| 00:01.07 | Micc | Why is it that maxexpirey does nothing? |
| 00:03.52 | Micc | I set it to 160 but still I have a bunch of panasonic KX-TGP500's that register with 3600 |
| 00:05.43 | ManxPower | ritzt3ch, you have port 5 configured then |
| 00:06.06 | ManxPower | but that in an indicaton that 3 might be crossed with whatever is wired in to 5 |
| 00:10.34 | *** join/#asterisk ruben23 (~ITadmin@125.212.40.2) |
| 00:16.39 | *** join/#asterisk fofware (~Fabian@host106.186-110-110.telecom.net.ar) |
| 00:19.35 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 00:22.42 | *** join/#asterisk drclue (~drclue@ip-65-49-164-42.wireless.dyn.beamspeed.net) |
| 00:24.11 | drclue | Howdy all. When using the Queue application , what is the syntax for placing a call at a particular position in the call queue, and additionally if that position does not exist will the call simply be appended? |
| 00:25.06 | [TK]D-Fender | drclue: There is none |
| 00:25.25 | p3nguin | Can't that only be done via AMI? |
| 00:25.52 | pabelanger | marksaitis: It ships with asterisk 1.4+ |
| 00:26.03 | thehar | fra la la |
| 00:26.25 | drclue | That's odd , I've seen vague references to positionally adding calls to the queue in recent versions of Asterisk , but specific details have been a little hard to come by |
| 00:27.39 | [TK]D-Fender | drclue: So far the only vague reference I've seen is you asking if it existed |
| 00:27.46 | marksaitis | Can somebody, PLEASE, recommend me the best, easy and well thought, fully featured, free asterisk 1.7 GUI to manage all of it? |
| 00:28.03 | drclue | I do all my asterisk applications in HTML5 these days via an Asterisk AMI middleware I wrote that uses Web Sockets and XML. |
| 00:28.18 | thehar | Asterisk has a 1.7? |
| 00:28.22 | thehar | boggles |
| 00:28.51 | marksaitis | thehar? I think 1.7 is only the AsteriskNOW platform |
| 00:29.00 | marksaitis | not the asterisk itself |
| 00:29.15 | marksaitis | Can somebody, PLEASE, recommend me the best, easy and well thought, fully featured, free asterisk 1.7 GUI to manage all of it? |
| 00:29.25 | thehar | then asterisknow would be the gui |
| 00:29.28 | thehar | why are you spamming the channel? |
| 00:29.45 | [TK]D-Fender | thehar: No |
| 00:29.51 | [TK]D-Fender | marksaitis: FreePBX |
| 00:29.51 | thehar | heh |
| 00:29.55 | marksaitis | asterisknow is outofthebox thing |
| 00:30.10 | marksaitis | not the gui |
| 00:30.15 | thehar | freepbx comes attached to the iso on the asterisknow versions |
| 00:30.35 | marksaitis | is that the best one out there??????? freepbx? |
| 00:30.48 | drclue | @fender I've seen references in the official asterisk documentation but the docs were not as clear as I would like |
| 00:30.51 | thehar | i would use that as that is what digium has tested |
| 00:31.26 | [TK]D-Fender | marksaitis: For what you asked, yes |
| 00:31.56 | [TK]D-Fender | drclue: And where would I see this? |
| 00:32.24 | marksaitis | Cant believe it, I thought thers something better and easier |
| 00:32.49 | *** join/#asterisk sahafeez (~sahafeez@65-119-47-100.dia.static.qwest.net) |
| 00:34.17 | drclue | @fender http://www.asterisk.org/docs/asterisk/trunk/applications/queue |
| 00:34.21 | [TK]D-Fender | marksaitis: What is "hard" about it? |
| 00:34.32 | ManxPower | marksaitis, telecom and pbxs are complicated things. There is nothing in the world that will make it uncomplicated. |
| 00:35.20 | [TK]D-Fender | drclue: Ok. So what happens when you try this? |
| 00:35.21 | ManxPower | ~toolkit |
| 00:35.22 | infobot | hmm... toolkit is Remember Asterisk isn't really a PBX. Asterisk is a TOOLKIT that helps you build a PBX from scratch, much like libraries help you build an application from scratch. |
| 00:36.03 | [TK]D-Fender | DrAnd what ver of * are you running? |
| 00:36.29 | marksaitis | woops, I think I confused freepbx with something else quite simmilar |
| 00:36.38 | marksaitis | that one doesnt look that bad |
| 00:36.43 | marksaitis | I will give it a spin |
| 00:37.00 | drclue | @fender I've not actually tried it yet but I'm about to, and wanted to see if someone else had already used it to see what their syntax was. I'm running 1.6 |
| 00:37.39 | [TK]D-Fender | drclue: I'm sure their syntax follows the INSTRUCTIONS just like it says. Why would it be any DIFFERENT? |
| 00:38.31 | ritzt3ch | ohy vey ima TRY and move outbound to sip trunks now and take the dam HAVOK of analog |
| 00:40.24 | drclue | @fender Not saying it is different, but seeing the syntax as used in a working example saves me from some of the grief I find when the many different sets of version specific documentation collide. Especially since at the same time I'm working out a new javascript class and want to minimize the variability |
| 00:40.56 | drclue | BUT if nobody has a working example , I'll just go ahead and play with it |
| 00:41.43 | [TK]D-Fender | drclue: .... "core show application queue" <--- read YOUR VERSION'S INSTRUCTIONS |
| 00:41.57 | [TK]D-Fender | drclue: Working sample? Holy shit, just count your COMMAS |
| 00:42.16 | [TK]D-Fender | drclue: If you can't count commas you don't deserve to call yourself a programmer. |
| 00:43.18 | [TK]D-Fender | drclue: this staring at some page that we can't prove matches your version business is just nonsense |
| 00:45.18 | drclue | @fender I give you an A++ in the school of wise arse. :) If you have not experienced the issues of version centric documentation as relates to asterisk and cannot recognize the value in crowd sourcing a question like this then perhaps that is more a reflection of yourself |
| 00:47.01 | [TK]D-Fender | drclue: Asking if a parameter you see listed works is like asking your grocery "is this piece of fruit edible?" for every piece of fruit in the damn store |
| 00:47.47 | [TK]D-Fender | drclue: Means you are being a total pussy and doubt the instructions that say "Do X and you'll get Y", and need someone to TELL you it does what it says when you're too lazy to even validate your own wasted suspicions |
| 00:48.50 | ManxPower | drclue, the thing is, "crowdsourcing" will give you the wrong information. |
| 00:49.09 | ManxPower | Asterisk changes to fast for you to do anything other than check the applications for YOUR version of Asterisk. |
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| 00:49.43 | drclue | @fender You expressions are nonsequitur. It is a FACT that often asterisk documentation differs by version. You yourself swore up and down there was not even such functionality, so you are not as omnipresent as you would like folks to think you are |
| 00:49.53 | [TK]D-Fender | drclue: Dial(Technology/Resource[&Technology2/Resource2[&...]][,timeout[,options[,URL]]]) <--- it says I can pass a timeout. Does that really work? Please provide 10 people to back up your answer, a full set of fingerprints and a photo of yourself posing with a Dalmatian. Thanks. |
| 00:50.30 | [TK]D-Fender | drclue: And you see it tell you to your face that there is, and you don't even TRY it. Do you need a giant flashing neon sign? |
| 00:51.11 | bougyman | drclue: he really is pretty omnipresent. |
| 00:51.41 | [TK]D-Fender | drclue: DrWould you need an entire team of Microsoft engineers to tell you ever command you might have heard about for Excel actually DOES what it says? |
| 00:51.42 | underdog | cisco ata 18x not hanging up FXS port when the remote caller disconnects <-- anyone have any help on this issue? |
| 00:52.01 | underdog | found some info on CPC for the SPA's but this is the old ata's |
| 00:52.21 | ManxPower | underdog, do you have real phones plugged into the ATA? |
| 00:52.24 | [TK]D-Fender | underdog: do you see the complete CANCEL transation and all in SIP debug you could show us? |
| 00:52.28 | underdog | cordless |
| 00:52.51 | drclue | No , not a giant neon sign , but it would be handy to verify that someone else has actually gotten this to work and if the crowd has indeed been down that path to elicit further any issues they might have experienced. But fender , you keep being yourself as each forum of discussion needs one of you for at least amusement value |
| 00:53.12 | underdog | I can get the debug info later...trying to herd kids to clean up right now...excercise in futility |
| 00:53.59 | underdog | ManxPower: I have a panasonic cordless base plugged into it |
| 00:54.01 | [TK]D-Fender | drclue: Are you going to ask "have you gooten X" to work for EVERY parameter of EVERY dialplan app? |
| 00:54.17 | [TK]D-Fender | drclue: Otherwise why is THIS one special? |
| 00:55.26 | drclue | @fender actually most applications are fairly common replete with examples and reasonably available commentary. This one is sorta special as even your greatness did not know the feature existed |
| 00:56.13 | [TK]D-Fender | drclue: You see the parameter there. Congratulations. Go use it. Maybe there is little commentary because it doesn't NEED any. |
| 00:56.30 | [TK]D-Fender | drclue: Maybe it jsut does what it says so nobody has any REASON to have to bring it up |
| 00:56.44 | [TK]D-Fender | drclue: You would be fishing for problems that don't exist |
| 00:57.00 | [TK]D-Fender | drclue: All because you can't seem to just TRY it for yourself |
| 00:57.09 | [TK]D-Fender | drclue: And why is that? Just lazy? |
| 00:57.24 | [TK]D-Fender | drclue: Do you need a support group? |
| 00:57.39 | [TK]D-Fender | drclue: Terminal co-dependence? |
| 00:59.25 | drclue | Obviously I had to show YOU that it existed. Crowd sourcing for obscure topics is a valuable resource in most cases , allowing one to from the expressions of others, better plan implementations. Just because you don't see the value in such things does not mean that such value exists. Rather you seem to see it as an opportunity to BE YOURSELF. |
| 01:00.21 | marksaitis | dear oh dear, to install a damn GUI Manager, I need to read a whole bible of shit on freepbx install guide, whereas half of it is complete nonsense |
| 01:00.23 | drclue | Since you have NOTHING BUT LIP to offer , perhaps you should simply crawl back under your rock |
| 01:00.51 | [TK]D-Fender | drclue: why do you have to doubt an option that you seem to see clear documentation for does what it says? |
| 01:01.02 | [TK]D-Fender | drclue: drclue Where is the logic in that? |
| 01:01.25 | drclue | @fender if I have to explain that one to you , perhaps your asterisk experience is even less than you say |
| 01:01.27 | [TK]D-Fender | drclue: There becomes no reason to not second-guewss EVERYTHING once you do that. |
| 01:01.52 | [TK]D-Fender | drclue: Any other documented parameters you now doubt as well? |
| 01:02.30 | [TK]D-Fender | drclue: Except that your "second guessing" requires others to corroborate what the docs say since youa re unwilling to try it yourself. |
| 01:02.38 | p3nguin | If most things described are not working as they say, then it might make sense to have doubt. |
| 01:02.47 | drclue | I regularly find documentation whose lack of specific details causes me to question same and rightly so. |
| 01:02.53 | [TK]D-Fender | drclue: You are clearly not part of your own solution |
| 01:03.13 | [TK]D-Fender | drclue: It says "puts them in position X if possible". What is there to question? |
| 01:03.16 | drclue | @fender and obviously your not part of the solution either |
| 01:03.27 | p3nguin | I dare you to name 37 things in Asterisk that do not work as documented. |
| 01:04.25 | [TK]D-Fender | drclue: Apparently the only solution for you is if someone does your research for you. Research which noone has any reason to do. Proving something works just for you doesn't sound like a productive use of their time especially when you won't even do it for yourself |
| 01:04.55 | [TK]D-Fender | p3nguin: Nice abstract number. ++ |
| 01:06.52 | [TK]D-Fender | 1. Place 10 calls in queue. 2. place 11th call and have it call the app inserting to position 6. Look at Queue. <- no reason this should take longer than this conversation. |
| 01:07.11 | [TK]D-Fender | But no... someone ELSE has to tell you it works. |
| 01:07.13 | drclue | Often what works and what does not differs by version , however the available documentation often neglects to state exactly where the topic it discusses begins and ends version-wise. Sometimes commands get renamed , sometimes there are undocumented bugs or considerations not covered. |
