Re: [asterisk-users] Stumped with Dial - $50 for answer
On Monday 15 January 2007 6:21 pm, Anselm Martin Hoffmeister wrote: could you verify or negate that adding the T option makes it work? That or transfer=no in iax.conf for hte user/peer entries involved. I never thought of IAX2 transfers here, for some reason I thought that Asterisk was terminating the call to TDM itself (one of the two ends). I wouldn't try transfer=mediaonly at this point; remove the transfer capability altogether. -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
Re: [asterisk-users] Stumped with Dial - $50 for answer
its notransfer=yes in iax.conf not transfer=no :) On 16/01/07, Andrew Kohlsmith [EMAIL PROTECTED] wrote: On Monday 15 January 2007 6:21 pm, Anselm Martin Hoffmeister wrote: could you verify or negate that adding the T option makes it work? That or transfer=no in iax.conf for hte user/peer entries involved. I never thought of IAX2 transfers here, for some reason I thought that Asterisk was terminating the call to TDM itself (one of the two ends). I wouldn't try transfer=mediaonly at this point; remove the transfer capability altogether. -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 ___ --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
On Tuesday 16 January 2007 10:09 am, Vicky wrote: its notransfer=yes in iax.conf not transfer=no :) Ahh yes. force consistency in the CLI where it doesn't necessarily belong, but use idiotic variable names in the config files. :-) -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
Re: [asterisk-users] Stumped with Dial - $50 for answer
notransfer has been deprecated in 1.4 in favor of transfer ast_log(LOG_NOTICE, The option 'notransfer' is deprecated in favor of 'transfer' which has options 'yes', 'no', and 'mediaonly'\n); - Andrew Kohlsmith [EMAIL PROTECTED] wrote: On Tuesday 16 January 2007 10:09 am, Vicky wrote: its notransfer=yes in iax.conf not transfer=no :) Ahh yes. force consistency in the CLI where it doesn't necessarily belong, but use idiotic variable names in the config files. :-) -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 -- Jason Parker Digium ___ --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
On Tuesday 16 January 2007 11:58 am, Jason Parker wrote: notransfer has been deprecated in 1.4 in favor of transfer ast_log(LOG_NOTICE, The option 'notransfer' is deprecated in favor of 'transfer' which has options 'yes', 'no', and 'mediaonly'\n); Sure, make an ass out of me, or rather Vicky eggs me on so I do it to myself. :-) I'm very glad to see that change. :-) -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
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 - close
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
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!
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. 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. If you mean me (being in Germany and all that), and if you intend to hand out any money to me (which is not absolutely clear from that statement), please donate to openvpn.org - they accept paypal :-). It is one of the many open source projects whose software I use regularly and have no time ressources (let us not talk about skills :-) to contribute to. BR Anselm ___ --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 - $50 for answer
chester c young wrote:
cannot make Dial(...,,g) work correctly, although Dial(...,,gh) works
just fine. (to make matters worse, it does seem to work sometimes,
although once working or not working between changes it either works or
doesn't work all the time.)
g option to Dial only continues the dialplan if the destination
(called) leg of the call hangs up. It will NOT cause the dialplan to
continue if the source (calling) leg of the call hangs up.
When the calling channel hangs up, Asterisk will send the remaining leg
of the call to exten = h.
My paypal address is [EMAIL PROTECTED]
Example
exten = _91NXXNXX,1,Dial(${PSTN}/${EXTEN:1},,g)
exten = _91NXXNXX,2,Noop(DESTINATION HANGUP)
exten = h,1,Noop(SOURCE HANGUP)
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Re: [asterisk-users] Stumped with Dial - $50 for answer
On Monday 15 January 2007 11:03 am, Eric ManxPower Wieling wrote: 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. I was going to give him the exact same answer, but he specifically said it's not going on when the called party hangs up. I'm using 'g' just fine and it works exactly as you describe, so I'm guessing that something else is the case. :-) -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
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
On 1/15/07, chester c young [EMAIL PROTECTED] wrote: 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. 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, well, that's the problem. We have this with a few of our TDM400's, as well as an old X100P. callprogress=yes did not seem to fix them much. So, the result is that our phone system always thinks we are the ones hanging up. Sometimes that causes a bit of a problem when a person is in a queue and hangs up before they get to an agent. In those cases, the agent gets the dead line. But, when they hang up, the line is freed. In that case, you would just have to use the 'h' flag, and put the rules there, and realize that your system will always believe you hung up. The other option is to get a line with disconnect supervision from your phone company, or some type of digital trunk (PRI, etc). Hope that helps, David ___ --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
Am Montag, den 15.01.2007, 14:22 -0800 schrieb chester c young: 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. Chester, could you verify or negate that adding the T option makes it work? Did you look if there is a bug report somewhere that has to do with call teardown problems when Asterisk is not in the Audio path? BR Anselm ___ --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
Anselm Martin Hoffmeister wrote: Am Montag, den 15.01.2007, 14:22 -0800 schrieb chester c young: 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. Chester, could you verify or negate that adding the T option makes it work? Did you look if there is a bug report somewhere that has to do with call teardown problems when Asterisk is not in the Audio path? Curious - is this still a $50 thread? ___ --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
