| 00:09.40 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
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| 00:35.57 | smeet2002 | nice feature I didn't know....add "s" to VoiceMailMain() and it skips password authentication... |
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| 01:50.43 | leifmadsen | smeet2002: heh ya that's been around for a bit :) |
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| 02:20.05 | leifmadsen | anyone write dialplan on wordpress blog posts and have any tips for code highlighting? :) |
| 02:22.14 | ChannelZ | errr |
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| 02:32.24 | smeet2002 | @leifmadsen yea..but I didn't know that...it was annoying to enter password all the time in my home network |
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| 03:04.19 | dan__t | I haven't yet done this because.. well, I don't know how. But I'll be returning an array of data from ODBC, and I'd like to run Swift() against each of those options. Can I iterate through an array or do I need to treat each element of an array as its own piece of data? |
| 03:07.55 | dan__t | Oh.... "ARRAY can only be written to, not read from.". So that means I can only read (assuming I know it's there) $ARR{val} or something |
| 03:08.00 | dan__t | er, var |
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| 03:31.09 | ChannelZ | Arrays aren't arrays like you think |
| 03:31.25 | ChannelZ | the ARRAY function is more like "multi-assign" |
| 03:33.50 | dan__t | I know they're not like I think, I'm trying to come uip with how Asterisk thinks of them, but I'm not able to find a good example |
| 03:34.10 | dan__t | I see, where delim = #, #val1#val2#val3#val4 etc etc |
| 03:37.47 | dan__t | May I bother you for an example please? |
| 03:39.38 | ChannelZ | sorry I dont use ODBC+Asterisk |
| 03:40.29 | dan__t | I can figure out the ODBC part, just looking for an example of what a normal array looks like. |
| 03:40.43 | dan__t | I'm sure I can hack that to pull data from ODBC rather than static assignment |
| 03:41.13 | ChannelZ | As I said Asterisk doesnt have arrays |
| 03:41.42 | dan__t | You know what I'm talking about because you said I was wrong. |
| 03:42.30 | ChannelZ | What, the ARRAY function? |
| 03:42.47 | dan__t | Oh, *multi*assign*. |
| 03:44.27 | dan__t | I think HASH might work. |
| 03:44.29 | ChannelZ | It's rather badly named |
| 03:45.07 | dan__t | Hehe, I get it now. The Array() part anyway. |
| 03:45.24 | dan__t | Array(var1=val1,var2=val2,var3=val3) etc etc |
| 03:46.43 | dan__t | Nope, that's like an auto-Array(), that won't work. |
| 03:48.16 | dan__t | http://ofps.oreilly.com/titles/9780596517342/asterisk-DB.html, under "Multirow Functionality with func_odbc" if you're curious. |
| 03:48.17 | ChannelZ | HASH is more of a proper array but in the end what is lacking is the means to do things with arrays like you do in programming languages like 'foreach' and such. |
| 03:49.44 | dan__t | Yep. |
| 03:50.14 | ChannelZ | in fact for doing much more than getting/setting a couple of columns, I'd probably write AGI |
| 03:50.36 | dan__t | i won't argue that |
| 03:51.10 | ChannelZ | indeed it's how I use databases with Asterisk.. my Caller ID/blacklisting lookups and things are PHP |
| 03:51.50 | dan__t | hehe |
| 03:51.53 | dan__t | I get it now, thanks |
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| 03:58.58 | *** join/#asterisk Micc (~Micc@c-98-232-46-178.hsd1.wa.comcast.net) |
| 03:59.25 | Micc | I've got a realtime question if there is anyone alive in here. |
| 03:59.50 | ChannelZ | oh, the irony! |
| 04:00.27 | Micc | ChannelZ, well the last ten times it feels like I get no response when I ask a question. anyways... |
| 04:01.32 | Micc | Can I put a ton of sip peers info in the database and have asterisk only use the ones that register? I mean if I can only put 500 to 700 sip peers on a box, can I have 10K in the database and only have it use the ones that actual register to it? |
| 04:01.40 | ChannelZ | I was just making an ironic joke about the term "realtime" |
| 04:01.56 | Micc | yeah, I just realized that. pretty funny. |
| 04:03.10 | dan__t | Well. I can't interrupt Swift(). Period. I can't treat it like I would Background(). That sucks. |
| 04:04.55 | Micc | It looks like 1.6.2.20 only fixes one or two bugs, so I would assume its just as broken as 1.6.2.19 then, or is it pretty good? |
| 04:05.15 | ChannelZ | dan__t: Never used Swift but according to a certain wiki there are arguments for it |
| 04:05.34 | dan__t | There are, but they don't do anything with that regard 'til after Swift() exits |
| 04:05.46 | dan__t | Or "stops talking", rather. Before it exits, though. |
| 04:13.41 | *** join/#asterisk riwarren (~androirc@S0106c0c1c0227c72.vf.shawcable.net) |
| 04:16.19 | riwarren | Got a quick question. Ive setup an asterisk server using a voip provider for an outgoing call center/autodialer. We are looking at the option of hosting our own trunkbwith a TDM card. But confused aboutba few things. If we got an analog card with 1 fxo port, would that mean we could only have one outgoing call at a time? Or is it just the incoming calls that would be restricted? |
| 04:19.16 | WIMPy | It is one call, no matter what direction. |
| 04:19.41 | WIMPy | You are probably looking for a PRI (card). |
| 04:20.04 | riwarren | You mean like a digital t1 card? |
| 04:20.36 | riwarren | Im trying to avoid having to pay for a t1 line, can a dsl connection suffice for that? |
| 04:21.48 | *** join/#asterisk p3nguin (~xwQ5kwYl6@cobalt.esxi.hosts.a2infotech.com) |
| 04:22.03 | WIMPy | Didn't you say you're already using an ITSP? |
| 04:22.27 | riwarren | Yes I am. We want to cut costs by running our own trunk |
| 04:23.04 | WIMPy | Well, either you get telephone lines or you let an ITSP do it for you. |
| 04:24.27 | riwarren | Or a t1 line. Hmmm. A dsl modem cant act as a t1 line? T1 is only like 1.544 Mbit upstream no? |
| 04:24.53 | X-Rob | riwarren, a T1 is 24 simultaneous calls. |
| 04:24.58 | WIMPy | Don't know what the situation is in the US, but E1 are usually delivered via DSL. |
| 04:25.09 | X-Rob | an E1 is 30 simultaneous calls |
| 04:25.22 | WIMPy | But the transmission technology shouldn;t be your concern. |
| 04:26.11 | WIMPy | You order an interface and don't care how it gets to you. |
| 04:27.06 | riwarren | Yes rob. And right now we are autodialing 24 similtaneous calls via an itsp. Whom tells us that we can run as many outgoing calls as we want from our one DID, so there must be a way to do that ourselves |
| 04:27.28 | riwarren | What does an E1 line usually cost? |
| 04:27.39 | X-Rob | you buy a 1/24th of a T1 for every simultaneous calls you want. |
| 04:27.49 | X-Rob | (assuming you're in the US) |
| 04:28.09 | riwarren | Canada |
| 04:28.22 | WIMPy | Not sure. I think standard is around 300.- but if you take more, you easily get below 100.-. |
| 04:28.37 | riwarren | A month? |
| 04:28.39 | X-Rob | May I point out that the point of an ISSP is that it's cheaper than running the lines yourself. |
| 04:28.48 | X-Rob | ITSP |
| 04:28.56 | X-Rob | so I'm not really sure what you're aiming for here. |
| 04:29.10 | riwarren | How can it be cheaper? How do they make money then? |
| 04:29.18 | X-Rob | they buy lots and lots. |
| 04:29.24 | WIMPy | Phone lines are cheper than IP here. At least for large volumes. |
| 04:29.54 | X-Rob | riwarren, you'll need to speak to your telco about it. |
| 04:29.54 | riwarren | Can you get 30 phone lines intoba residential address? |