| 01:07.36 | drclue | @fender the only thing really taking time in this conversation is dealing with YOU |
| 01:07.55 | [TK]D-Fender | drclue: You aren't testing it. You aren't looking at YOUR VERSION's instructions. |
| 01:08.08 | [TK]D-Fender | drclue: You could have had your own answer on your own by now. |
| 01:08.14 | underdog | [TK]D-Fender: http://pastebin.com/rVxtmiv7 <-- this is a pb of what it did when I hung up from the remote end. 2005 is the ATA |
| 01:08.58 | [TK]D-Fender | [20:41]<[TK]D-Fender>drclue: .... "core show application queue" <--- read YOUR VERSION'S INSTRUCTIONS |
| 01:09.10 | [TK]D-Fender | drclue: 1/2 hour ago. Could have read your own instructions. |
| 01:09.28 | [TK]D-Fender | drclue: Could have TRIED. But you don't. You need someone ELSE to do it. |
| 01:09.39 | [TK]D-Fender | drclue: Makes me suspect you don't know enough * to even test it |
| 01:09.45 | underdog | i plugged an old touch tone phone into the ata to rule out any cordless phone issue... |
| 01:10.35 | [TK]D-Fender | [20:36]<drclue>@fender I've not actually tried it yet but I'm about to, and wanted to see if someone else had already used it to see what their syntax was. I'm running 1.6 <- 1.6 isn't a version. Or a branch. |
| 01:11.03 | [TK]D-Fender | underdog: that isn't SIP debug |
| 01:11.41 | [TK]D-Fender | underdog: and who hung up, and which is the ATA? |
| 01:11.54 | underdog | 2005 is the ATA...i hung up from 2000 ext |
| 01:12.00 | underdog | hold on...i'll enable sip debug |
| 01:12.05 | underdog | get that to ya |
| 01:12.18 | [TK]D-Fender | underdog: Hold on that. |
| 01:12.39 | [TK]D-Fender | underdog: Ok, so 2000 hung up. What are you expecting to have happen on 2005? |
| 01:13.07 | drclue | @fender , those instructions do not cover all the conversation I would find of value from people who might have deployed the feature. Being as it is a rarefied topic , it's reasonable to ask the crowd and see what their take is on it. In the course of programming for Asterisk , I have discovered lots of things that are never made clear in the documentation. |
| 01:13.30 | [TK]D-Fender | drclue: Deploying a feature? You plug a NUMBER in there. Stop making it sound like highly integrated magic |
| 01:13.45 | underdog | the ata off-hook light stays on and so the phone starts playing fast busy after about 4 seconds |
| 01:13.50 | [TK]D-Fender | DrunkenMaster: You need a COMMITEE jsut to test a NUMBER is a field. |
| 01:13.55 | [TK]D-Fender | drclue: even. |
| 01:13.56 | underdog | so the ata should go on-hook when the remote end hangs up |
| 01:13.56 | drclue | @fender , just how much AMI programming do you do? |
| 01:13.56 | [TK]D-Fender | gah |
| 01:14.08 | [TK]D-Fender | drclue: I've done a bit |
| 01:14.33 | drclue | Then obviously you;ve run across many things that are not as simple as the documentation alone |
| 01:14.37 | [TK]D-Fender | underdog: no, the ATA acts as an FXO. the telco can't go "onhook" to you |
| 01:14.41 | [TK]D-Fender | underthat's backwards |
| 01:15.10 | [TK]D-Fender | underdog: PHONES go onhook. the ATA is the telco side, not the phone side. |
| 01:15.48 | [TK]D-Fender | underdog: When you pickup up a phone the loop completes and STAYS complete. it has to otherwise the telco won't know when you pickup/hangup the phone again |
| 01:16.48 | [TK]D-Fender | underdog: normally when the ATA is told to end things it issues a reorder tone |
| 01:17.39 | [TK]D-Fender | underdog: but that phone is OFF-HOOK, because that is its state. Don't means its a practical thing... is just doesn't know to do anything else. Its jsut sitting there off-hook. |
| 01:19.22 | [TK]D-Fender | [21:14]<drclue>Then obviously you;ve run across many things that are not as simple as the documentation alone <- You need someone else to specifically validate a feature whose intent is clear. What if they wrote a whoe page sying that yes, this does indeed put the right where it says it will? How much do you need to see before you reach the point of not having to crowdsource it? |
| 01:19.49 | underdog | i understand what you are saying...but there has to be a way for the ata to disconnect the loop when the remote end hangs up and asterisk performs a hangup...otherwise every ata out there would require the end user to hangup or they'd get the fast busy tones |
| 01:19.59 | underdog | it's not a huge deal...just annoying |
| 01:20.06 | [TK]D-Fender | underdog: that is PRECISELY what happens |
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| 01:20.30 | [TK]D-Fender | underdog: Same with every POTS telco line on the planet |
| 01:20.50 | [TK]D-Fender | underdog: Have you never had an analog line before? |
| 01:22.17 | drclue | @fender When I research a command and find as little documentation about it or a feature it is indeed the time to ask around, as unlike a vast majority of the features I work with , this one has scant documentation and little discussion of side-effects , gotchas and other topics which are far more common in other features. |
| 01:22.35 | underdog | [TK]D-Fender: I get it...I've been using cells for the majority of last 10 years and other phones were VoIP or other systems that hang up when the remote end hangs up |
| 01:22.48 | underdog | i wasn't thinking |
| 01:22.56 | [TK]D-Fender | drclue: Maybe it has none. How much documentation is there on the timout parameter for it? |
| 01:23.37 | [TK]D-Fender | drclue: its function doesn't take 10 pages. |
| 01:23.49 | [TK]D-Fender | drclue: No news is good news? |
| 01:24.26 | [TK]D-Fender | drclue: maybe everyone using it is happy and were successful the memnt they tried because it DOES do exactly what it says. |
| 01:24.49 | drclue | While there is plenty of documentation and discussion about previously existing features in the queue application all over the net , this feature has hardly a mention at all |
| 01:24.56 | [TK]D-Fender | drclue: Would bad news make you feel better. If noone does your testing for you does that mean that you keep asking every day about it? Or just give up entirely? |
| 01:25.25 | [TK]D-Fender | drclue: Again, if it work and is simple as the instructions say, why should anyone be talking about it? |
| 01:26.02 | [TK]D-Fender | drclue: Nobody talks about how Wait()'s parameter specifying how long to wait actually works. because it does. What is there to talk about? |
| 01:26.36 | underdog | [TK]D-Fender: i feel stupid now....hahaha |
| 01:26.43 | [TK]D-Fender | drclue: So again, if you don't get your crowd-sourced answer, are you going to keep asking for a long time? Would you pass it over in short order? |
| 01:26.54 | underdog | or maybe it's laziness....hang up for me dangit |
| 01:26.56 | underdog | heh |
| 01:27.14 | [TK]D-Fender | drclue: If that is the 4th parameter of a command, would that stop you from looking at the 5th because you're not sure if you should add the extra comma to "skip it? |
| 01:27.35 | drclue | If existing documentation of an obscure feature is not well discussed I will typically ask around to see what the experience of others is in using that feature. If in the end I'm pretty much alone in using that feature , then I just go for it and let the chips fall where they may. |
| 01:28.17 | [TK]D-Fender | drclue: What makes it "obscure"? Just because nobody is talking about it? Maybe they all take it for granted because it always did exactly what it clearly says that it does |
| 01:29.00 | drclue | Or just as likely (As in your case) not many people even know about the feature |
| 01:29.04 | [TK]D-Fender | drclue: and why aren't you just trying it yourself? Don't even have an * install to test it on? |
| 01:29.52 | [TK]D-Fender | drclue: Even if it is obscure, that lack of people would seem to grind your productivity to a halt because you need someone else to validate something you won't look at yourself. |
| 01:30.52 | drclue | Actually I have the code written to use it, but life is not always that simple. But since you know everything about asterisk (of course except the feature being discussed) , your the perfect one to ask RIGHT? |
| 01:31.20 | [TK]D-Fender | drclue: Want to know if it works? Go try it. Why are you so lazy? what makes someone else's statement that it "works for them" valid? maybe they are wrong and didn't notice. What does it take for you to trust their input? How many people do yuo need to hear from? |
| 01:31.56 | [TK]D-Fender | drclue: "life is not that simple". Well in this case it is. Put 10 calls in queue. inster #11 into position 6. LOOK.. The end. |
| 01:32.10 | [TK]D-Fender | drclue: You are supposed to have all the tools to do this. |
| 01:32.16 | [TK]D-Fender | drclue: why aren't you doing it? |
| 01:32.27 | drclue | Not just to know if the feature works , but also other issues that might not show up right away that others who have used it for a time might know about , which ceratinly would not be you |
| 01:32.38 | [TK]D-Fender | drclue: Nothing says it doesn't. |
| 01:33.40 | drclue | Not having a document to tell you that there is a snake behind that rock is not proof that such a snake does not exist |
| 01:33.48 | *** join/#asterisk fofware (~Fabian@host106.186-110-110.telecom.net.ar) |
| 01:34.07 | [TK]D-Fender | drclue: And if you live your life like there is a snake behind every rock you'll die of a heart attack long before your time. |
| 01:34.17 | dongs | lol. so i boot asteriskNOW and its asking me to install 1.6 or 1.4 and FreePBX or AsteriskGUI. seeing how last asterisk I seriously used was like 1.0 what do i pick? |
| 01:34.28 | dongs | ... and it just started booting something. |
| 01:34.30 | dongs | without waiting |
| 01:34.59 | [TK]D-Fender | dongs: It probably did... just not long enough for your indecision :) |
| 01:35.10 | dongs | well uh |
| 01:35.12 | dongs | so what do i choose |
| 01:35.21 | [TK]D-Fender | dongs: what do you want to do? |
| 01:35.32 | dongs | i want to install it and have it work. |
| 01:35.38 | dongs | and not edit any weird text files |
| 01:35.43 | dongs | this is 2010, not 2003 |
| 01:35.59 | dongs | which one gets me that goal? |
| 01:36.30 | [TK]D-Fender | dongs: then 1.4 & FreePBX. And you'll probably have to tweak a little thing or two somewhere along the way. |
| 01:36.39 | [TK]D-Fender | dongs: No matter what |
| 01:36.44 | [TK]D-Fender | dongs: Deal with it |
| 01:36.46 | underdog | and edit some files |
| 01:37.14 | dongs | theres nothign that "just works"? |
| 01:37.25 | [TK]D-Fender | dongs: and once you've started your install, please take the second door to your left for support... |
| 01:37.27 | [TK]D-Fender | ~freepbx |
| 01:37.27 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there, or fpbx |
| 01:37.32 | dongs | haha. |
| 01:37.33 | underdog | well, for enough money anything can "just work" |
| 01:37.42 | [TK]D-Fender | dongs: It works... but people tend to break stuff. |
| 01:37.47 | underdog | for free...well...beggers can't be choosers |
| 01:38.02 | [TK]D-Fender | underdog: sure they can. the can also fuck off :) |
| 01:38.41 | underdog | just messing around with freepbx has shown it has a lot of limitations |
| 01:38.50 | underdog | you are at the mercy of what they allow you to do from the gui |
| 01:38.53 | dongs | [TK]D-Fender: break stuff how? if I never edit anything text and just use GUI, does it work? |
| 01:39.07 | [TK]D-Fender | underdog: Of course. Its a cookie-cutter system, so stop looking for MUFFINS |
| 01:39.22 | [TK]D-Fender | dogDoesn't what you want it to DO <- |
| 01:39.29 | [TK]D-Fender | dongs: ^ |
| 01:39.43 | [TK]D-Fender | dongs: "Just work" doesn't eman what YOU want might not require something special. |
| 01:40.21 | saxa | pabelanger: thx. |
| 01:40.29 | dongs | i just want some lines to ring phones, when all those lines are busy, last line will play some 'we're busy' message, and have some sorta voicemail/wahtever afterhours |
| 01:40.32 | dongs | does that work? |
| 01:41.45 | pabelanger | dongs: depends, what type of interface are you using? |
| 01:41.53 | dongs | all ip |
| 01:42.19 | [TK]D-Fender | dongs: Fine, go for it |
| 01:44.56 | dongs | i coulsnt even find anythign about "Asterisk-GUI" what is that anyway. |
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| 01:45.01 | dongs | wiki page was useless. |
| 01:45.36 | [TK]D-Fender | dongs: just forget it |
| 01:46.16 | pabelanger | dongs: you'll need to create a dialplan, using func_group |
| 01:47.47 | pabelanger | *CLI> core show function GROUP_COUNT |
