555,1234,1 Noop(caller waits to be bridged) 555,1234,2 Background(soothingmusic) ;context for connection - is this needed? [connect] from the API: (do I need to create a new context/extension first?) Action: Originate Channel: IAX2/upstream/999 -- calls 999222 thru upsteam IAX Context: ?? Exten: ?? Priority: ?? __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --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 -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; -- Su nombre es GNU/Linux, no solamente Linux, mas info en http://www.gnu.org; ___ --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 __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --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] ztdummy on 1.6
does anyone know if ztdummy is requires under 1.6 or are they using Linux' rtc? __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --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] API: how to bridge originated call?
how is this fitting into 1.4? - can it be compiled against 1.4 or only 1.2? - if not, are there leanings in that direction? __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --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] Best Hardware for Asterisk Server?
--- Mark Greene [EMAIL PROTECTED] wrote: Hey guys, In your experience what is the best way to go for a production asterisk box in your offices? (In the US) I have had very good luck with Opterons in Tyson rackmounts bought from Newegg. __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --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] Dial - g option
Dial(...|30|g) does not seem to work whereas Dial(...|30|gh) works just fine __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --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