| 04:30.08 | X-Rob | and you'll need to learn about ISDN |
| 04:30.19 | X-Rob | (start with wikipedia, and work from there) |
| 04:30.25 | WIMPy | You need to ask your Telco. But I can;t see why they wouldn;t do it. |
| 04:30.42 | riwarren | Ok thanks. Bah I hate telus lol |
| 04:32.17 | WIMPy | And I can't see why I always hit ; instead of ', either. |
| 04:33.46 | riwarren | its because the knuckle in your pinky finger is not made allow your finger to move left and right, and muscles in that finger are weak |
| 04:34.20 | WIMPy | Or I need a new brain. |
| 04:35.37 | riwarren | Thanks for clearing that up for me. I took a job setting up a pbx server and got it done with no prior telephony knowledge, surprisingly, but now I'm getting asked all these questions I dobt know answers to lol |
| 04:41.34 | p3nguin | <riwarren> Yes rob. And right now we are autodialing 24 similtaneous calls via an itsp. Whom tells us that we can run as many outgoing calls as we want from our one DID, so there must be a way to do that ourselves <--- someone is confused. DIDs aren't used for outgoing calls AT ALL. |
| 04:42.03 | riwarren | I know a did is an incoming number |
| 04:42.15 | riwarren | But its our did that,shows up on call display |
| 04:42.20 | p3nguin | Then it's the ITSP that is confused, I guess. |
| 04:42.37 | p3nguin | The number that shows up on a call display is just Caller ID. |
| 04:42.55 | p3nguin | It *CAN* be the same number as your DID, but they aren't related. |
| 04:43.32 | riwarren | Ok I was not aware of that. I thought they were the same as oura is the same number |
| 04:44.13 | p3nguin | Many ITSPs allow you to put any number in the caller id. I can put your phone number in my caller id, for example. |
| 04:44.19 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 04:44.45 | p3nguin | Anyway, just wanted to clear that up. |
| 04:44.45 | riwarren | So long story short, I would be better off telling my client to simply upgrade his upstream internet connection and keep his itsp account topped up,with funds? |
| 04:45.22 | p3nguin | It'll be cheaper than going with a T1 circuit. |
| 04:45.42 | riwarren | Hes wanting to increase to 48 simultaneous outgoing calls |
| 04:45.43 | p3nguin | It costs us around $300 each month for a T1 circuit in this area. |
| 04:46.08 | riwarren | Its not based on call volume? |
| 04:46.36 | p3nguin | For two T1s (for 46 or 48 calls, depending on circuit type), you're looking at over $500 every single month just for the phone lines. |
| 04:47.06 | p3nguin | With the ITSP, you pay for minutes used and that's all. |
| 04:47.32 | WIMPy | And you need enough internet bandwidth. |
| 04:47.43 | p3nguin | Or, if you have lots of minutes, you're better off getting an unmetered account, where you pay a set fee every month for unlimited outbound calls. |
| 04:48.11 | riwarren | I couldnt find an itsp that offered that |
| 04:48.26 | p3nguin | I think bandwidth.com is a popular one. |
| 04:49.54 | riwarren | Right now, 24 calls, 60 min, 8 hours, 5760 minutes a day. |
| 04:50.10 | p3nguin | Maybe broadvoice.com also. |
| 04:50.36 | riwarren | At $0.125 a minute, billed at 1/6, is over $100/day |
| 04:50.57 | riwarren | Err sorry $0.0125 |
| 04:51.44 | p3nguin | Some ITSPs do not allow autodialers/telemarketing. |
| 04:51.45 | WIMPy | Ouch |
| 04:53.03 | p3nguin | That's a lot of minutes. |
| 04:53.15 | p3nguin | That's more than most people use in an entire month. |
| 04:53.25 | WIMPy | That's a lot per minute. |
| 04:54.52 | p3nguin | For that kind of volume, 1.25 cents per minute certainly is. |
| 04:55.13 | WIMPy | Is that a normal price otherwise? |
| 04:55.18 | p3nguin | I would have expected more like 0.5 cents per minute. That company is getting rich off them. |
| 04:55.44 | p3nguin | I think I pay around 1 cent per minute for my very low volume of calls. |
| 04:56.04 | p3nguin | I think it's 1.05 cents per minute to be precise. |
| 04:56.24 | WIMPy | And on a phone line it would be even more? |
| 04:56.44 | riwarren | The client in question is a registered charity, not telemarketing |
| 04:56.53 | p3nguin | If I called over a phone line, I wouldn't pay for minutes at all. |
| 04:57.08 | p3nguin | I'd pay for the service each month and calling is included. |
| 04:57.15 | WIMPy | Ok, that's better. |
| 04:57.51 | riwarren | Phonebooth.com looks good, but will it let me run as many outgoing calls simultaneously as my bandwidth will permit? |
| 04:57.53 | p3nguin | A typical price for a regular landline with nationwide unlimited calling included is around $60 per month. |
| 04:58.34 | WIMPy | Hmm. Telephony seems to be quite expensive there. |
| 05:00.28 | p3nguin | For small business, phone and DSL internet, it starts at $70/mo. |
| 05:01.11 | WIMPy | Here you get the standard package (16/1 DSL + BRI) for around 30. |
| 05:01.23 | WIMPy | Including unlimited calls to national landlines. |
| 05:02.03 | p3nguin | AT&T is offering their U-Verse residential phone service starting at $35/mo. |
| 05:02.36 | WIMPy | That BRI is off course a crippeld one coming out of a SIP gateway. |
| 05:03.59 | p3nguin | They don't want me to price just internet/phone for some reason. |
| 05:04.20 | p3nguin | They'll show me internet/phone/tv... starting at $89/mo for the first 12 months. |
| 05:04.40 | p3nguin | They'll show me internet/tv... |
| 05:05.01 | WIMPy | Yes, they all want to sell that here, as well. |
| 05:05.30 | riwarren | (riwarren) Phonebooth.com looks good, but will it let me run as many outgoing calls simultaneously as my bandwidth will permit? |
| 05:06.00 | p3nguin | The last time I checked pricing, they weren't offering the U-Verse services here yet. Nice to see they're screwing with things again. :/ |
| 05:07.14 | riwarren | <PROTECTED> |
| 05:07.55 | p3nguin | You'd have to ask them about limitations. I don't use their services, so I don't know. |
| 05:09.45 | riwarren | K thanks will do |
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| 05:14.52 | *** join/#asterisk tzafrir_laptop (~tzafrir@212.179.75.202) |
| 05:38.53 | *** join/#asterisk qakhan (~qakhan@203.130.22.202) |
| 05:39.16 | qakhan | hi all |
| 05:39.40 | qakhan | can anyone plz tell me how to setup call forwarding on cell phone |
| 05:41.09 | justdave | on a cell phone or to a cell phone? |
| 05:41.27 | ChannelZ | Ask Apple |
| 05:41.34 | justdave | (if you really mean "on" then it's completely off-topic for this channel ;) |
| 05:41.38 | ChannelZ | or whatever |
| 05:42.29 | qakhan | to a cell phone |
| 05:44.53 | ChannelZ | There isn't any automatic way.. but you can make your Asterisk extension Dial() something else (like your cell phone number) |
| 05:45.12 | p3nguin | We've already been over this in its entirety earlier. |
| 05:45.31 | ChannelZ | Or if your normal device is a SIP phone, most of them have a forward function of their own |
| 05:45.32 | p3nguin | There's a word I'm thinking of to describe the situation. |
| 05:45.35 | p3nguin | "dense" |
| 05:45.43 | ChannelZ | Futile? |
| 05:46.02 | justdave | if you're using one of the third-party config guis for asterisk, a lot of them provide a feature code for it |
| 05:46.53 | qakhan | guys let me describe you what i want |
| 05:47.04 | p3nguin | I explained carefully that call forwarding is not a feature of Asterisk, but of the phones. I learned that he's using something called SJphone, but he refuses to see if it has a call forward button. |