| 01:48.12 | dongs | i thought the whole point was that i never have to deal with CLI shit. |
| 01:48.18 | [TK]D-Fender | pabelanger: No |
| 01:48.35 | [TK]D-Fender | dongs: Ignore that |
| 01:50.29 | pabelanger | Then your in the wrong channel |
| 01:59.35 | underdog | you could always wait for Steve Jobs to make it...then it'll be all GUI all the time :) |
| 01:59.51 | underdog | but it'll only cost you $5000 |
| 02:02.49 | *** part/#asterisk ManxPower (~manxpower@user-24-214-153-32.knology.net) |
| 02:03.17 | underdog | pabelanger: could you also do someting like exten => s,1,Dial(SIP/110&SIP/112,20,tr) |
| 02:03.32 | underdog | or is func_group a cleaner way? |
| 02:05.56 | pabelanger | underdog: Yup, I may have misunderstood the issue. I focused on 'when all those lines are busy, last line will play some 'we're busy' message', meaning into the Asterisk box, not the extensions he was going to ring. |
| 02:06.39 | pabelanger | Regardless, it could be handled may ways. A queue, as another example |
| 02:06.49 | *** part/#asterisk drclue (~drclue@ip-65-49-164-42.wireless.dyn.beamspeed.net) |
| 02:07.13 | underdog | yeah...as with most things involving *nix |
| 02:09.55 | [TK]D-Fender | pabelanger: 2 words : Mixed Nuts :p |
| 02:18.17 | *** join/#asterisk b0gatyr_ (~b0gatyr_@adsl-10-116-246.mia.bellsouth.net) |
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| 02:26.07 | eMBee | tzafrir: sorry missed your reply. WARNING chan_dahdi.c: Channel '1' ignored. comes on restart, didn't check if it comes on reload too |
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| 02:39.04 | itiliti | good evening |
| 02:39.26 | itiliti | exit |
| 02:39.30 | *** part/#asterisk itiliti (~itiliti@mail.applync.com) |
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| 03:13.39 | sshock | Anyone here played with google voice? |
| 03:14.18 | sshock | I've got one way audio when making an outside call into *. |
| 03:14.55 | sshock | Calling to a google voice number, that forwards to my voip provider (gizmo), which then goes to *. |
| 03:15.33 | sshock | If I call directly to * or to my Gizmo number, it works fine. So this must be a problem with Google Voice, right? |
| 03:16.06 | sshock | Maybe they don't like being used as a free DID.... |
| 03:16.29 | ChannelZ | hard to say |
| 03:16.51 | ChannelZ | what way audio do you have? |
| 03:18.39 | sshock | from the outside line into * |
| 03:19.28 | sshock | I've monitored the RTP packets back to the WAN side of my * box and they are making it there fine |
| 03:19.49 | sshock | I think somewhere between Gizmo and GV they are getting dropped... |
| 03:21.14 | pabelanger | sshock: Yes, what version of asterisk you running? |
| 03:21.56 | pabelanger | sshock: http://blog.polybeacon.com/2010/10/17/asterisk-1-8-and-google-voice/ |
| 03:22.01 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 03:22.21 | sshock | * version 1.4.21.2 |
| 03:23.12 | pabelanger | sshock: Ya, I suspect that is the problem. 1.8+ works like a champ. |
| 03:23.42 | sshock | :( |
| 03:24.15 | sshock | of course, that is something totally different I think |
| 03:24.22 | sshock | that is connecting * to your google talk |
| 03:24.41 | sshock | I'm connecting * to Gizmo, and GV can forward calls to your Gizmo account |
| 03:25.08 | sshock | I wouldn't even use GV, except that my Gizmo account has no DID |
| 03:25.40 | sshock | and ever since Google bought Gizmo, you can't add DIDs to your Gizmo account, nor create new accounts |
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| 03:49.20 | pabelanger | sshock: Ya, sounds like google is porting all functionality from gizmo to its servers |
| 03:51.23 | ChannelZ | jizmo |
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| 03:57.52 | sshock | yeah |
| 03:58.23 | *** join/#asterisk ritzt3ch (~ritzt3ch@ip70-189-221-21.lv.lv.cox.net) |
| 03:58.38 | sshock | who was it that mentioned flowroute to me the other day? |
| 03:59.26 | ritzt3ch | what are some test tools/ logs / tcpdump commands for things to see where my SIP trunk is failing |
| 03:59.38 | ritzt3ch | inbound is fine but not outbound |
| 04:00.31 | sshock | ritzt3ch: I usually dump my traffic using "tcpdump -i eth0 -s 0 -w out.dat" |
| 04:00.40 | sshock | then I open out.dat using wireshark |
| 04:00.58 | sshock | wireshark is awesome; it can even decode and playback RTP streams |
| 04:01.39 | ritzt3ch | yea i use to use it WAY log ago i can just take that .dat and change it to a pcap file then ? |
| 04:02.31 | ritzt3ch | im geeting a bunch of these but not sure why |
| 04:02.31 | ritzt3ch | Got SIP response 301 "Moved Permanently |
| 04:02.50 | sshock | I think it already is essentially a pcap file, but not sure |
| 04:03.27 | sshock | anyone had experience porting their existing phone number to a voip provider? is it usually a pain or not too bad? |
| 04:06.03 | ritzt3ch | theres a fee and a hefty wait |
| 04:06.42 | sshock | just read on flowroute.com they charge only $25; not too bad |
| 04:06.42 | *** join/#asterisk Sheath (Smegma@unaffiliated/coil) |
| 04:09.18 | ritzt3ch | in the wireshark trace in my 100 trying it shows the From and the To: going to the same address ? |
| 04:09.56 | ritzt3ch | then i get a 301 moved permanently Reason code "the endpoint is not being served here" |
| 04:14.23 | sshock | ritzt3ch: I think to and from being the same is normal when doing things like, e.g., REGISTER |
| 04:15.27 | ritzt3ch | damm i was hoping to find where the heck it breaks |
| 04:15.28 | ritzt3ch | <PROTECTED> |
| 04:15.52 | ritzt3ch | how do you know what context to put it in anyways cause im not seeing a SIP trying outbound right |
| 04:19.58 | ritzt3ch | i might not even have the right context set hmmm |
| 04:20.39 | ritzt3ch | i do have a request REGISTER sip:10.10.10.2 |
| 04:20.54 | ritzt3ch | but Contact binding shows sip:s@10.10.10.2 |
| 04:22.01 | sshock | have you already tried enabling nat and stuff |
| 04:22.38 | sshock | (BTW, I'm new to * too so I don't know much) |
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| 04:30.45 | ritzt3ch | chan_sip.c: No such host: sip-mitel |
| 04:33.34 | pabelanger | hehe, mitel :) |
| 04:43.17 | sshock | I want to set up voicemail now, and I have a question about users.conf. |
| 04:43.56 | sshock | So if I only have SIP users, should I just specify these settings lik vmsecret inside sip.conf ? |
| 04:45.51 | [TK]D-Fender | user.conf ... BLEH |
| 04:46.53 | ritzt3ch | yea if you know mitel enough there always bugs involed |
| 04:50.36 | sshock | I'm confused about the relationship between voicemail.conf and other conf files... |
| 04:52.19 | [TK]D-Fender | sshock: its largely one-way. |
| 04:52.29 | [TK]D-Fender | sshock: Things refer to voicemail. |
| 04:52.55 | sshock | ok, so I start with my dialplan, which can call VoiceMail(number@context) |
| 04:53.18 | [TK]D-Fender | sshock: Sure.... |
| 04:53.20 | sshock | so then that looks up that number in the [context] section in voicemail.conf; so far so good... |
| 04:53.26 | [TK]D-Fender | yes |
| 04:53.40 | sshock | there I define my password/pin, a name, and an email |
| 04:54.12 | [TK]D-Fender | sshock: And other box specific options, etc |
| 04:54.32 | [TK]D-Fender | sshock: General other bits, formatting, TZ.... |
| 04:54.33 | sshock | ok |
| 04:54.59 | sshock | so then for listening to my voice mails, I do that using VoiceMailMain(number@context) |
| 04:55.23 | sshock | so what the heck is users.conf for? |
| 04:55.49 | [TK]D-Fender | sshock: people using asterisk-gui |
| 04:56.04 | [TK]D-Fender | sshock: If you aren't then its worthless |
| 04:56.04 | sshock | oh, ok; I may or may not even use that... |
| 04:56.39 | sshock | ok, then what about sip.conf? should I be putting anything in there related to voicemail? |
| 04:56.40 | pabelanger | ritzt3ch: Yar! I have a 3300 sitting next to me in my home office. Fun times |
| 04:57.12 | [TK]D-Fender | sshock: Do you want a SIP device to know there is a VM in a box? |
| 04:57.48 | sshock | hmm, sure; I guess that could be useful |
| 04:58.04 | sshock | I see this vmexten |
| 04:58.05 | [TK]D-Fender | sshock: there you go. |
| 04:58.22 | sshock | so let's say I set vmexten=voicemail, then that means in extensions.conf I should have a [voicemail] context |
| 04:58.23 | [TK]D-Fender | sshock: vmexten = largely worthless |
| 04:58.32 | sshock | oh, ok |
| 04:58.33 | [TK]D-Fender | no |
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| 05:00.10 | sshock | ok, so mailbox= is what I'm looking for |
| 05:01.01 | sshock | So I just set mailbox=number@context in my phone/device section? |
| 05:01.20 | sshock | doesn't it need the mailbox password? |
| 05:02.19 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
| 05:06.49 | [TK]D-Fender | no, this only NOTIFIES the device there is VM in a box |
| 05:06.59 | [TK]D-Fender | It has nothing to do with getting them INTO the box |
| 05:07.14 | sshock | ok |
| 05:07.24 | sshock | so what's the difference between a notification email and a pager email? |
| 05:08.03 | [TK]D-Fender | sshock: primary can have the VM recording ATTACHED to it |
| 05:08.38 | sshock | ok |
| 05:09.05 | sshock | well, I don't have a pager anyways so I probably won't use that :) |
| 05:10.05 | sshock | although suppose I wanted it to send email to multiple email address; how would I do that? |
| 05:11.00 | [TK]D-Fender | sshock: Set up the mail server to distribute from the one you send it to |
| 05:11.07 | [TK]D-Fender | sshock: AKA, not 8's job |
| 05:11.09 | [TK]D-Fender | * |
| 05:11.17 | sshock | ok, I can accept that |
| 05:12.14 | sshock | is there any reason I should or should not equate mailbox number with the extension ? |
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| 05:14.58 | [TK]D-Fender | sshock: whatever way makes sense |
| 05:16.33 | Montys | elo, In your personal opinion what is a best way to implement a high-availability asterisk solution?. I know for a fact that Asterisk does not provide a easy way implement load-balancing or redundancy, so I think that I will have to use other tools and/or software in conjunction asterisk in order to achieve that. Could you give a suggestion about the softwares and how to do it? |
| 05:17.14 | [TK]D-Fender | SER <- |
| 05:17.23 | [TK]D-Fender | Montys: Highly Google-able |
| 05:17.48 | Montys | I was thinking: OpenSer (load Balacing) + Asterisk/ DUNDi but I not sure if this is the best way |
| 05:18.47 | Montys | [TK]D-Fender: I know is google-able, But I want to know your opinion :'( |
| 05:19.43 | [TK]D-Fender | Montys: It's my optinion that statistically thats what the big users use |
| 05:20.15 | [TK]D-Fender | DUNDi isn't meant for HA. Someone hacked a "solution out of it". Just because you paint stripes on a horse doesn't mean you should call it a zebra |
| 05:25.05 | Montys | DUNDI will be used to figure out where the SIP Phone is register -- you right, it's not meant for HA |
| 05:25.55 | [TK]D-Fender | Montys: Not even. It is an EXTEN finder. that isn't related to a device. |
| 05:26.29 | [TK]D-Fender | Montys: Has nothing to do with distributed registrations really.. unless someone uses regexten and that propageates |
| 05:26.38 | [TK]D-Fender | but that starts getting fugly... |
| 05:27.23 | Montys | [TK]D-Fender: I accept defeat... I forgot that devices and extensions are different things in * |
| 05:30.37 | sshock | how come when I go to the mailbox it's asking me for the mailbox (number) and password? |
| 05:30.56 | sshock | i already told it the mailbox number with VoiceMailMain(100) |
| 05:31.34 | [TK]D-Fender | sshock: PASTEBIN <- |
| 05:31.36 | [TK]D-Fender | ~pb |
| 05:31.36 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 05:31.45 | sshock | and who is this "Comedian Mail" ? |
| 05:32.14 | sshock | want me to paste my extensions.conf? |
| 05:32.19 | sshock | ok |
| 05:35.08 | sshock | http://pastebin.com/f43s5k7F |
| 05:35.23 | [TK]D-Fender | sshock: VM config and your actual call CLI |
| 05:35.25 | sshock | I XXX out a few things; look at the [voicemail] context |
| 05:35.34 | sshock | oh, now you tell me |
| 05:36.43 | jkroon | [TK]D-Fender, you object to using regexten? Or perhaps have a better suggestion? I'm in a similar situation now where users can register to one of many asterisk servers, and IMHO regexten + DUNDI provides a reasonably clean way of locating those users. |
| 05:36.44 | [TK]D-Fender | sshock: Odd that nowhere in your dialplan do you actually have a place to LEAVE voicemail |
| 05:36.56 | sshock | [TK]D-Fender: just because I haven't got to it yet |
| 05:37.04 | sshock | don't worry; that's next :) |
| 05:37.21 | [TK]D-Fender | jkroon: a series of hacks that "kinda" work, but YUCK. thats what real distributed proxies like SER are for |
| 05:37.47 | [TK]D-Fender | sshock: Here's your issue : you aren't specifying the VM CONTEXT to look for the box in |
| 05:37.54 | [TK]D-Fender | sshock: AN issue anyway |
| 05:38.19 | [TK]D-Fender | sshock: Currently the key problem is the box isn't initialized either... this is after you leave your first VM. |
| 05:38.29 | sshock | the documentation said it would default to default |
| 05:38.40 | sshock | oh |
| 05:38.45 | [TK]D-Fender | sshock: Explicit > implicit |
| 05:38.48 | sshock | maybe I should have done leaving voicemails first :) |
| 05:38.50 | jkroon | [TK]D-Fender, ok, understood as such. |
| 05:39.20 | jkroon | thanks, as per usual. |
| 05:39.28 | sshock | ok, let me try that part... |
| 05:39.32 | [TK]D-Fender | jkroon: before you know it you're designing your entire dialplan around it and the first failure of the wrong piece and it all goes down the tubes |
| 05:39.52 | [TK]D-Fender | jkroon: this is whay you usually keep separate app servers, routers, etc |
| 05:41.24 | jkroon | yea, i've been wondering about this issue for a while now, and would love to keep separate registration/core processors but am having trouble motivating the extra hardware and hosting expenses to the $$$ controllers. so need to do the best I can with the resources I have. |
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| 05:43.12 | jkroon | anyway, thanks for an honest opinion on the matter. are there any proxies similar to ser but that can deal with IAX/2? |
| 05:43.36 | [TK]D-Fender | jkroon: http://www.despair.com/fail24x30pri.html |
| 05:43.48 | [TK]D-Fender | jkroon: No. IAX2 isn't a routed protocol |
| 05:44.01 | [TK]D-Fender | jkroon: Not even possible really |
| 05:46.03 | jkroon | lol, gee thanks :). ok, you consider SIP a routed protocol? would you mind explaining what the difference then is between a routed protocol (eg SIP?) and a non-routed protocol (eg IAX/2?) please? |
| 05:47.47 | [TK]D-Fender | jkroon: SIP has proxies, ther is a recorded router in the packets, as is the destination server, realm concepts, etc |
| 05:47.57 | sshock | now it's saying "No entry in voicemail config file for '101'", but I know I have an entry in there for it |
| 05:48.06 | [TK]D-Fender | jkroon: All evident if you've ever really looked at the packets before |
| 05:48.18 | [TK]D-Fender | sshock: Did you restart * since adding them? |
| 05:48.30 | [TK]D-Fender | sshock: a basic "reload" isn't enough |
| 05:48.38 | sshock | oh, that's why duh |
| 05:48.48 | [TK]D-Fender | sshock: you need "app_voicemail.so' to be reloaded |
| 05:49.03 | [TK]D-Fender | sshock: You could do it specifically |
| 05:49.05 | sshock | thanks |
| 05:49.17 | *** join/#asterisk oej (~olle@2001:470:1f15:d79:225:ff:fe44:74ec) |
| 05:49.51 | sshock | hey, it worked :) now to try and check voicemail... |
| 05:50.00 | sshock | yep, this is working too |
| 05:51.37 | sshock | next time I'll remember to restart * :) |
| 05:52.15 | [TK]D-Fender | sshock: this and chan_dahdi.so are the primary ones that require the module be reloaded entirely |
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| 05:57.02 | sshock | this voicemail menu is pretty cool |
| 06:00.18 | ritzt3ch | what does this mean |
| 06:00.23 | ritzt3ch | An alternate location is included in the header. This error is being generated by the SIP proxy server. |
| 06:00.56 | ritzt3ch | this is the ERROR sip response 301 moved permanently |
| 06:00.58 | ritzt3ch | ? |
| 06:01.10 | jkroon | [TK]D-Fender, have looked, just never thought about it that way. thanks for pointing it out. |
| 06:02.22 | adelas | damniit |
| 06:02.43 | adelas | the hardware canceller/card was why i get d/c problem |
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| 06:09.54 | coppice | but its common knowledge that a hardware EC solves all problems, including cancer |
| 06:10.34 | ChannelZ | cancer.. cancer.. cancer.. |
| 06:10.35 | [TK]D-Fender | checkout time, later all |
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| 08:06.21 | titter | So I just compiled Asterisk 1.6.2.13, didn't get any errors ... but the service isn't found ... hrmm anyone seen this before? ./configure -> make menuselect -> make -> make install ... no errors I can see, but no dice. |
| 08:06.43 | *** join/#asterisk aidinb (~Aidin@unaffiliated/aidinb) |
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| 08:18.36 | titter | Nevermind found the init.d in the contrib folder ... now the question is why didn't this do this automatically |
| 08:18.43 | ManxPower | titter, "make config" |
| 08:19.03 | ManxPower | as, I believe it suggests at the end of building |
| 08:19.26 | ManxPower | make install will NOT install the init scripts. That is what "make config" is for. |
| 08:19.44 | ManxPower | (make samples installs the sample configs, overwriting your existing configs, so you don't want to do that) |
| 08:19.51 | titter | Interesting, don't remember doing this in the past |
| 08:20.09 | ManxPower | then you forgot or used a package. Asterisk never installed init scripts by default. |
| 08:20.32 | titter | I must of forgot, I did it when I installed dahdi |
| 08:20.36 | titter | I blame the time |
| 08:20.37 | titter | thanks! |
| 08:20.40 | ManxPower | np |
| 08:20.52 | titter | Well at least I learned how to manually do it lol |
| 08:25.31 | titter | ManxPower: Interestingly enough it didn't suggest at the end of building, it suggest make samples and make progdocs, but not make config. dahdi does suggest make config |
| 08:26.02 | titter | thanks again though, I feel silly now |
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| 09:12.16 | marksaitis | hohoho, merry christmas kids :D |
| 09:15.58 | ChannelZ | where's my free stuff? |
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| 09:17.17 | marksaitis | you been a bad boy mother**** ;] |
| 09:20.17 | ectospasm | anyone have an idea why chan_iax.so would report a bad IPv4 address? I did something and the message went away. I used my configs from a 1.6.2.11 install in 1.8.0... |
| 09:21.41 | ChannelZ | Yes. Doing something usually fixes it. |
| 09:23.03 | *** join/#asterisk timahvo1 (~rogue@41.223.57.78) |
| 09:36.51 | ectospasm | ChannelZ: thanks. Only problem is I don't know exacly what I did, other than take out a bunch of comments (-; |
| 09:37.34 | ChannelZ | well you probably had a syntax error or something of some sort, but I can't go back in time and look |
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| 09:54.12 | ectospasm | ChannelZ: yeah... oh well, gone now! |
| 09:54.30 | ectospasm | I think it stemmed from having two [general] contexts in iax.conf |
| 09:55.21 | ectospasm | in other news, I called 911 again last night, testing my phone system |
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| 11:01.57 | TobSnyder | is it a bad idea to use asterisk outbound dial command options like t or w, as it might impact DTMF tone detection? |
| 11:02.23 | ofauchon | Hi... I'm looking for a web callback tool for Asterisk/AsteriskNow. Any idea ? thx |
| 11:02.55 | tzafrir | ofauchon, a simple one is trivial to write with the manager interface |
| 11:03.25 | tzafrir | What phones do you want it to call from? |
| 11:05.22 | TobSnyder | hi tzafrir, nice to see you on |
| 11:05.41 | tzafrir | TobSnyder, what's up? |
| 11:07.01 | ofauchon | tzafrir, I want to add a form on my website, where visitors can type theyr phone number. Then when there is a available operator in the company, asterisk will connect the visitor and the operator . |
| 11:07.30 | tzafrir | ofauchon, a web-based phone? |
| 11:08.05 | ofauchon | no... just a form "Enter your phone number, and an operator will call you back soon". |
| 11:08.19 | ofauchon | on the contact page of my website |
| 11:09.20 | *** join/#asterisk hehol (~hehol@buero-gw.dortmund.loca.net) |
| 11:09.56 | TobSnyder | ofauchon: you can use call files or Asterisk Manager interface and write a PHP script that makes a call using the mentioned options |
| 11:10.37 | ofauchon | I'll have a look to Asterisk Manager interface then, thanks |
| 11:11.21 | TobSnyder | call files http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out |
| 11:11.55 | TobSnyder | manager api: http://www.voip-info.org/wiki/view/Asterisk+Manager+API+Action+Originate |
| 11:12.02 | TobSnyder | or http://www.voip-info.org/wiki/view/Asterisk+manager+dialout |
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| 11:21.33 | *** join/#asterisk fleixius (~fleixius@unaffiliated/fleixius) |
| 11:21.55 | fleixius | Is chan_zap not included anymore within asterisk? |
| 11:25.02 | *** part/#asterisk mrfrenzy (~mrfrenzy@abydos.syntaxis.se) |
| 11:26.00 | petern_ | try dahdi |
| 11:26.08 | petern_ | it was renamed at some point, i think |
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| 11:27.33 | fleixius | Im using an older sangoma card which uses wanrouter and the support for dahdi was iffy. |
| 11:27.36 | fleixius | I'll try again |
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| 11:29.41 | ectospasm | Zaptel hasn't been used with Asterisk for over two years |
| 11:30.15 | fleixius | ectospasm: Are you talking <= 1.2 ? |
| 11:30.58 | ectospasm | No, sometime after 1.4 |
| 11:31.20 | ectospasm | actually, 1.4 didn't have support for DAHDI until 1.4.22 |
| 11:32.20 | ectospasm | fleixius: 1.4.22 came out over two years ago: 2 Oct 2008 |
| 11:32.46 | ectospasm | fleixius: if you're still using 1.2 it's soon to be EOL'd |
| 11:32.56 | ectospasm | http://asterisk.org/asterisk-versions |
| 11:33.25 | fleixius | ectospasm: Im waiting on sangoma to officially update their wanrouter drivers to support DAHDi for their usb fxo cards. |
| 11:33.40 | ectospasm | fleixius: I can't help you, I work for Digium (-; |
| 11:33.58 | ofauchon | TobSnyder, call files was really perfect ! The job is done ... php creating the Ascii file in the spool directory |
| 11:34.05 | ofauchon | Many thanks |
| 11:34.17 | fleixius | ;-) ha, my PR was for digium...apparently the 50$ less solution was MUCH better...yeah right |
| 11:34.29 | ectospasm | fleixius: you get what you pay for |
| 11:34.56 | ectospasm | fleixius: did you tell whoever that with Digium you get 5 years of top-notch installation support? |
| 11:34.58 | TobSnyder | ofauchon: no problem - and when you have created a nice script that does the job well for you - let the community know about so it might help some other people! |
| 11:38.08 | ofauchon | sure... Where's the best place to push the howto/script ? |
| 11:39.35 | fleixius | ectospasm: I did pleaded my case, but me being a 'NOVICE' ineexperienced PBX admin compared (meanwhile I've been plenty of a call manager solutions in my day) was trumphed by the nortel guy. |
| 11:39.57 | ectospasm | fleixius: heh, and where *IS* Nortel, nowadays? |
| 11:40.33 | TobSnyder | ofauchon: not sure, probably voip-info.org wiki? |
| 11:40.52 | ectospasm | ofauchon: that, or post to forums.digium.com |
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| 11:47.56 | ofauchon | Thanks guys ! I'll show my hard work to the boss :-) |
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| 11:54.57 | fleixius | Ahh conversion to dahdi went smoothly. |
| 11:55.13 | ectospasm | yeah, it's easier than it should be (-; |
| 11:55.18 | ectospasm | they did that on purpose (-; |
| 11:55.27 | fleixius | With wanpipe, nothing is easy. |
| 11:55.39 | ectospasm | I couldn't say, one way or the other. |
| 11:56.06 | ectospasm | All I can say is it's the driver for my employer's competitor |
| 11:56.14 | ectospasm | ...or one of them. |
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| 12:03.14 | fleixius | Hmm interesting problem now. When calling my conference line, and hanging up, the dahdi channel is still open. |
| 12:04.25 | ectospasm | fleixius: what kind of line is it? Analog or digital? |
| 12:04.39 | fleixius | ectospasm: analog. |
| 12:05.01 | ectospasm | sounds like a disconnect supervision problem. Where are you, what telco? |
| 12:05.15 | fleixius | verizon - us |