| 05:47.45 | p3nguin | Using sequential Dial() commands was also covered, but it apparently still wasn't good enough. |
| 05:48.40 | WIMPy | p3nguin: How do you forwar a call if the phone isn't reachable on the phone? |
| 05:49.18 | qakhan | i have 100 users and most of them want their call forward to their cell phone, but there is a company policy only sales user call activate their call to be forward to their cell phone. not everyone can do that |
| 05:50.31 | ChannelZ | so hard-code it into the dialplan and make them ask you to do it. NEXT! |
| 05:51.14 | qakhan | dear there are 50+ sales persons |
| 05:51.27 | qakhan | i want it to do their own |
| 05:51.35 | ChannelZ | Ok coacoa-muffins |
| 05:52.39 | qakhan | i want to setup an IVR which map their cell number with their ext and if user is unavailable then call forward to their cell phone |
| 05:52.42 | sunfone | SJphone is a linux softphone AFAIK |
| 05:52.54 | ChannelZ | So do it |
| 05:52.55 | p3nguin | wimpy: You don't -- you just Dial() another location. |
| 05:53.06 | ChannelZ | Use ASTDB or something to keep track of who gets forwarded where |
| 05:53.07 | p3nguin | wimpy: That's not "call forwarding," though. |
| 05:53.50 | WIMPy | Sure it is. And for remote voip clients an important one. |
| 05:54.34 | sunfone | coacoa-muffins... heh |
| 05:54.36 | sunfone | lol |
| 05:54.48 | qakhan | Channelz what is ASTDB? |
| 05:54.58 | *** join/#asterisk elec- (~elec@64.89.7.253) |
| 05:55.04 | ChannelZ | A little built-in database in Asterisk |
| 05:55.25 | WIMPy | qakhan: That's where you want to store your CF configuration. |
| 05:55.26 | qakhan | can u plz help me in this |
| 05:55.30 | ChannelZ | Look up 'hot desking', there's a ton of examples to give you the basic idea |
| 05:55.55 | qakhan | can u send me some? |
| 05:55.55 | elec- | anyone know why i might be getting invites with a + in front of the from and to? the + means something special? |
| 05:56.03 | ChannelZ | GOOOOOOGLE |
| 05:56.12 | ChannelZ | ~book |
| 05:56.12 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://ofps.oreilly.com/titles/9780596517342/ or ~buybook |
| 05:56.16 | sunfone | qakhan: if you load FreePBX on top of your asterisk installation you will get a web interface that lets the users setup their own "follow-me", which I think is what you are trying to accomplish |
| 05:56.19 | ChannelZ | There's two |
| 05:56.37 | WIMPy | elec-: It means it is followed by an complete unformatted number. |
| 05:57.11 | p3nguin | wimpy: It's not call forwarding. Dial()ing a phone is just dialing a phone. If I use Dial(SIP/wimpy) to call your phone, that's not forwarding. Likewise, if I use Dial(SIP/voipms/13145551212), that's also not forwarding. It's just calling another location just like any other Dial() command. Forwarding is something completely different. |
| 05:57.42 | ChannelZ | uh oh, down another rabbit hole |
| 05:57.44 | elec- | wimpy so its like for telling the receiving side, to strip any kind of special strings? |
| 05:57.48 | sunfone | "tromboning" might be a more accurate term |
| 05:57.57 | elec- | chars* |
| 05:57.59 | ChannelZ | sounds dirty |
| 05:58.03 | sunfone | heh |
| 05:58.06 | WIMPy | From the users perspective it is call forwarding. How you implement it in your dialplan is another thing. |
| 05:58.08 | ChannelZ | I like it |
| 05:58.23 | WIMPy | But you might use real forwarding there as well. |
| 05:58.23 | elec- | ah okay |
| 05:58.36 | p3nguin | wimpy: I don't care what a "user" thinks it is; the user has no business touching a phone system. |
| 05:58.44 | sunfone | With ISDN signaling you can do a real forward from within the dialplan |
| 05:58.51 | sunfone | with a PRI |
| 05:59.15 | ChannelZ | HEY! Which Digium analog card do I need to do SS7? |
| 05:59.24 | p3nguin | the expensive one |
| 05:59.28 | sunfone | you don't - you need a digital card |
| 05:59.37 | ChannelZ | runs away giggling |
| 05:59.41 | WIMPy | sunfone: You missed something :-) |
| 05:59.44 | p3nguin | haha, you got me! |
| 05:59.49 | sunfone | facepalm |
| 05:59.55 | *** join/#asterisk gogasca (~gogasca@166.205.136.217) |
| 06:03.22 | jeffspeff | is anybody familiar with how to modify the softkey xml file for cisco 7945 phones? I've found a default file, but can't figure out how to modify it, to make it dial and/or do different things. |
| 06:03.45 | *** join/#asterisk BuenGenio (~BuenGenio@cm61-10-82-188.hkcable.com.hk) |
| 06:05.41 | gogasca | Im familiar with it u have different states during a call u edit the softkeys u want for each state in the softkey file defined in the configuration .XML file |
| 06:06.25 | gogasca | I don't have it handy now but shoot me an email and I can send u my sample config |
| 06:06.38 | *** join/#asterisk lanning (~lanning@173-8-187-197-SFBA.hfc.comcastbusiness.net) |
| 06:16.00 | justdave | is there any timetable on getting RHEL6 packages built for asterisk? |
| 06:21.00 | qakhan | channelz i save cell number in DB plz tell me now what is the code for call to be forward if user not available |
| 06:23.44 | ChannelZ | Well you have to write it.. make a special extension someone dials that asks them to type in the number you want to 'forward' to, etc. and store it in the DB. Then you need to hook that up to the rest of your dialplan so if someone dials the extension for person X it looks up in the DB to see if they are 'forwarding' and Dial appropriately |
| 06:24.26 | *** join/#asterisk cerberus_za (~coert@196-215-103-15.dynamic.isadsl.co.za) |
| 06:26.54 | qakhan | yes here i got my completely |
| 06:27.00 | qakhan | thats i want to do |
| 06:27.10 | qakhan | plz help me :( |
| 06:33.17 | justdave | looks like atrpms has it |
| 06:33.31 | ChannelZ | It's more than just adding a couple of lines to your dialplan, I can't write the whole thing for you. |
| 06:34.37 | justdave | if you're having that much trouble writing it, you might be well off to use one of the GUIs that already implements it |
| 06:35.16 | ChannelZ | or pay someone |
| 06:35.29 | ChannelZ | Roughly you could make an exten that uses Read() to let them type in their extension and Read() again the number they want to forward to, and use Set() with the DB function to set an item in the AstDB. |
| 06:36.23 | ChannelZ | Then for your normal dial extensions, you use the DB() function again to read the forward number for their exten, and if it exists, Dial() that number instead of their normal device |
| 06:36.44 | ChannelZ | But depending on how your dialplan is currently setup in the first place, there's a dozen ways to do it |
| 06:45.24 | qakhan | ok i have saved cell number in DB |
| 06:45.55 | qakhan | now what is the code to read the DB and dial the numver |
| 06:45.58 | qakhan | number* |
| 06:52.58 | ChannelZ | well it's the DB function that you use kind of like a variable.. NoOp(Value is ${DB(foo/bar)}) |
| 06:54.30 | qakhan | ok setup exten => _12,1,Dial(${DB(CFIM/${EXTEN})}) |
| 06:54.34 | qakhan | is it right? |
| 06:55.03 | ChannelZ | sure, possibly |
| 06:55.26 | ChannelZ | depends on what is really stored in CFIM/12 in that example |
| 06:56.13 | ChannelZ | IE it would have to contain DAHDI/1/1115551212 or SIP/someplace/1115551212 or the like, depending on your setup |
| 06:56.43 | qakhan | what i did |
| 06:57.22 | qakhan | call forward of ext 3288 to 111222333 |
| 06:57.41 | qakhan | exten => _*21*X.,1,Set(DB(CFIM/${CALLERID(num)})=${EXTEN:4}) |
| 06:57.57 | qakhan | i saved the cell number through this |
| 06:58.57 | qakhan | is it right Dial(DAHDI/g1/${DB(CFIM/${EXTEN})}) to forward the call |