| 12:05.40 | ectospasm | so this was working before? |
| 12:06.01 | ectospasm | how is the channel defined, fxs_ks in chan_dahdi.conf? |
| 12:06.19 | fleixius | ectospasm: yep fxs_ks |
| 12:06.34 | ectospasm | ...and you recompiled Asterisk after you installed DAHDI? |
| 12:06.56 | fleixius | ectospasm: yep 1.6.2 |
| 12:08.28 | ectospasm | you can try putting busydetect and busycount in chan_dahdi.conf (above the channel lines), or maybe hanguponpolarityswitch=yes |
| 12:08.46 | ectospasm | ...but the latter shouldn't be necessary on US lines |
| 12:08.52 | ectospasm | I say *shouldn't* |
| 12:09.43 | ectospasm | you may want to test whether you've got disconnect supervision on the line, it's called many, many things by telcos the world over. |
| 12:09.46 | ectospasm | one moment |
| 12:09.47 | fleixius | Again the *nortel* ran these lines - for all I know they are routed through china. |
| 12:10.09 | jamko | Can I get an example of how to set an alternate parking lot for a dialplan context, for one step parking? |
| 12:12.37 | ectospasm | Disconnect supervision is also called "calling party control", "forward disconnect", "open loop disconnect","open switch interval", "adjunct control", and probably other names (AT&T apparently calls this Cutoff On Disconnect or COD, but that may not apply to all recombined Baby Bells). |
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| 12:14.18 | ectospasm | fleixius: you can test whether you have COD/disconnect supervision by connecting a POTS handset with a lighted keypad, and making a call. Let the far end hang up, and if the keypad blinks, you've got disconnect supervision |
| 12:14.38 | ectospasm | ...if it stays steady, you probably don't |
| 12:15.00 | fleixius | ectospasm: Ahh I could try that. |
| 12:15.07 | ectospasm | basically, chan_dahdi.so is looking for a battery drop from 6-7V to 0V for 250ms |
| 12:15.13 | ectospasm | hard to catch on a voltmeter |
| 12:15.32 | ectospasm | that may be an oversimplification |
| 12:15.41 | ectospasm | but you set that in chan_dahdi.conf |
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| 12:16.40 | fleixius | And if I do note have disconnect supervision ;X |
| 12:17.15 | ectospasm | fleixius: good luck trying to find someone at Verizon who knows what it is (-; |
| 12:19.19 | fleixius | I wonder if they have us configured for battery reversal... |
| 12:19.50 | fleixius | That would be covered by hanguponpolarityswitch...right? |
| 12:20.06 | ectospasm | fleixius: yep |
| 12:20.29 | ectospasm | you could always hook up a voltmeter and check the voltage as someone hangs up. |
| 12:20.54 | ectospasm | if the polarity reverses (for about 250ms, not long, but not too short to see, either) |
| 12:21.07 | ectospasm | ...then you know you've got to set hanguponpolarityswitch |
| 12:21.20 | ectospasm | it won't hurt anything |
| 12:21.41 | ectospasm | unless polarity reversals mean something else on Verizon's network, but that'd be silly |
| 12:21.59 | ectospasm | like if they signaled call waiting with it (-; |
| 12:24.35 | fleixius | Meh always a fun day when dealing with the telcos |
| 12:25.04 | jamko | Morning... Is it possible to use an alternate parking lot, with one step parking?? |
| 12:27.24 | ectospasm | jamko: that would be in features.conf if anywhere |
| 12:28.29 | jamko | ectospasm: yea it is, but it states to use the CHANNEL dialplan function, which I don't see how to use for a dtmf initiated parking. |
| 12:28.47 | ectospasm | jamko: what version of Asterisk? |
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| 12:29.34 | jamko | ectospasm: 1.6.2.12 |
| 12:30.16 | ectospasm | jamko: I don't have that installed. I'm using 1.8.0 |
| 12:30.28 | ectospasm | jamko: if you wait a moment, I can install it |
| 12:31.07 | jamko | ectospasm: if you are down, I am here all day. Thanks! |
| 12:31.08 | fleixius | ectospasm: thanks for the help - I'm gonna sit for hours calling waiting for verizon to answer. |
| 12:32.03 | ectospasm | jamko: shouldn't take too long |
| 12:32.05 | jamko | I am thinking I might be able to use func_odbc to make it work, but am hoping for something a little less involved for the current task. |
| 12:32.27 | ectospasm | jamko: shouldn't need that (unless you *want* to do it the database way) |
| 12:33.28 | jamko | ectospasm: not today.. |
| 12:34.08 | ectospasm | jamko: compiling now, let's see how long it takes on my VM! |
| 12:34.39 | jamko | ectospasm: nice.. |
| 12:35.35 | ectospasm | without looking, you can probably hook the DTMF code as an extension in the dialplan, then use the Park() application and pass the parking lot |
| 12:35.44 | ectospasm | ...but I'm not sure without looking |
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| 12:36.11 | srt | is there a way to report bugs for fax for asterisk? |
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| 12:37.44 | ectospasm | srt: FFA or FAX? |
| 12:37.52 | srt | FFA |
| 12:38.17 | ectospasm | srt: issues.asterisk.org, or pay for a 1-channel license of FAX and call Digium support |
| 12:38.40 | ectospasm | srt: disclaimer, I work for Digium support |
| 12:39.25 | srt | i am currently adding support for the ami actions/events to asterisk-java so there is no need for paid support ;) |
| 12:39.46 | srt | which category should i choose on issues.asterisk.org? |
| 12:39.59 | jamko | ectospasm: probably something involving applicationmap in features.conf |
| 12:40.55 | ectospasm | jamko: see "core show application Park" |
| 12:41.49 | ectospasm | srt: I dunno, I only deal with the paid one |
| 12:42.43 | srt | k then i'll report against ami |
| 12:43.04 | srt | btw: is there a free dev license for skype like the one for ffa? |
| 12:44.16 | ectospasm | jamko: that seems an awfully complicated way to do it. |
| 12:44.29 | ectospasm | jamko: why isn't setting the CHANNEL dialplan function not good enough? |
| 12:44.38 | ectospasm | s/isn't/is/? |
| 12:44.58 | ectospasm | srt: I don't believe so |
| 12:45.13 | ectospasm | srt: Skype keeps their stuff locked down tight. |
| 12:45.44 | srt | k |
| 12:50.13 | jamko | ectospasm: I agree, definitely complicated. How can I set the CHANNEL function with a one step dtmf initiated parking.. So for example, on a live call, I dial #72, which automatically parks the call in the default parking lot. How can I set the CHANNEL function on that #72? |
| 12:51.43 | ectospasm | jamko: exten => *2,1,Set(CHANNEL(blah)=foo) |
| 12:52.00 | ectospasm | jamko: exten => *2,2,Dial(...) |
| 12:52.12 | ectospasm | gotta be two steps, but the user doesn't need to know that happened |
| 12:52.46 | jamko | ectospasm: Is asterisk going to recognize *2 as an extension while on a live call? |
| 12:53.15 | ectospasm | jamko: ah, no, you didn't say that |
| 12:53.38 | ectospasm | You'd have to set the CHANNEL in dialplan before the call is bridged |
| 12:54.02 | ectospasm | ...then use the feature to jump to the parking lot defined by CHANNEL |
| 12:55.04 | jamko | ectospasm: So I would have to define CHANNEL on every dial plan action where I want to access that parking lot at some point in the call? |
| 12:55.56 | jamko | ectospasm: I guess every dialplan sequence would be the better term. |
| 12:56.11 | ectospasm | jamko: yep, or define the default parking lot to be that |
| 12:56.41 | jamko | ectospasm: I think maybe two step parking is the better route .. lol |
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| 13:03.16 | jamko | ectospasm: So now that I have decided on two step parking, I am seeing something strange. I make an outbound call, wait until it is bridged. Then I park the call, and it's parks successfully.. However after I park the call, asterisk is continuing on in the dialplan from the original outbound call, and dialing the original number again on my secondary provider in route. |
| 13:10.11 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 13:10.11 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 13:11.51 | jamko | How do I stop asterisk from continuing in the dialplan after parking a call? |
| 13:12.01 | jamko | It just keeps on trucking.. |
| 13:12.37 | leifmadsen | jamko: show the dialplan you're using and the console output (use pastebin) |
| 13:14.05 | jamko | ok |
| 13:24.35 | jamko | leifmadsen: http://pastebin.com/ZLTWAaeP |
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| 13:35.42 | jamko | leifmadsen: I should mention that it continues in the dialplan pretty much immediately after I park the call. So if I hang up right after parking, there is no issue, but if I take second or two to disconnect, asterisk will continue on where it left off from the original call. So if it were an outbound call that I parked, that same number would get called again after being parked, dialed... |
| 13:35.43 | jamko | ...through my secondary provider in route. |
| 13:36.16 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.216.2) |
| 13:54.50 | jamko | Also wondering where exactly I set the CHANNEL function for parking calls in alternate parking lots. It seems asterisk does not run the extensions dialed through the dialplan, when using a featuremap transfer such as #1. Do I need to set CHANNEL in the dialplan sequence prior to the call being bridged? |
| 13:57.13 | [TK]D-Fender | jamko: last I heard there was a variable to set to secify the lot series to use, not the CHANNEL() function. |
| 13:58.18 | *** join/#asterisk dandate2 (~gtejkgjke@112.198.78.84) |
| 13:58.33 | dandate2 | where can i trace the network of a DID? ie. level 3, mobile, landline |
| 14:00.09 | [TK]D-Fender | dandate2: As in knowing who carries that #? |
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| 14:01.48 | dandate2 | ye |
| 14:01.51 | jamko | Is there a way to get asterisk to read from the dialplan, when transferring with a featuremap, such as #1? Or do I need to use applicationmap to accomplish this? |
| 14:05.09 | [TK]D-Fender | jamRead what from the dialplan? that just immediately calls app_park. Where does dialplan even come in? |
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| 14:07.59 | jamko | TK: hmmm... #1 states it is a blindxfer option, and there is also an attended xfer option. I was kicking around the idea of transferring to extensions within the dialplan by using dtmf commands. |
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| 14:08.39 | jamko | So #1 --> call goes on hold, then dial another number, and drop the two together. |
| 14:09.00 | jamko | I was not aware that #1 just called app_park |
| 14:09.19 | [TK]D-Fender | jamko: what # would you transfer it to? |
| 14:10.29 | jamko | TK: any number that matched a pattern in the extensions.conf dialplan, if that was possible. The only way I have done that in the past was with applicationmap. |
| 14:12.38 | jamko | TK: I guess the confusion comes from the statement in features.conf "use the CHANNEL function to specify an alternate parking lot." That made me think there was a way to get asterisk to run the dialed extension after #1 back through the dialplan. Why else would it be mentioned to use CHANNEL, |
| 14:13.06 | jamko | unless you have to specify CHANNEL in every dialplan sequence that you are parking calls in.. Which seems very tedious. |
| 14:13.51 | [TK]D-Fender | jamko: any number? How do you pick the SPECIFIC numner? |
| 14:16.37 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 14:16.52 | jamko | TK: No not any number, the number dialed after #1 ... So I want to xfer to extension 2000, but I dont want to use the phone to do it... I would think I could do #1, dial 2000, asterisk finds 2000 in the dial plan, and drops the call to that extension |
| 14:17.39 | jamko | But apparently asterisk won't go back to the dialplan without an applicationmap telling it to do so. Unless I am missing somethign. |
| 14:18.05 | FiReSTaRT | hey guys, i'm sure you get asked this a lot so i won't get mad if someone points me to a link.. what's the difference between freepbx and *now.. from what i know, *now has some sort of a gui (is it gnome, kde, web based?) that also offers full flexibility on editing config files.. what about freepbx? |
| 14:18.44 | FiReSTaRT | new to *, familiar with linux, including the cli but liking the flexibility of also having a gui |
| 14:20.41 | jamko | TK: but really that is kind of off topic from what I am trying to do. What I really want is to setup alternate parking lots, and how to get it parked without specifying CHANNEL in every dialplan sequence. |