| 06:59.13 | ChannelZ | ok so you need to add the technology to the Dial since it's not stored in there.. otherwise your Dial command is bogus (IE Dial(12345) is invalid) |
| 06:59.29 | ChannelZ | Yeah that's better |
| 07:01.03 | WIMPy | See, that worked out a lot easier than SS7 on analog. |
| 07:01.54 | ChannelZ | And he learned something new. Confidence soars! |
| 07:01.56 | qakhan | yayayayayayyaya |
| 07:02.04 | qakhan | its working....... :) |
| 07:02.20 | qakhan | i love you channelz :) |
| 07:02.30 | ChannelZ | Yes dear! |
| 07:02.42 | qakhan | you are the MAN |
| 07:03.11 | ChannelZ | looks down |
| 07:03.21 | dan__t | Alright... just to see what the env vars looked like, I made a simple AGI script that piped 'set' to a temp file. I didn't see any of the variables that I created in * through ODBC commands, things like that. |
| 07:03.22 | ChannelZ | Yes, yes I am |
| 07:03.37 | dan__t | I guess that's expected - but how can I Make those available? Is my only option to send them as args to my agi script? |
| 07:03.53 | dan__t | User-defined channel variables, that is. |
| 07:05.15 | ChannelZ | not sure exactly what you are meaning.. the 'get variable' AGI command can fetch channel variables |
| 07:05.30 | dan__t | Oh, duh. |
| 07:05.46 | dan__t | Thank you. |
| 07:06.19 | ChannelZ | sho thang |
| 07:08.27 | AlecTaylor | What project can I use for a web-frontend (Java, Flash/Flex, Javascript) to connect to a conference-call, with moderation capability (mute caller)? |
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| 07:46.05 | *** join/#asterisk Marquel (~Marquel@static.132.171.47.78.clients.your-server.de) |
| 07:48.16 | Marquel | good morning. |
| 07:49.23 | Marquel | i have a problem with dahdi-2.4.1 on 2.6.39 kernel (with pax/grsecurity patches): every time i try running dahdi_cfg i get "Operation not permitted(1)", though this configuration worked great before i switched to the new kernel. |
| 07:50.43 | jeffspeff | anybody have a featurepolicydefault.xml for a 79xx series phone? 7945 would be great. |
| 08:15.40 | dan__t | Weird. Using STREAM FILE via AGI, I read: digit pressed: 200 result=<digit> endpos=<offset> |
| 08:15.58 | dan__t | endpos appears to be correct, but result is always two digits lower than what I actually send in dtmf |
| 08:16.19 | dan__t | Or, is result in some other character set or something...? |
| 08:17.18 | dan__t | Oh, huh - "or the ASCII numerical value of the digit if one was pressed" |
| 08:17.37 | dan__t | Why..... bother? Why not just return the digit that was found....... |
| 08:19.06 | dan__t | That makes -1 sense. |
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| 08:41.42 | qakhan | channelz what wrong with this code |
| 08:41.45 | qakhan | exten => _32XX,2,GotoIf(${EXTEN}=${DB(CFIM/${EXTEN})}?3:5) |
| 08:41.46 | qakhan | exten => _32XX,3,PlayBack(welcome) |
| 08:41.56 | qakhan | exten => _32XX,5,VoiceMail(${EXTEN}@itc,u) |
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| 08:54.12 | qakhan | channelz u there? |
| 08:56.40 | SunTsu | qakhan: http://www.catb.org/~esr/faqs/smart-questions.html |
| 08:57.04 | kaldemar | qakhan: gotoif syntax. expression needs to be surrounded with $[]. |
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| 09:15.16 | qakhan | it doesnt work |
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| 09:27.21 | adnc | hello, I've a problem with my asterisk, incomming calls are interrupted after 15 minutes. what could be the problem? |
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| 11:52.12 | freeedrich| | hi, how would I go about.. I have an incoming call from skype, which I forward to my cellphone via exten => skype,1,Dial(${MYMOBILE}) |
| 11:52.48 | freeedrich| | now the thing is... the skype call gets attended as soon as asterisk starts dialing my mobile |
| 11:53.20 | freeedrich| | so.. the caller has a big round of nothing until I either attend or my mailbox jumps in. |
| 11:53.52 | freeedrich| | so.. how would I let the skype call ringing until mymobile is picked up by either me or the mailbox? |
| 12:04.47 | Marquel | nobody? |
| 12:09.48 | freeedrich| | actually.. I just tested again - it seems it really only answers as soon as the mobile line is being picked up. |
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| 12:30.55 | adnc | anyone here who could help me understand why my asterisk interrupts incoming calls exact after 15 minutes? |
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| 12:42.08 | WIMPy | adnc: session-timers? |
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| 14:25.08 | Kobaz | hmm |
| 14:25.26 | Kobaz | okay... asterisk 1.8.5, Answer() MusicOnHold() no music |
| 14:25.29 | Kobaz | what's a good way to debug |
| 14:25.39 | Kobaz | <PROTECTED> |
| 14:27.00 | p3nguin | Check that default has files in it: moh show files |
| 14:27.02 | ChannelZ | moh show classes |
| 14:27.13 | ChannelZ | and that one too |
| 14:27.43 | p3nguin | I also use the echo test a lot when something like that arises. |
| 14:31.14 | Kobaz | yeah |
| 14:31.22 | Kobaz | i mean like, all that stuff is good |
| 14:31.29 | Kobaz | i have the same setup on 234928374744 other servers |
| 14:31.40 | Kobaz | echo test sounds good |
| 14:31.42 | p3nguin | That number seems inflated to me. |
| 14:31.43 | Kobaz | all other audio works |
| 14:32.31 | Kobaz | p3nguin: yeah maybe a little |
| 14:32.37 | Kobaz | okay Echo() works |
| 14:32.54 | Kobaz | i'm using custom mode using madplay to play mp3s |
| 14:32.56 | p3nguin | Then it has to be moh. |
| 14:33.09 | p3nguin | Why not let asterisk play it normally? |
| 14:33.12 | Kobaz | i'm just wondering if something changed in 1.8, maybe i need to configure this differently |
| 14:33.15 | p3nguin | It'll use mpg123. |
| 14:33.41 | p3nguin | I think 1.2 was the last branch that you had to use hackery to play an mp3. |
| 14:33.46 | Kobaz | http://pastebin.com/t1X9hXPA |
| 14:34.10 | Kobaz | well i like streaming the mp3s rather than mode=files |
| 14:34.22 | p3nguin | Me too. mpg123 is my man. |
| 14:34.24 | Kobaz | because then every time someone gets put on hold for the first time, they start at the first file |
| 14:34.40 | Kobaz | i've never had a problem with madplay |
| 14:34.44 | Kobaz | lemme try my other 1.8 boxc |
| 14:34.49 | p3nguin | With streaming, that's not going to happen. |
| 14:35.33 | Kobaz | yeah |
| 14:35.34 | Kobaz | i know |
| 14:35.36 | Kobaz | which is why i do it |
| 14:36.00 | *** join/#asterisk TimeRider (steve@188-220-33-123.zone11.bethere.co.uk) |
| 14:36.12 | p3nguin | You're contradicting yourself. |
| 14:36.25 | Kobaz | nope |
| 14:36.43 | Kobaz | i do streaming through madplay so that you don't always get put on hold on the same file |
| 14:37.56 | p3nguin | And a random order using the normal mp3 playback method won't do that? |
| 14:38.59 | Kobaz | it starts at the beginning of the file |
| 14:39.20 | ChannelZ | Can't have anyone listening to a whole song! |
| 14:39.46 | p3nguin | Another contradiction. |
| 14:40.00 | p3nguin | Streaming files will never start over at the beginning. |
| 14:40.23 | ChannelZ | uhm.. He's saying the same thing you are... |
| 14:40.57 | p3nguin | He keeps trying to say he's streaming to prevent what streaming does. |
| 14:41.02 | ChannelZ | No |
| 14:41.29 | ChannelZ | He's saying he doesn't use files/random because of those reasons |
| 14:41.59 | p3nguin | I guess if you can understand all the double talk, you can help him. |
| 14:42.02 | Kobaz | p3nguin: you just want to pick a fight.. i'm fine with that |
| 14:42.10 | p3nguin | points out that his own moh works |
| 14:43.06 | Kobaz | <p3nguin> Streaming files will never start over at the beginning. |
| 14:43.15 | Kobaz | that's exactly what I said |
| 14:43.26 | Kobaz | so far, contradictions: 0 |
| 14:44.43 | p3nguin | Just so you know, using the NORMAL mp3 play method, which uses mpg123, does not restart the audio file until after it has played to the end. |
| 14:44.46 | ChannelZ | In any event you've double checked your madplay is spitting out the right type of data? |
| 14:45.18 | p3nguin | Once it starts, it can't stop and start at the beginning. |
| 14:45.49 | Kobaz | p3nguin: that's not the normal moh method though |
| 14:45.59 | Kobaz | which is mode=files directory=/foo |
| 14:46.05 | p3nguin | At least in the asterisk branch I use, that's how it is. If improvements have been made in newer versions, no one has reported it to me. |
| 14:46.25 | p3nguin | [mp3] |
| 14:46.25 | p3nguin | mode=mp3 |
| 14:46.25 | p3nguin | directory=/var/lib/asterisk/mohmp3 |
| 14:46.34 | p3nguin | Seems normal to me. |
| 14:46.42 | ChannelZ | sighs |
| 14:47.10 | p3nguin | No special trickery, no additional apps being used. |
| 14:47.16 | Kobaz | and where do you see that (as a default) in the musiconhold.conf.sample ? |
| 14:47.28 | p3nguin | Just good ol' asterisk playing mp3s how it knows to play mp3s -- with mpg123. |
| 14:47.51 | p3nguin | I'll have to go look at a sample. |
| 14:48.58 | Kobaz | this whole argument is really lame, but the point is. that's not the standard. yes it may work... it's not what I'm using, so it's not very helpful either |
| 14:49.07 | p3nguin | I don't see in my sample file what I just pasted. That doesn't mean it's wrong. |
| 14:49.16 | Kobaz | I never said it's wrong |
| 14:49.40 | p3nguin | mp3 is a valid mode. That much IS written in the sample. |
| 14:49.48 | Kobaz | agreed |
| 14:50.04 | Kobaz | i really like all the straw man arguments |
| 14:50.06 | Kobaz | but it's not helping |
| 14:50.40 | p3nguin | And using mp3 as the mode plays mp3s using mpg123. And playing back with mpg123 plays files from beginning to end without stopping to start over. That's what you've indicated you want. |
| 14:50.49 | Kobaz | that's fine |
| 14:50.55 | Kobaz | i don't disagree that it can do that |
| 14:51.03 | p3nguin | And you can randomize the files from the directory. |
| 14:51.03 | *** join/#asterisk x1user (~x1user@host-212-75-8-69.bbccable.net) |
| 14:51.20 | ChannelZ | he doesn't want that! |
| 14:51.26 | Kobaz | my point is I think there's a bug in asterisk, which i would like to further investigate |
| 14:51.28 | x1user | Hi, anyone using chan_mobile? Can you please share /etc/asterisk/mobile.conf ? |
| 14:51.37 | p3nguin | Then don't randomize it, I couldn't care less. |
| 14:52.13 | Kobaz | my question isn't "help me get music working, using any configuration" |
| 14:52.23 | Kobaz | my question is "This should work, why doesn't it, I think there's a bug" |
| 14:53.06 | ChannelZ | I'll do something productive and try it over here |
| 14:53.13 | Kobaz | :) |
| 14:54.08 | Kobaz | ChannelZ: you saw my config options in the pastebin? |
| 14:54.20 | ChannelZ | no |
| 14:55.00 | Kobaz | http://pastebin.com/t1X9hXPA |
| 14:56.13 | Kobaz | the other problem with using the built-in mp3 stuff is you can't control the volume other than quiet/loud |
| 14:56.52 | ChannelZ | It's working for me with the sample options, let me try yours |
| 14:57.23 | Kobaz | p3nguin: and... i've had nasty nasty problems with mpg123 sucking up 100% cpu, or lots of memory, and crashing |
| 14:57.41 | Kobaz | ChannelZ: yeah, the samples should work |
| 14:57.54 | Kobaz | some of my setups use regular options like mode=files |
| 14:58.07 | Kobaz | I don't have any problem with that method |
| 14:58.50 | ChannelZ | I get an error with your -Qrz |
| 14:59.08 | Kobaz | which version of madplay? |
| 14:59.31 | ChannelZ | 0.15.2 I guess |
| 14:59.43 | ChannelZ | -Q -r -z works though |
| 14:59.45 | Kobaz | madplay -V |
| 15:00.00 | Kobaz | interesting |
| 15:00.10 | Kobaz | i have the same madplay 0.15.2, but I don't get any errors |
| 15:00.17 | ChannelZ | I actualy got shitloads on the console |
| 15:00.36 | ChannelZ | "res_musiconhold.c:643 monmp3thread: poll() failed: Interrupted system call" |
| 15:00.42 | Kobaz | mmmmm |
| 15:01.00 | ChannelZ | anyways it's not parsing the options stacked up like that so just separate them and see what happens |
| 15:01.33 | Kobaz | are you just trying to run that on the commandline? |
| 15:01.41 | Kobaz | or you were getting other errors from asterisk |
| 15:01.46 | ChannelZ | Errors from asterisk |
| 15:01.56 | ChannelZ | (res_musiconhold) |
| 15:01.58 | Kobaz | if you're running it straight up, it wants a list of files |
| 15:02.00 | Kobaz | k |
| 15:02.24 | ChannelZ | As soon as I loaded moh I got one error, and if I tried to do MusicOnHold it spat out dozens of those a second |
| 15:02.31 | Kobaz | weird |
| 15:02.43 | Kobaz | and you used the same config from the pastebin? |
| 15:03.15 | Kobaz | and you have mp3s in the directory specified? |
| 15:03.19 | ChannelZ | yes, basically the same thing as the Asterisk 'solaris' class sample except your options |
| 15:03.20 | Kobaz | ..just sanity checking |
| 15:03.25 | Kobaz | k |
| 15:03.39 | ChannelZ | I tried the sample first which only differ from yours by having -Q vs -Qrz |
| 15:03.43 | Kobaz | yeah |
| 15:03.49 | ChannelZ | But as I said I just tried -Q -r -z and it's fine |
| 15:04.20 | ChannelZ | and yes I gave it the dir (although I only copied one file there) |
| 15:04.27 | Kobaz | -z is shuffle and -r is go forever |
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| 15:06.36 | Kobaz | the solaris options don't work for me either |
| 15:06.43 | Kobaz | my options do though |
| 15:06.52 | Kobaz | well madplay loads, but there's no music pumping through |
| 15:07.31 | Kobaz | ah, wait... /usr/bin/madplay worked |
| 15:07.40 | Kobaz | instead of just plain madplay |
| 15:07.55 | ChannelZ | you have the path in your .conf |
| 15:08.17 | ChannelZ | did you unload/reload res_musiconhold after changing the config? |
| 15:08.55 | Kobaz | yeah |
| 15:09.05 | Kobaz | hmm. res_musiconhold doesn't have much in the way of debugging |
| 15:09.09 | ChannelZ | Double-checked your mohmp3 dir and files are readable by whatever user your Asterisk runs as? |
| 15:10.11 | *** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
| 15:10.40 | Marquel | nobody with any hint to my dahdi problem? |
| 15:11.05 | Kobaz | yeah, i mean like, i have an image i set these systems up from |
| 15:11.10 | ChannelZ | Marquel: what problem? didn't see it |
| 15:11.15 | Kobaz | everything is the same in terms of files |
| 15:11.28 | Kobaz | what is different, is asterisk... this is one of the few 1.8's i have going |
| 15:11.47 | leifmadsen | 1.8! |
| 15:11.51 | Kobaz | but yeah, madplay is perfectly happy when run from the commandline, and spitting out audio data to stdout |
| 15:11.53 | ChannelZ | Well I'm running 1.8.5.0 and there is music coming out of it. |
| 15:11.58 | Kobaz | yeap |
| 15:12.08 | Marquel | ChannelZ: upgrading to kernel 2.6.39 i get "Operation not permitted" when running dahdi_cfg with the very same config which worked for months. |
| 15:12.11 | Kobaz | i have one 1.8 that does have music working through the madplay method |
| 15:12.14 | Kobaz | and one that doesn't |
| 15:12.32 | ChannelZ | Marquel: did you rebuild DAHDI with matching kernel headers? |