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| 14:22.16 | jamko | TK: and if it's feasible to do one step parking #72 to alternate parking lots. |
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| 14:24.35 | [TK]D-Fender | jamko: .... why not just do a DTMF transfer then? You are reinventing the wheel |
| 14:26.01 | [TK]D-Fender | FiReSTaRT: AsteriskNOW is a DISTRO based on CentOS which includes Asterisk, FreePBX, and a bunch of other stuff. FreePBX is just a GUI add-on. |
| 14:26.04 | jamko | TK: That's what I thought #1 did .. Maybe I am missing another feature for a DTMF transfer? |
| 14:26.24 | [TK]D-Fender | jamko: "core show application dial" |
| 14:26.24 | jamko | or # by default, whichever. |
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| 14:26.29 | [TK]D-Fender | jamYou missed the big print |
| 14:26.41 | jamko | TK: not unusual for me. |
| 14:27.26 | FiReSTaRT | [TK]D-Fender: so freepbx is a web management gui? and the asterisknow distro has gnome built in? |
| 14:28.56 | [TK]D-Fender | FiReSTaRT: No. Asterisk has no screen-GUI built in. FreePBX is a WEB GUI for configuring Asterisk |
| 14:29.11 | [TK]D-Fender | FiReSTaRT: And AsteriskNOW comes with FreePBX |
| 14:30.15 | FiReSTaRT | [TK]D-Fender: the reason i was asking is that asterisk - the future of telephony gave me a sense that they were 2 different things, but i could have misread it |
| 14:30.50 | [TK]D-Fender | FiReSTaRT: They are. FreePBX = wheel. AsteriskNOW = CAR. Car COMES with wheel |
| 14:30.51 | FiReSTaRT | [TK]D-Fender: but i could open any config file via freepbx' web interface? |
| 14:31.16 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 14:31.24 | [TK]D-Fender | FiReSTaRT: AsteriskNOW is an entire OPERATING SYSTEM. What if you want to use FreePBX on your OWN server with your OWN OS, etc? Then install FreePBX YOURSELF |
| 14:31.28 | [TK]D-Fender | FiReSTaRT: Clearer? |
| 14:32.58 | *** join/#asterisk moy_ (~moy@74.12.105.118) |
| 14:33.19 | FiReSTaRT | [TK]D-Fender: much clearer.. btw, it looks like i could run gnome on the *now distro, as it's centos based? while i wouldn't do it on a production system, it might make my learning process easier? |
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| 14:35.47 | [TK]D-Fender | FiReSTaRT: FreePBX isn't for "learning" |
| 14:36.17 | [TK]D-Fender | FiReSTaRT: It isn't a proper stepping stone to learning *. It does everything for you with a metric ton of scripts and complicated dialplan |
| 14:37.07 | FiReSTaRT | [TK]D-Fender: so what would you recommend for learning? running *now, installing gnome and no freepbx? |
| 14:37.20 | [TK]D-Fender | FiReSTaRT: well what are you actual goals? |
| 14:38.28 | FiReSTaRT | [TK]D-Fender: mostly to get the ins and outs of asterisk.. eventually i'd like to deploy it with a web interface so less tech-savvy people can still do some minor admin, but i don't want everything on a silver platter |
| 14:39.19 | FiReSTaRT | for some reason i thought that the asteriskgui and freepbx gui were different things.. at least that's how it came off in the book |
| 14:39.28 | [TK]D-Fender | FiReSTaRT: FYI that so far there isn't a really viable in-between yet |
| 14:40.08 | [TK]D-Fender | FiReSTaRT: What you might be hoping for would be a GUI to do some of the basics but still leave you real control. There isn't a released on that does this yet |
| 14:40.18 | [TK]D-Fender | FiReSTaRT: Do you have a *NIX server already? |
| 14:40.55 | FiReSTaRT | [TK]D-Fender: yeah.. stripped down lucid server |
| 14:41.09 | [TK]D-Fender | FiReSTaRT: You could simply install * on that and get started. |
| 14:41.32 | [TK]D-Fender | Learn the basics of * from scratch so you understand how the bits work and can make up your mind from there |
| 14:41.54 | FiReSTaRT | [TK]D-Fender: i might do that or reinstall it with xfce and then install asterisk |
| 14:42.04 | jamko | Anyway, I'm still apparently missing the big print, but will get back to that later. Asterisk is continuing in the dialplan after I park a call, and I'm not sure how to stop it. Here is a pastebin of my dialplan and cli output: http://pastebin.com/jV5qzNg0 |
| 14:42.17 | FiReSTaRT | b/c right now it's without any gui whatsoever |
| 14:43.28 | FiReSTaRT | [TK]D-Fender: and i wanted to follow my openwrt learning curve.. started by only doing config in xwrt, then just using its file editor... now it's just ssh into the console to edit and/or recover my config files |
| 14:44.05 | [TK]D-Fender | FiReSTaRT: Ok, well XFCE has nothing to do with *.... |
| 14:44.40 | [TK]D-Fender | FiReSTaRT: You should avoid talking about actual screen DE's in reference to your Asterisk intentions |
| 14:45.14 | [TK]D-Fender | FiReSTaRT: FreePBX = WEBS scripts. Stuff you run under Apache. Not a Desktop Envirnment |
| 14:46.45 | [TK]D-Fender | FiReSTaRT: Just making sure you aren't mixing stuff up |
| 14:47.50 | FiReSTaRT | [TK]D-Fender: xfce would give me some basic tools like a file manager and a gui text editor.. but i was hoping i could also run something LIKE freepbx |
| 14:48.29 | FiReSTaRT | [TK]D-Fender: so there is no such thing as an "asteriskgui" web management setup? |
| 14:48.33 | [TK]D-Fender | FiReSTaRT: You only have 1 computer? |
| 14:49.03 | [TK]D-Fender | [10:48]<FiReSTaRT>[TK]D-Fender: so there is no such thing as an "asteriskgui" web management setup? <- AsteriskGUI is ANOTHER web-interface for configuring * |
| 14:49.19 | [TK]D-Fender | FiReSTaRT: You are beginning to mix things up again |
| 14:49.41 | FiReSTaRT | [TK]D-Fender: actually i'm trying to disambiguate between freepbx and asteriskgui |
| 14:50.03 | FiReSTaRT | [TK]D-Fender: i only have 1 computer i can install and run * on right now and that's only under a virtual environment for now |
| 14:50.21 | [TK]D-Fender | FiReSTaRT: Both Web scripts to configure *. Which means usually you'll view them from ANOTHER BOX. Not using the screen on the SERVER itself |
| 14:50.28 | [TK]D-Fender | FiReSTaRT: Follwing me here? |
| 14:50.37 | [TK]D-Fender | FiReSTaRT: WEB BROWSER. |
| 14:50.49 | FiReSTaRT | [TK]D-Fender: loud and clear.. but that's something i'd like to be able to do as well |
| 14:51.04 | [TK]D-Fender | FiReSTaRT: You don't needs GNOME/KDE/XFCE/E11/etc on your SERVER for any of this so far |
| 14:51.20 | [TK]D-Fender | FiReSTaRT: Now if you feel like it sure... but that has nothing to do with * itself |
| 14:51.40 | FiReSTaRT | [TK]D-Fender: i know i don't need a gui on the server itself and that it carries heavy performance penalties, but i'd still like to, at least while im learning :) |
| 14:52.21 | [TK]D-Fender | FiReSTaRT: AsteriskGUI = WEB GUI (unmaintained, incomplete, noticably buggy). FreePBX = WEB GUI (very complete, active development, completely owns your ass... mostly) |
| 14:53.14 | [TK]D-Fender | FiReSTaRT: Gnome won't have any impact on learning *. And it only lets you run a graphical broswer should you actually go about installing FreePBX to run the show. |
| 14:53.47 | [TK]D-Fender | FiReSTaRT: Ever use Webmin before? CPanel? |
| 14:54.13 | FiReSTaRT | [TK]D-Fender: ok, now lets say i install freepbx at some point later on down the line.. i can just swap the dialplans and other config files with the ones i created? |
| 14:54.42 | [TK]D-Fender | FiReSTaRT: Not at all |
| 14:54.45 | FiReSTaRT | [TK]D-Fender: never used either |
| 14:54.59 | [TK]D-Fender | FiReSTaRT: FreePBX is meant to run the show. You'd be starting from scratch with it |
| 14:55.32 | [TK]D-Fender | FiReSTaRT: You add little extras in merged-in configs, but none of your previous structures will apply |
| 14:55.56 | FiReSTaRT | [TK]D-Fender: in that case i'll probably just do without it and see how bad the asteriskgui bugs are |
| 14:57.21 | Orbixx | The GUIs limit your ability to customise Asterisk somewhat. |
| 14:57.44 | Orbixx | They're restrictive, but that's the intent. |
| 14:57.48 | [TK]D-Fender | FiReSTaRT: Also be aware that AsteriskGUI was made for * 1.4. * 1.8 has just been released. AsteriskGUI does not have a maintainer and also uses one specific config files that changes numerous was you interact with * outside of the "regular" configs. There will be a price to pay when that comes |
| 14:58.10 | Orbixx | Anyway, somebody help me with an issue I'm having with Asterisk. I'm not sure if it's one for the bugtracker. |
| 14:58.27 | FiReSTaRT | Orbixx: i figured that even though they don't have to.. xwrt in openwrt gives you access to the whole file system.. that allows you to manually edit any config file |
| 14:58.36 | Orbixx | Trying to place a call over SIP, but the 401 I receive from the trunk doesn't get responded to by my Asterisk installation. |
| 14:59.16 | Orbixx | FiReSTaRT: Which you could do over shell with any Asterisk installation, but don't expect any GUI to be able to recognise some of your custom changes. |
| 14:59.26 | FiReSTaRT | [TK]D-Fender: aww crap.. so no happy medium :( |
| 14:59.45 | [TK]D-Fender | FiReSTaRT: AsteriskGUI is a medium... just not so happy right now. |
| 14:59.53 | [TK]D-Fender | FiReSTaRT: It has fallen to neglect. |
| 15:00.18 | [TK]D-Fender | FiReSTaRT: It does have 1 or 2 active users in-channel ;) |
| 15:00.34 | [TK]D-Fender | FiReSTaRT: But the Op/User ration is about 5:1 |
| 15:00.35 | FiReSTaRT | [TK]D-Fender: pretty bad neglect, by the looks of it.. a new version came out and the beta for the next version is out there as well |
| 15:01.08 | FiReSTaRT | so it looks like it's ssh, vi, paste, save, gtfo |
| 15:01.24 | [TK]D-Fender | FiReSTaRT: It wouldn't be a "terrible" way to get into *, but its use of users.conf will become quickly limiting |
| 15:01.37 | Orbixx | FiReSTaRT: Familiarise yourself with `scp`. |
| 15:01.39 | [TK]D-Fender | (could) |
| 15:01.47 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 15:02.16 | [TK]D-Fender | FiReSTaRT: Just be aware that NO GUI is supported in this channel. They each have their own. |
| 15:02.17 | FiReSTaRT | Orbixx: i could do that, but i like to do a final review in addition to some last minute tweaks :P |
| 15:02.46 | FiReSTaRT | [TK]D-Fender: i know and i wouldn't be asking any gui-specific questions.. just trying to get the lay of the land before i decide on my implementation |
| 15:02.57 | Orbixx | [TK]D-Fender: Don't suppose you could assist with my issue? I'm at a complete loss. |
| 15:03.20 | [TK]D-Fender | Orbixx: Don't suppose you could actually SHOW US your issue? We're not PSYCHIC |
| 15:03.24 | [TK]D-Fender | :p |
| 15:03.39 | [TK]D-Fender | FiReSTaRT: Ok, hopefully you've got that now |
| 15:03.57 | Orbixx | Right, I'll grab anything pertinent and pastebin it. |
| 15:04.38 | FiReSTaRT | [TK]D-Fender: basically one gui is worse than windows, the other is unmaintained and if i wanted to set something up that would allow a typical receptionist to at least add a user, i'd have to set it up myself :) |
| 15:05.37 | *** join/#asterisk Chodorenko (~chodorenk@by.one.by) |
| 15:06.05 | [TK]D-Fender | FiReSTaRT: Well this does depend on if you'd be happy with FreePBX for the core needs and if it has enough room for the interesting stuff you might want to fit into that installation. |
| 15:06.24 | Orbixx | [TK]D-Fender: http://pastebin.com/FFwmxrCw sip.conf |
| 15:06.24 | [TK]D-Fender | FiReSTaRT: Doesn't stop you from learning * on your own side and running FreePBX for a business production. |
| 15:06.34 | [TK]D-Fender | FiReSTaRT: Just depends how the mix sits with you and your goals. |
| 15:06.45 | [TK]D-Fender | FiReSTaRT: Real pain comes when you change your mind mid-swing |
| 15:07.07 | jamko | [TK]D-Fender: http://pastebin.com/jV5qzNg0 .... After doing a blind DTMF xfer, asterisk is continuing in the dialplan from the original call. In the pastebin, it's doing it after I park the call, but it does it with DTMF xfer as well. |
| 15:08.27 | FiReSTaRT | [TK]D-Fender: so i can play with * first until i learn the basics, then i learn more about freepbx and work from there... nothing preventing me from havin a couple of virtual machines running (other than hard drive space) :P |
| 15:08.31 | [TK]D-Fender | jamko:Dial(SIP/1${EXTEN}@gafachi_rochester) [pbx_config] ------------------- -- Executing [5555555555@jamko_phones:2] Dial("SIP/9000-000000a4", "SIP/5555555555@gafachi_rochester,120,T") in new stack |
| 15:08.32 | *** part/#asterisk Chodorenko (~chodorenk@by.one.by) |