| 15:12.50 | *** join/#asterisk ferdna (~ferdna@cpe-67-10-220-35.elp.res.rr.com) |
| 15:12.53 | ChannelZ | It's drivers so they need to be built under the same kernel version |
| 15:13.36 | Marquel | ChannelZ: well, headers were 2.6.36 before and it worked with kernel-2.6.38. |
| 15:13.43 | ChannelZ | Kobaz: does "ps ax |grep madplay" show it running? |
| 15:14.36 | Kobaz | yeap |
| 15:14.48 | Kobaz | yeah that was the first thing i checked an hour ago |
| 15:14.53 | p3nguin | I bet pc -C madplay shows it, too. |
| 15:15.00 | p3nguin | ps -C madplay, even |
| 15:15.08 | p3nguin | pc -C obviously wouldn't. |
| 15:15.25 | ChannelZ | Marquel: perhaps you got lucky |
| 15:17.27 | Marquel | ChannelZ: interestingly running dahdi_cfg -vv tells me it gets killed by SIGKILL... |
| 15:17.34 | Kobaz | p3nguin: i tend to like ps ax or aux because it shows much more info |
| 15:17.42 | ChannelZ | I bet there's 1000 other argument combinations you can pass to ps too |
| 15:17.48 | Kobaz | yeah |
| 15:17.49 | Kobaz | there is |
| 15:17.54 | Kobaz | what i really like is ps auxf |
| 15:17.57 | p3nguin | Okay, then use ps -C madplay u |
| 15:19.11 | Kobaz | but what if i wanted to show asterisk and madplay |
| 15:19.19 | p3nguin | I wonder how many $0.99 Nachos Supremes from Taco Bell I could eat before I puke, pop, or just plain die. |
| 15:19.22 | Kobaz | ps aux | egrep "asterisk|madplay" |
| 15:19.45 | ChannelZ | You should try to find out and report back |
| 15:20.57 | coppice | p3nguin: consumed at speed, or pacing yourself? |
| 15:21.56 | p3nguin | either way |
| 15:22.22 | p3nguin | I figure I could shovel more in if I went fast. |
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| 15:25.49 | p3nguin | *sigh* To be honest with myself, I probably won't eat any Nachos Supremes at all today. :( As much as I want to, I doubt I'll actually do it. |
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| 15:27.16 | coppice | I suspect in this case going too fast will cause choking, and this will actually prevent death by slowing you down |
| 15:28.16 | p3nguin | Damned built-in preservation mechanisms. |
| 15:29.29 | coppice | never fear. blendtec and a feeding tube to the rescue |
| 15:30.18 | p3nguin | For amusement, I sometimes check the Darwin awards for ways to override those preservation mechanisms. |
| 15:30.29 | p3nguin | Creativity there is never lacking. |
| 15:30.36 | *** join/#asterisk catphish_ (~charlie@2001:9d8:2005:2::3) |
| 15:31.08 | coppice | I like the idea of your memorial being a "Will it blend?" video on youtube |
| 15:31.26 | *** join/#asterisk russellb (~russellb@asterisk/contributor-and-cool-guy/russellb) |
| 15:31.27 | *** mode/#asterisk [+o russellb] by ChanServ |
| 15:31.31 | catphish_ | how does asterisk handle conflicting dialplan entries? |
| 15:31.37 | Kobaz | last one wins |
| 15:31.45 | catphish_ | for example _9. and _900. |
| 15:31.54 | p3nguin | catphish_: The best match wins. |
| 15:31.55 | Kobaz | oh, that type of conflict |
| 15:32.06 | leifmadsen | russellb: you're kind of a big deal |
| 15:32.18 | leifmadsen | catphish_: that's not a conflict :) |
| 15:32.18 | catphish_ | best match? |
| 15:32.24 | leifmadsen | most explicit match |
| 15:32.32 | Kobaz | if someone dials 900xxxxxx it will match the 900 |
| 15:32.33 | leifmadsen | 9001 will match _900. instead of _9. |
| 15:32.37 | leifmadsen | Kobaz: we win! |
| 15:32.41 | Kobaz | if someone dials 93234234234 it will match the 9 |
| 15:32.41 | p3nguin | catphish_: _900. is more "exact" |
| 15:33.07 | catphish_ | leifmadsen: thanks, apparantly that's not working since we switched to a realtime dialplan |
| 15:33.21 | p3nguin | catphish_: And, as Leif said, it's not a conflict; it's just an overlap. |
| 15:33.27 | *** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
| 15:33.29 | leifmadsen | pukes at the thought of a realtime dialplan |
| 15:33.35 | Kobaz | yeah |
| 15:33.48 | leifmadsen | I fundamentally don't believe in the dialplan living in the database |
| 15:33.57 | Kobaz | i like database driven dialplan |
| 15:34.03 | Kobaz | but not realtim |
| 15:34.12 | catphish_ | is there a better way? |
| 15:34.13 | leifmadsen | Kobaz: +1 |
| 15:34.15 | leifmadsen | func_odbc |
| 15:34.18 | leifmadsen | that's the better way |
| 15:34.24 | leifmadsen | static logic, dynamic data |
| 15:34.55 | catphish_ | i'm not familiar with odbc sadly |
| 15:34.57 | leifmadsen | heck, you can even make the routing dynamic too using func_odbc and dundi |
| 15:35.10 | leifmadsen | catphish_: there is a whole chapter showing examples in the book I helped write :) |
| 15:35.19 | leifmadsen | Database Integration chapter is your friend |
| 15:36.47 | catphish_ | i'm using realtime because i need to add an remove large numbers of sip peers and their associated dialplan entries, though of course i'm happy to consider odbc if it can more effectively use mysql as a backend |
| 15:37.40 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 15:38.06 | catphish_ | according to the wiki, realtime extensions should load the full context into memory before looking for the best match |
| 15:38.13 | catphish_ | so i really don't know why i'd be having a problem |
| 15:39.19 | leifmadsen | catphish_: the point with func_odbc is that you shouldn't have to write anything to the dialplan to lookup routing/destination for an extension -- you just accept all possible options, then ask the database if there is data to be returned, then write logic to handle both returned data and no returned data |
| 15:40.33 | catphish_ | what does that avoid? |
| 15:40.47 | *** join/#asterisk trumee (~trumee@cpc3-cmbg14-0-0-cust113.5-4.cable.virginmedia.com) |
| 15:41.14 | trumee | anybody used zrtp with twinkle or zfone? |
| 15:41.30 | trumee | <PROTECTED> |
| 15:41.42 | trumee | I have canreinvite=yes specified in freepbx, and can see 'Native Bridging' in asterisk logs happening |
| 15:42.18 | leifmadsen | catphish_: it avoids changing the dialplan at all |
| 15:42.34 | leifmadsen | catphish_: you just change the data, not the dialplan |
| 15:42.49 | catphish_ | my in-memory dialplan is nothing but a list of contexts |
| 15:42.55 | leifmadsen | anyways, that chapter is your friend and better explains the "why" than what I can do in a couple of lines on IRC |
| 15:42.57 | catphish_ | can that be avoided? |
| 15:43.00 | trumee | is there any other magic apart from 'canreinvite=yes' to get zrtp work over the internet? |
| 15:43.18 | leifmadsen | catphish_: do you still need to do a 'dialplan reload' for that data to be available? |
| 15:43.43 | catphish_ | leifmadsen: sadly yes, only when adding a context though |
| 15:43.45 | leifmadsen | or does it try to ping the database for every single channel created and try to load the dialplan into memory for every channel? |
| 15:43.51 | leifmadsen | catphish_: then just use #exec |
| 15:44.02 | leifmadsen | write a very simple script to load the dialplan into memory |
| 15:44.16 | catphish_ | leifmadsen: but yes, for each call it still queries the db for that exten |
| 15:44.21 | leifmadsen | use PHP or whatever, load it into memory, and you're done and you're not further behind than using realtime dialplans |
| 15:44.31 | catphish_ | its only the list of contexts thats static |
| 15:44.44 | leifmadsen | I still think func_odbc is going to be significantly more efficient |
| 15:44.54 | leifmadsen | ok, banking time |
| 15:45.00 | leifmadsen | heads off on his trusty mountain bike |
| 15:45.14 | catphish_ | can func_odbc be used without statically defined contexts? |
| 15:45.58 | catphish_ | right now for each context i need to add [context1] switch => Realtime and reload |
| 15:46.05 | catphish_ | it doesnt reload the list of extens |
| 15:46.09 | catphish_ | just the list of contexts |
| 15:46.52 | catphish_ | but what's really puzzling me is that a colleague claims that most-explicit matching is broken, whereas the wiki says it should load the full context and then find the best match |
| 15:46.56 | catphish_ | i'll have to test more |
| 15:50.35 | p3nguin | Why do people call me using CID numbers like 9998887777 and 0000001111? Do they think I'll be like, "Oh, those look like valid phone numbers and I should probably answer because it could be an important call!" or something like that? |
| 15:53.55 | catphish_ | p3nguin: i'd complain to the provider |
| 15:54.00 | catphish_ | that should never happen |
| 15:54.19 | catphish_ | well at least not in this country |
| 15:54.48 | p3nguin | You're saying my provider should not allow a call with a bogus caller ID number to pass through? |
| 15:54.56 | catphish_ | no |
| 15:55.06 | p3nguin | No you're not saying that? |
| 15:55.14 | catphish_ | sorry, i'm not saying that |
| 15:55.24 | catphish_ | they should complain to the provider who sent them the call |
| 15:55.37 | p3nguin | So THAT provider shouldn't allow it? |
| 15:55.41 | catphish_ | until it reaches the provider who generated the phony number |
| 15:55.49 | p3nguin | I see. |
| 15:56.08 | catphish_ | and that provider should stop it or risk losing their upstream account |
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| 15:56.27 | p3nguin | If I complained to my provider, they'd tell me it's not their problem. Nothing more would happen. |
| 15:56.56 | catphish_ | ie. i can set any outbound CID I want for my customers, but if i started abusing that i'd probably get disconnected from my upstream providers |
| 15:56.59 | p3nguin | Now if I reported it to the FCC, the investigation may go a little deeper, but I think the end result would still be the same. |
| 15:57.46 | catphish_ | anyone large enough to provide a phone service to others is trusted to set valid CIDs for their customers |
| 15:58.08 | catphish_ | most likely it's a provider who is being dumb and allowing an end user to set their own |
| 15:58.26 | Marquel | catphish_: there's actually a feature for that. |
| 15:58.32 | catphish_ | for what? |
| 15:59.01 | Marquel | catphish_: it is used by companies which forward incoming calls to road warriors, so their employees see the actual caller, not always their company calling them. |
| 15:59.15 | Marquel | catphish_: setting "any" CID... |
| 15:59.18 | catphish_ | yes |
| 15:59.27 | catphish_ | but end users don't need to do that |
| 15:59.41 | Marquel | as said: companies may need to do it. |
| 16:00.01 | catphish_ | a company that needs to do that internally might ask to be trusted to do so |
| 16:00.11 | catphish_ | but i'd expect them to be terminated if they abused it |
| 16:00.22 | catphish_ | i understand the requirement |
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| 16:00.41 | catphish_ | my isp allows customers to set up call forwards to road warriors' mobiles with a web interface |
| 16:00.43 | Marquel | yeah, the terms&conditions of this feature require responsible use. |
| 16:00.56 | catphish_ | so we handle the callerid issue there |
| 16:01.54 | Marquel | yeah, but you'll run into problems if that should be possible by hitting just one button on a phone upon leave/enter office ;) |
| 16:02.29 | catphish_ | well we offer it to be configured with a web UI |
| 16:02.34 | catphish_ | rather than from the phone itself |
| 16:02.43 | catphish_ | so it can be fixed remotely if necessary |
| 16:02.55 | Marquel | ChannelZ: recompiling with more recent kernel headers have the very same problem: dahdi_cfg gets killed by SIGKILL |
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| 16:12.03 | Marquel | ChannelZ: okay, seems a problem with a null deref in the actual driver module. i will revert to an older kernel until that's fix'd. |
| 16:12.15 | gauner1986 | hi guys.. i'm trying to use asterisk 1.6 as sip client.. now on register my username for the sip server is an email-adresse which contains an @.. so i needed to configure it in sip.conf as followed: register => XX@t-online.de:XX@tel.t-online.de/XXX |
| 16:12.33 | gauner1986 | can i escape the @ in the email address somehow? |
| 16:12.48 | p3nguin | You don't need to escape it. |
| 16:12.55 | gauner1986 | [Aug 13 16:05:20] WARNING[2015] chan_sip.c: Got 423 Interval too brief for service XX@t-online.de@tel.t-online.de, minimum is 240 seconds |
| 16:12.55 | gauner1986 | [Aug 13 16:05:20] WARNING[2015] chan_sip.c: Got 404 Not found on SIP register to serviceXX@t-online.de@tel.t-online.de, giving up |
| 16:13.00 | gauner1986 | this is what i'm getting |
| 16:13.39 | p3nguin | Check the sample file for correct syntax. |
| 16:14.19 | gauner1986 | thanks.. i'll look again |
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| 16:47.52 | catphish_ | realtime mysql seems to make an insane number of identical queries |
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| 16:51.02 | trumee | anybody understand key generation with genmc for Linksys ATAs? |
| 16:51.33 | trumee | The genmc utility is here http://forum.voxilla.com/cisco-linksys-sipura-voip-support-forum/anyone-use-genmc-certificate-generator-37028.html |
| 16:52.03 | trumee | do i need the same mini-certificate and private key on both the ATAs which i want secured? |
| 16:59.20 | catphish_ | seems unlikely you should be using the same private key on 2 devices |
| 16:59.31 | catphish_ | but i have no idea about the device you're discussing |
| 17:02.03 | catphish_ | i'm pretty sure there's a problem with wildcard matching in the mysql live dialplan :( |
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| 17:12.27 | carrar | moocow |
| 17:13.06 | catphish_ | in soviet russia, moo say cow |
| 17:15.01 | carrar | Hai! |
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| 17:28.52 | trumee | catphish_: right |
| 17:30.21 | gauner1986 | [Aug 13 17:27:44] NOTICE[3887] chan_sip.c: Call from '<my user name>' to extension '<the phone number i have dialed?!?!?!?!>' rejected because extension not found. |
| 17:30.28 | gauner1986 | any idea what that could be about? |
| 17:30.56 | catphish_ | gauner1986: seems self-explanatory to me |
| 17:31.12 | gauner1986 | catphish_: enlight me please |
| 17:31.18 | catphish_ | normally it also lists the context it was looking in |
| 17:31.29 | catphish_ | it means the number you dialed doesn't exist in your native context |
| 17:31.42 | catphish_ | simples |
| 17:40.03 | gauner1986 | hm |
| 17:40.12 | gauner1986 | this is an excerpt of my config |
| 17:40.14 | gauner1986 | http://pastebin.com/2ySCe34N |
| 17:40.27 | gauner1986 | the number exists for sure ;) |
| 17:41.36 | gauner1986 | any obvious fault there? |
| 17:43.43 | p3nguin | gauner1986: Your extensions.conf is all messed up. |
| 17:43.56 | gauner1986 | p3nguin: thats only an excerpt |
| 17:44.06 | p3nguin | The part you showed me is bad enough. |
| 17:44.10 | gauner1986 | okay |
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| 17:44.23 | p3nguin | Where is this call coming FROM? |
| 17:44.37 | p3nguin | From SIP/gauner? |
| 17:44.50 | gauner1986 | yeah.. the client is signed in as gauner |
| 17:44.50 | catphish_ | gauner i assume |