| 15:08.40 | *** join/#asterisk Chodorenko (~chodorenk@by.one.by) |
| 15:08.40 | [TK]D-Fender | jamko: You are clearly showing me 2 things that don't match |
| 15:09.49 | [TK]D-Fender | FiReSTaRT: Of course not. Everything depends on your goals, resources and approach |
| 15:10.09 | [TK]D-Fender | FiReSTaRT: What are your business needs with *? |
| 15:10.57 | FiReSTaRT | [TK]D-Fender: currently none.. eventually.. i'd like to offer a pbx solutions service to local small-medium sized businesses |
| 15:11.01 | Orbixx | http://asterisk-home.xen.exoware.net:8000/dump |
| 15:11.21 | Orbixx | [TK]D-Fender: tcpdump of what is happening ^ |
| 15:11.30 | jamko | [TK]D-Fender: It prepends the 1 in line 14.. |
| 15:11.37 | [TK]D-Fender | FiReSTaRT: Hosted? as-deployed on-site with a box? |
| 15:11.43 | FiReSTaRT | [TK]D-Fender: our current business needs are well-served by 2 routers and an ata box, but our voip provider has a pretty decent dashboard |
| 15:11.52 | FiReSTaRT | [TK]D-Fender: most likely on site |
| 15:12.27 | [TK]D-Fender | jamko: The dialplan line does NOT have the "t" option in it and the CLI output shows it DOES. You are showing me 2 DIFFERENT things |
| 15:12.50 | jamko | TK: damn it.. caught.. hold on.. |
| 15:12.53 | [TK]D-Fender | FiReSTaRT: A client would require something rather "complete", which FreePBX is far better suited for |
| 15:13.10 | *** join/#asterisk TimeRider (~steve@188-220-33-123.zone11.bethere.co.uk) |
| 15:13.32 | [TK]D-Fender | Orbixx: WTF is with that BIN file? FFS go get ASTERISK SIP DEBUG |
| 15:13.34 | FiReSTaRT | [TK]D-Fender: but i still like the idea of learning * as it is first so i know what freepx is doing in the first place |
| 15:13.54 | Orbixx | [TK]D-Fender: It's a tcpdump. |
| 15:13.57 | [TK]D-Fender | FiReSTaRT: By all means... jsut giving a fiar assessment of what your business direction would point towards. |
| 15:14.09 | Orbixx | Drop it in Wireshark if you'd like to see what happens during my SIP outgoing call attempt. |
| 15:14.46 | FiReSTaRT | [TK]D-Fender: much appreciated, btw... did a whole lot of googling, but it can't beat shootin' the **** with someone who already knows the lay of the land |
| 15:14.54 | [TK]D-Fender | FiReSTaRT: Clients are DUMB. The want things they can most normal things by a few clicks. AsteriskGUI isn't it. If its for YOUR companies use and you want to hand minial control to a receptionist to just adda user then fine, but it still takes an admin at some larger level |
| 15:15.05 | [TK]D-Fender | FiReSTaRT: FreePBX really covers just about everything |
| 15:15.30 | [TK]D-Fender | can do* |
| 15:15.39 | [TK]D-Fender | gah... skipping entire words now... |
| 15:16.11 | Orbixx | [TK]D-Fender: http://pastebin.com/Mr1xE3rW - as requested |
| 15:17.08 | jamko | [TK]D-Fender: http://pastebin.com/ibPhSK0j |
| 15:17.21 | FiReSTaRT | [TK]D-Fender: in general, i'd prefer a model where a secretary or an "it guy" (someone who knows how to install windows) would do the minimal stuff, but i'd always be available for (paid) support calls... freepbx might give'em too much control (ways to screw things up) |
| 15:19.11 | [TK]D-Fender | FiReSTaRT: Then I know of no product that is a suitable in-between. |
| 15:19.31 | [TK]D-Fender | FiReSTaRT: Doesn't stop you from developing your own of course |
| 15:20.21 | [TK]D-Fender | Orbixx: They really don't like your auth I guess... check users/pass/account activation, etc |
| 15:20.27 | [TK]D-Fender | Orbixx: An call them up |
| 15:20.28 | *** join/#asterisk [sr] (~kvirc@pal-213-228-140-150.netvisao.pt) |
| 15:20.32 | [sr] | hi guys |
| 15:20.48 | FiReSTaRT | [TK]D-Fender: i might develop my own that says "hello world" but that's where my coding knowledge basically ends... so i'll most likely try both and see which one can be ghettoed to an acceptable level of suitability |
| 15:20.58 | Orbixx | [TK]D-Fender: I've already been through this with them; auth is correct; account is active. |
| 15:21.08 | Orbixx | [TK]D-Fender: The problem is... It's not authing full stop. |
| 15:21.19 | FiReSTaRT | [TK]D-Fender: actually, giving them ways to screw things up would not be so bad for my bottom line... more support hours billed ;) |
| 15:21.19 | Orbixx | They send a 401 and Asterisk doesn't respond with auth. |
| 15:26.46 | [TK]D-Fender | Orbixx: Not all 401 is a challenge. some are a GTFO |
| 15:27.15 | [TK]D-Fender | Orbixx: Are you successfuly registered to them? |
| 15:27.50 | FiReSTaRT | [TK]D-Fender: in any case thanks for all the help.. i gotta run for a few mins now and install *now when i get back :) |
| 15:27.51 | *** join/#asterisk wizhippo (~Adium@64.229.234.139) |
| 15:29.23 | Orbixx | [TK]D-Fender: Yes, I can receive calls. |
| 15:29.36 | Orbixx | Well, that's not 'successfully registered, I suppose'. |
| 15:29.41 | Orbixx | One mo' |
| 15:30.17 | *** join/#asterisk Sipster (~Sipster@modemcable045.5-200-24.mc.videotron.ca) |
| 15:30.23 | Orbixx | [TK]D-Fender: http://pastebin.com/weT8hHpX |
| 15:30.59 | wizhippo | Hi guys. I have two interfaces on my box. eth0 - internet eth1 - locanet. When a transfer is made from a device on the localnet to one on the internet the audio is dropped. Do I just use externip and locelnet setting in sip.comf to fix this? I thought that was used if you have one interface going via nat. |
| 15:31.41 | [TK]D-Fender | Orbixx: No. sip show registry |
| 15:31.56 | Orbixx | [TK]D-Fender: It's not registered. |
| 15:32.12 | [TK]D-Fender | Orbixx: Leadas me to believe all your auth is wrong |
| 15:32.16 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 15:32.24 | [TK]D-Fender | wizhippo: "canreinvite=no" <- essential |
| 15:33.21 | Orbixx | [TK]D-Fender: I can see why you'd think that, but the guys who supply my trunk dumped the traffic on their server on their end the moment I was supposed to be authing and they received nothing containing auth. |
| 15:34.01 | wizhippo | [TK]D-Fender, what about when the calls are transfer local to local? they should be able to reinvite |
| 15:34.48 | Orbixx | [TK]D-Fender: Okay, I've managed to get it registered, but it still fails on outbound calls. |
| 15:35.37 | *** part/#asterisk Chodorenko (~chodorenk@by.one.by) |
| 15:35.47 | *** join/#asterisk Chodorenko (~chodorenk@by.one.by) |
| 15:36.47 | Orbixx | [TK]D-Fender: http://pastebin.com/WXBvthLv |
| 15:36.59 | *** join/#asterisk Trixboxer (~Trixboxer@office.supportdepartment.net) |
| 15:37.29 | [TK]D-Fender | wizhippo: You need to isolate 1 side from the other |
| 15:39.15 | wizhippo | So I should just use nat and remove the internet interface. K |
| 15:39.38 | *** join/#asterisk WWGD (~WWGD@208.79.14.130) |
| 15:41.31 | [TK]D-Fender | wizhippo: This shouldn't be a load issue in a local LAN. just prevent reinvites |
| 15:41.50 | [TK]D-Fender | wiAt least on the OUTSIDE entries and in general, but specif it can for the internals |
| 15:42.53 | FiReSTaRT | [TK]D-Fender: just fyi.. *now offers the option of installing 1.6 with *gui |
| 15:43.08 | [TK]D-Fender | FiReSTaRT: I know. |
| 15:43.39 | [TK]D-Fender | FiReSTaRT: But who chooses that :p |
| 15:43.47 | *** join/#asterisk lirakis (~lirakis@ool-ad022bb1.dyn.optonline.net) |
| 15:44.06 | FiReSTaRT | [TK]D-Fender: i give you 3 guesses to figure out at least 1 person on this channel :D |
| 15:48.20 | *** join/#asterisk WWGD (~WWGD@208.79.14.130) |
| 15:49.57 | *** join/#asterisk applync (~applync@mail.applync.com) |
| 15:51.31 | FiReSTaRT | [TK]D-Fender: it looks like *now is set to run as / by default.. good for learning, bad for production :P |
| 15:55.20 | *** join/#asterisk Tim_Toady (~moi@79.103.14.105.dsl.dyn.forthnet.gr) |
| 15:55.21 | *** part/#asterisk TobSnyder (~schneider@dslb-088-073-204-184.pools.arcor-ip.net) |
| 15:57.00 | [TK]D-Fender | FiReSTaRT: ? |
| 15:57.33 | FiReSTaRT | [TK]D-Fender: the stock config for *now has it running from a root account |
| 15:58.36 | [TK]D-Fender | FiReSTaRT: Doubt that... |
| 15:58.50 | [TK]D-Fender | FiReSTaRT: Check the owners in /etc/asterisk |
| 15:59.12 | *** join/#asterisk ybit (~quassel@unaffiliated/ybit) |
| 16:02.10 | FiReSTaRT | [TK]D-Fender: ahh correct.. the install process just created a root password though and that had me (an almost 100% ubuntu user) a bit confused :) |
| 16:02.47 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 16:03.49 | [TK]D-Fender | FiReSTaRT: then again your ubuntu user has sudo rights to do jsut about everything |
| 16:04.00 | [TK]D-Fender | FiReSTaRT: So it's barely "secure" |
| 16:04.17 | [TK]D-Fender | by comparison |
| 16:04.57 | *** join/#asterisk nicoAMG (~nicoAMG@201.237.49.131) |
| 16:05.32 | FiReSTaRT | [TK]D-Fender: but the user has to take the admin rights and enter the password, so it's the right balance of security and convenience.. and that only applies to one admin account (unless you set it up differently) |
| 16:06.22 | [TK]D-Fender | FiReSTaRT: yeah, its still pretty good, but the door is wide very fast.... I use it as my desktop going on 2 years now so it's plenty fine |
| 16:07.31 | FiReSTaRT | [TK]D-Fender: agreed... that's why i make sure to give the speech to new users.. however, it's an excellent desktop system, so i'll keep it for my personal needs |
| 16:09.32 | wizhippo | [TK]D-Fender, thanks for the pointers but I found the issue. It was actually the client on the internet that was behind a nat messing it all up. set the stun server on it and all is well. thanks |
| 16:10.41 | *** join/#asterisk coppice (~chatzilla@112.203.17.210.dyn.pacific.net.hk) |
| 16:18.20 | Chodorenko | Hi All, Could some one suggest any software SIP client with support for SRTP for asterisk? |
| 16:18.27 | *** join/#asterisk neurosys (~neurosys@173-9-159-177-miami.txt.hfc.comcastbusiness.net) |
| 16:19.20 | Chodorenko | Twinkle do not report of support srtp on SDP , Eyebeam/X-lite not support SRTP which out of TLS .... |
| 16:19.47 | *** part/#asterisk WWGD (~WWGD@208.79.14.130) |
| 16:20.38 | Tim_Toady | Chodorenko check sflphone, but im not sure it works, never test it with * |
| 16:21.13 | *** join/#asterisk b_d (~brian@198.207.20.191) |
| 16:22.05 | b_d | I am trying to use the "Bridge" command / application with the Originate in PHPAGI, but I need to pass the original AGI channel variable to complete the bridge |
| 16:22.22 | b_d | what is the variable name? how do I access it? I thought may $channel |
| 16:22.25 | b_d | maybe * |
| 16:25.53 | ManxPower | get the channel using the AGI get variable function, then it will be a PHP variable |
| 16:28.40 | *** join/#asterisk Docteh (~mage@24.86.236.78) |
| 16:29.00 | b_d | $agi->get_variable('channel'); ? |
| 16:29.27 | jamko | [TK]D-Fender: http://pastebin.com/ibPhSK0j .... Asterisk continues on in the dailplan after doing a blind xfer, or parking a call... thanks.. |
| 16:33.14 | *** join/#asterisk titter (~Justin@c-98-208-152-139.hsd1.fl.comcast.net) |
| 16:33.36 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 16:41.04 | [TK]D-Fender | jamko: Odd. |
| 16:44.10 | *** join/#asterisk g_r_eek (~g_r_eek@173-9-142-122-Miami.FL.hfc.comcastbusiness.net) |
| 16:45.57 | Chodorenko | Tim_Toady: Yes its client is true work |
| 16:46.07 | Tim_Toady | :) |
| 16:46.26 | Chodorenko | Tim_Toady: Thx , may be client for Windows present ? |
| 16:46.41 | b_d | ? |
| 16:46.59 | Tim_Toady | i dont know, sfl seems to be unix only Chodorenko |
| 16:47.39 | leifmadsen | client for what? |
| 16:47.56 | Chodorenko | 19:18ChodorenkoHi All, Could some one suggest any software SIP client with support for SRTP for asterisk? |
| 16:48.13 | leifmadsen | zoiper |
| 16:48.26 | leifmadsen | pretty sure that supports SRTP |
| 16:49.14 | Tim_Toady | it does, but not in the free version |
| 16:49.15 | leifmadsen | and I think x-lite does too |
| 16:49.18 | Chodorenko | Zoiper only not free version |
| 16:49.22 | leifmadsen | k |
| 16:49.31 | leifmadsen | I'm sure google could answer this question :) |
| 16:49.33 | Chodorenko | 19:19ChodorenkoTwinkle do not report of support srtp on SDP , Eyebeam/X-lite not support SRTP which out of TLS .... |
| 16:49.43 | leifmadsen | shrugs |
| 16:51.47 | *** join/#asterisk timahvo1 (~rogue@41.223.57.75) |
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| 16:53.52 | Chodorenko | leifmadsen: i test more clients, from all only sflphone support SRTP in free version , correct and free |
| 16:54.01 | leifmadsen | well there ya go :) |
| 16:54.11 | leifmadsen | might turn out you have to pay some money |
| 16:56.11 | Chodorenko | leifmadsen i Use Hardware Grandstream pone :) , my Users software clients , i can bye for all corporate users Licenses ? no its no true path, if present freeware phone clients |
| 16:56.17 | coppice | SRTP isn't really the kind of thing which would cost money |
| 16:56.58 | leifmadsen | only the client which supports it |
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| 17:01.36 | Chodorenko | I think sending a clear voice through the Internet, it naruschenie any canons of safety. In fact, very sorry that so few people think about it. |
| 17:02.21 | Chodorenko | * naruschenie -> violation |
| 17:02.27 | leifmadsen | o.O |
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| 17:05.28 | Chodorenko | leifmadsen: decoding voice in transite router and analyze header of SIP not can hard processor job or very large knowledge |
| 17:06.47 | Chodorenko | leifmadsen: SRTP for All clients it requires at least now moment |
| 17:07.50 | Chodorenko | between servers tunel or TLS , for client better SRTP |
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| 17:32.27 | Docteh | huh i cant find details on how to put auth information into a dial command something like SIP/user:pass@itsp.server.com/18885551212 is that the right syntax? |
| 17:36.58 | [TK]D-Fender | Docteh: You should be using a PEER entry, not putting it all on the dial line |
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| 17:42.43 | b_d | <PROTECTED> |
| 17:44.35 | leifmadsen | Docteh: yep |
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| 17:46.42 | Docteh | kk, thx |
| 18:10.41 | srt | ectospasm: do you remember that you just told me to open an issue for ffa at issues.asterisk.org? the three issues just got closed there as they don't feel responsible. |
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| 18:44.20 | jamko | leifmadsen: http://pastebin.com/ibPhSK0j ... any idea why asterisk would continue through the dialplan after I do a DTMF transfer, or park a call? |
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| 20:05.02 | jamko | [TK]D-Fender: This seems to only be happening on boxes using static realtime dialplan from mysql db. Could that have something to do with it? It is happening from two different 1.6 boxes. However a 1.4 box, not connected to the DB, does not have the problem. |
| 20:06.05 | seanjohn | jamko, i wish I could say that for my 1.4 |
| 20:07.21 | seanjohn | asterisk crashed last night 1.4.36 and I can't find anything in the debug NOR the full log. I just says the last command executed on both, before asterisk started up again, was Hangup() |
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| 20:34.09 | jamko | [TK]D-Fender: Well it's not the database ... Must be something with 1.6 |
| 20:48.00 | applync | anyone get 1.8 RTM running? |
| 20:48.03 | applync | How is it? |
| 20:49.19 | pabelanger | applync: been running it for a few months. |
| 20:49.31 | pabelanger | applync: there are problems with chan_iax2 ATM |
| 20:55.49 | applync | same install method as 1.6? have you tried faxing with it yet? |
| 20:55.57 | applync | I read somewhere it is bult in... |
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| 21:05.34 | jamko | leifmadsen: It looks like my issue with parking was reported, and supposedly fixed in an 1.6.2.0, but I still have the issue in 1.6.2.12: "#14555: When i park a call after the slot announcement the call is not hangup - Revision: 192874" |
| 21:10.25 | *** join/#asterisk sshock (~sshock@63.248.133.83) |
| 21:10.59 | sshock | voip provider recommendations? |
| 21:12.02 | leifmadsen | bandwidth.com (USA) and Unlimitel (Canada) |
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| 21:14.27 | applync | I have been using vitelity for the past 4 months. They have been flawless... |
| 21:15.09 | applync | you can provision DID;'s in seconds, and if you go through the pbxinaflash site nerdvittles.com, you can get one with unlimited inbound minutes for $3.99 a month. |
| 21:15.50 | applync | you pay one time $1 to pickup a did, and $1 per month to keep it. then I am paying 1.2cents a minute for calls. |
| 21:22.01 | sshock | nice |
| 21:22.01 | sshock | that sounds even better than flowroute that I heard about the other day |
| 21:22.05 | sshock | will they let you put whatever you want in the Caller ID? |
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| 21:24.29 | sshock | applync: is 1.2c/min for outbound calls? what about inbound? is that also 1.2c/min or is it part of the $1/month for the DID? |
| 21:27.43 | sshock | oh, the 3.99 is the inbound; sorry, missed that |
| 21:28.36 | sshock | leifmadsen: bandwidth.com is expensive, no? |
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| 21:33.10 | leifmadsen | sshock: you get what you pay for -- you wanted a recommendation, and I gave it :) I run a couple of call centres on it, and they have practically zero down time |
| 21:33.17 | leifmadsen | I don't remember the last time I had an outage |
| 21:33.41 | sshock | ok |
| 21:33.58 | sshock | I should have clarified this is just for home use |
| 21:34.00 | leifmadsen | depends what your requirements are |
| 21:34.13 | leifmadsen | then you can pretty much use whoever you want :) |
| 21:39.59 | applync | bandwidth is nce, but it comes at a price. I also like broadvox a lot as well. |
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| 21:47.34 | jamko | Is there an option in the dial command , or some other way to tell asterisk that once the call is bridged, to stop processing the dial sequence? |
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| 21:52.36 | ruben23 | hi guys |
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| 21:58.03 | ManxPower | jamko, Your question makes no sense. |
| 21:58.28 | ManxPower | do you mean PAUSE the dialing of digits while you wait for the far end to answer? |
| 21:59.07 | ManxPower | Then send more digits once the far end answers? |
| 21:59.42 | jamko | ManxPower: It makes no sense because something is F'd with DTMF xfer, and call parking... Been pulling my hair out for 10 hours. Let me get the pastebin up, and explain the issue, and then you will understand why I am asking such a ridiculous question. |
| 22:01.07 | jamko | ManxPower: http://pastebin.com/ibPhSK0j ... The problem is, after I park a call, or execute a DTMF xfer, asterisk continues the dialplan sequence from the original call.... |
| 22:01.50 | ManxPower | I'm sorry, but I do not have time for an extended debugging session. If you want to send digits to the far end after it answers and are NOT using fxo ports, then dial your number and use the D() option to dial after the far end answers. You can put "w" in for a .5 second pause. D(wwww555#) in your dialoptions |
| 22:02.06 | titter | Can anyone point me in the direction for a single server fail over documentation located on a different subnet? Should I start to think about real time? |
| 22:02.39 | ManxPower | JamesHarrison, do you have exten => _. anywhere in your dialplan |
| 22:03.08 | jamko | yes |
| 22:03.08 | ManxPower | sorry. |
| 22:03.08 | JamesHarrison | 's cool. |
| 22:03.18 | ManxPower | jamko, Do you have exten => _. anywhere in your dialplan? |
| 22:03.48 | jamko | manxpower: I have _N .. but not _. |
| 22:03.56 | ManxPower | jamko, good. |
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| 22:05.58 | jamko | Thanks... It just keeps on trucking through the dialplan as if the call was never answered. |
| 22:06.12 | ManxPower | jamko, It is usually a bad idea to string multiple Dials in your dialplan like you are doing. Are you trying to dial all of them at the same time, or dial them in sequence, or dial the next destination only if the first destination does not answer? |
| 22:06.44 | ManxPower | What do you WANT to happen? |
| 22:06.57 | jamko | ManxPower: I am trying to make it so if the first provider in line bounces the call, asterisk will dial out on the second in line, and so on. |
| 22:07.16 | Ariel_Calzada | hi we're integrating with siemens hipath 3550 the problem is that the number called is not coming to my asterisk complete. Example: if somebody calls _9XXX i only receive 9 in my asterisk any idea why?? |
| 22:07.36 | jamko | ManxPower: which works in every situation so far except for dtmf xfer and parking in 1.6 .. however 1.4 does not have the issue. |
| 22:07.44 | ManxPower | jamko, multiple dial commands like that is not going to do that. After each Dial() find out what happened with the call by checking the value of the DIALSTATUS variable, then decide what to do in your dialplan based on that. |
| 22:07.58 | WIMPy | Ariel_Calzada: The HiPath does not support en-block dialling. |
| 22:08.26 | ManxPower | ah, a 1.6 specific issue. I can't help with that. Are your phones too stupid to do non-DTMF pasking? |
| 22:08.59 | jamko | ManxPower: lol .. no they are not.. It's really for call parking, but I noticed the issue was also with dtmf xfer, so I figured I would throw that in. |
| 22:09.24 | ManxPower | I can't think of any reason to use DTMF transfers for simple stuff. |
| 22:09.29 | ManxPower | use attended transfer to park |
| 22:10.16 | jamko | let me try that.. If that works, I will send you a keg. |
| 22:10.19 | Ariel_Calzada | WIMPy: there's a solution for this issue? |
| 22:10.23 | ManxPower | jamko, start with a noop(DIALSTATUS is ${DIALSTATUS}) after each dial and see if that provides anything useful |
| 22:10.58 | ManxPower | or heck, even a DumpChan() after each dial |
| 22:11.44 | WIMPy | Ariel_Calzada: Make user borth your channel driver and/or your dialplan support overlap dialling. |
| 22:11.54 | WIMPy | s/user/sure/ |
| 22:12.28 | WIMPy | Interesting typo *erm* |
| 22:14.30 | jamko | ManxPower: No dice on the attended xfer .. but I will see what I come up with for the DIALSTATUS .. thanks.. |
| 22:15.52 | WIMPy | BTW: Is there a way to make a faild blind transfer (not the DTMF type) go back to the transferer? |
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| 22:27.33 | Ariel_Calzada | WIMPy: we solved the problem, from the HIPATH 3550 the call now come complete, the problem now is from the asterisk to the hipath 3550 |
| 22:28.18 | Ariel_Calzada | WIMPy: when we send the call to the hipath the message is: "intercept call" |
| 22:28.33 | Ariel_Calzada | WIMPy: "interception call" |
| 22:28.42 | WIMPy | What messege? |
| 22:28.51 | Ariel_Calzada | winpy sorry "interception extension" |
| 22:29.09 | WIMPy | Where does that message come from? |
| 22:29.38 | Ariel_Calzada | WIMPy: ok it's not a message, the call is sended to another extension ( interception extension ) |
| 22:30.13 | WIMPy | Ah, then the dialled number doesn't match any configured one. |
| 22:30.20 | jamko | ManxPower: I was wrong, it is happening on the 1.4 as well, which means I have been getting away with doing it the wrong way for awhile now. Thanks again for the DialStatus info .. will start on my even longer weekend now.. |
| 22:30.30 | WIMPy | Do you have the HiPath connected to Asterisk or the other way round? |
| 22:31.44 | Ariel_Calzada | WIMPy: we call 0100 to the hipath but it make another extension respoonse |
| 22:32.04 | Ariel_Calzada | WIMPy: asterisk has a sip trunk of the hipath 3550 |
| 22:33.20 | WIMPy | Ok, SIP, I was just about to ask about number plans. Well, the HiPath might want the whole number, not just the extension. |
| 22:33.30 | WIMPy | But I haven't used SIP on a HiPath. |
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| 22:34.45 | Ariel_Calzada | WIMPy: oki |
| 22:35.28 | WIMPy | Is it a "trunk" on the HiPath as well? |
| 22:35.47 | WIMPy | I just know they are pretty picky with number matching. |
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| 22:43.15 | titter | Could someone explain what âimmediate,â âfull,â or âhalfâ means when being used with DAHDI buffers? |
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| 23:13.05 | [sr] | howdy WIMPy |
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| 23:24.46 | WIMPy | Hi [sr] |
| 23:24.55 | [sr] | time to bed WIMPy :) |
| 23:25.23 | WIMPy | Yes. |
| 23:25.44 | WIMPy | But first it's time for a snack :-) |
| 23:27.33 | [sr] | not me!! |
| 23:27.36 | [sr] | uim going ouyt |
| 23:27.39 | [sr] | stay ok!! |
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