| 17:44.54 | catphish_ | since thats what he pasted |
| 17:45.20 | p3nguin | What phone number are you dialing on that phone? |
| 17:45.51 | gauner1986 | 08003301000 that should go over the sip server i configured in sip.conf |
| 17:46.04 | gauner1986 | (my provider) |
| 17:46.15 | p3nguin | But you want to only send 003301000? |
| 17:46.18 | catphish_ | yes i don't see a problem with that |
| 17:46.22 | p3nguin | drop the 08? |
| 17:46.27 | catphish_ | oh yeah |
| 17:46.32 | catphish_ | that's an odd thing to do |
| 17:46.42 | catphish_ | shouldn't you have appended 44 |
| 17:46.51 | catphish_ | or not dropped the 08 |
| 17:47.02 | gauner1986 | why drop the 08? |
| 17:47.07 | p3nguin | That's what you wrote. |
| 17:47.13 | p3nguin | ${EXTEN:2} |
| 17:47.14 | ChannelZ | ${EXTEN:2} |
| 17:47.16 | catphish_ | EXTEN:2 |
| 17:47.17 | ChannelZ | JINX! |
| 17:47.18 | gauner1986 | ah okay |
| 17:47.18 | p3nguin | shave off the first two chars |
| 17:47.21 | catphish_ | lol |
| 17:47.27 | gauner1986 | wtf.. okay |
| 17:47.32 | gauner1986 | thats copy pasted |
| 17:47.33 | gauner1986 | ;) |
| 17:47.33 | catphish_ | your config :) |
| 17:47.36 | p3nguin | If you don't want to do that, remove the :2 |
| 17:47.41 | gauner1986 | i see |
| 17:47.57 | p3nguin | Also, remove the r option unless you need it. |
| 17:48.10 | catphish_ | and the 45 for that matter |
| 17:48.14 | catphish_ | for outgoing calls |
| 17:48.28 | p3nguin | exten => _0X.,1,Dial(SIP/t-online-out/${EXTEN}) |
| 17:48.37 | gauner1986 | okay |
| 17:48.44 | catphish_ | the X seems a little pointless too |
| 17:48.54 | p3nguin | catphish_: Good idea. Let the receiving side determine how long it'll ring. |
| 17:49.32 | p3nguin | Yeah, I don't know what that pattern is all about... but I also don't live in a country with goofy numbering plans. |
| 17:49.43 | catphish_ | exten => _0.,1,Dial(SIP/t-online-out/${EXTEN}) |
| 17:49.48 | catphish_ | that's all you need |
| 17:49.55 | ChannelZ | And we don't spell telecommunications with a 'k' |
| 17:49.55 | catphish_ | just to pass through numbers that begin with 0 |
| 17:50.04 | gauner1986 | yeah |
| 17:50.13 | gauner1986 | ChannelZ: i'm sorry ;) |
| 17:50.40 | p3nguin | I think we spell it that way in Germany. |
| 17:51.23 | p3nguin | Oh, well, you seem to be in Germany, so that does make more sense now. :/ |
| 17:51.58 | gauner1986 | -- SIP/t-online-out-005c6f20 is circuit-busy |
| 17:52.00 | gauner1986 | wtf |
| 17:52.04 | gauner1986 | circuit-busy? |
| 17:52.16 | catphish_ | means the number you're calling is busy |
| 17:52.18 | catphish_ | in theory |
| 17:52.19 | ChannelZ | It either likes your number less or it's busy. |
| 17:52.24 | p3nguin | They might not be accepting the number you're sending. |
| 17:52.38 | catphish_ | are you sure you're sending the right number format? |
| 17:52.58 | catphish_ | some providers require international standard numbers not local ones |
| 17:53.05 | ChannelZ | Perhaps you require an umlat |
| 17:53.24 | gauner1986 | [Aug 13 17:51:27] NOTICE[4054] chan_sip.c: Failed to authenticate on INVITE to '"02XXX" <sip:02XXXX@tel.t-online.de>;tag=as102a6d12' |
| 17:53.34 | gauner1986 | that seems to be the real problem |
| 17:53.39 | catphish_ | yes that'll do it |
| 17:54.05 | ChannelZ | You commented out your secret in sip.conf, any reason? |
| 17:54.05 | catphish_ | your secret is commented out |
| 17:54.09 | catphish_ | lol |
| 17:54.36 | gauner1986 | ChannelZ: yeah - whats it needed for? i already set up a username gauner with correct secret |
| 17:54.52 | catphish_ | you need to authenticate to the upstream provider |
| 17:55.00 | p3nguin | d'oh! |
| 17:55.02 | catphish_ | the password for your local phone is irrelivent |
| 17:55.13 | gauner1986 | that secret is already in register => user:pw:authuser@tel.t-online.de/My phone number at my provider |
| 17:55.21 | ChannelZ | That's just for the register. |
| 17:55.29 | catphish_ | oh yeah, i'd set it in both places anyway |
| 17:55.30 | ChannelZ | Registering has more or less nothing to do with actually calling |
| 17:55.38 | catphish_ | you missed the username too |
| 17:55.49 | catphish_ | and fromuser may be important |
| 17:55.53 | catphish_ | they can be quite fussy |
| 17:56.02 | catphish_ | and fromdomain |
| 17:56.33 | gauner1986 | hm |
| 17:56.39 | catphish_ | most providers require you to authenticate to make a call even if you're already registered |
| 17:56.43 | gauner1986 | they require to authenticate with phone number AND username |
| 17:56.54 | p3nguin | gauner1986: The register statement tells THEM how to reach you. The peer entry tells your Asterisk how to authenticate a call to them. |
| 17:57.00 | catphish_ | then put in the username and secret again |
| 17:57.23 | catphish_ | in theory registering would be enough to allow calls from your IP |
| 17:57.43 | gauner1986 | register => my phone number:pw:myemail@t-online.de@tel.t-online.de/my phone number |
| 17:57.48 | catphish_ | but they rarely leave it at that in case someone else picks up your IPs |
| 17:57.55 | gauner1986 | so where to put that myemail@t-online.de there? |
| 17:58.19 | catphish_ | "my phone number" is the username |
| 17:58.25 | catphish_ | pw is the secret |
| 17:58.25 | gauner1986 | yeah |
| 17:58.33 | gauner1986 | and that third thing? |
| 17:58.44 | catphish_ | i have no idea what your email is doing there |
| 17:58.58 | catphish_ | never seen 3 auth factors in a register line |
| 17:59.07 | gauner1986 | asterisk sample logs tell that this is the auth user |
| 17:59.27 | catphish_ | register => user[:secret[:authuser]]@host[:port][/extension] |
| 17:59.31 | catphish_ | there you go |
| 17:59.34 | gauner1986 | yeah |
| 17:59.40 | catphish_ | no idea how user and authuser are different |
| 17:59.53 | gauner1986 | i dunno.. but it doesnt work without it |
| 17:59.54 | gauner1986 | ;) |
| 17:59.54 | catphish_ | nor do i know which to put in user and which to put in fromuser |
| 18:00.08 | catphish_ | secret will be your pw |
| 18:00.19 | catphish_ | but username and fromuser are a mystery to me |
| 18:00.32 | catphish_ | usually for my providers they're the same |
| 18:00.46 | gauner1986 | okay |
| 18:00.49 | gauner1986 | but it did the trick |
| 18:00.49 | catphish_ | and fromdomain would be tel.t-online.de |
| 18:00.50 | gauner1986 | :) |
| 18:00.59 | catphish_ | you might have to try some combinations |
| 18:01.04 | gauner1986 | yeah |
| 18:01.06 | gauner1986 | it works already |
| 18:01.08 | gauner1986 | :) |
| 18:01.11 | catphish_ | id put username in username and fromuser :) |
| 18:01.13 | catphish_ | great :) |
| 18:01.18 | gauner1986 | thank you guys |
| 18:01.24 | catphish_ | no problem |
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| 18:39.30 | x1user | Call from '' to extension '' rejected because extension not found in context ''. ( I've stripped the real name and number) Why is that happening? |
| 18:40.26 | WIMPy | One should think that message is rather obvious. |
| 18:42.21 | robl^ | doesn't realtime have issues with devstate / hints? I recall reading something about that before, but I don't seem to see it now. |
| 18:48.03 | MI1 | good evening, is there a chance someone can help me with modifying config files for asterisk to enable calls from cell phone to sip phone? i am using openbts and asteris, i can only call from sip phone to gsm for now |
| 18:52.06 | MI1 | this is what i have so far - http://pastie.org/private/9fpldeywetnnng322zqa |
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| 18:57.35 | x1user | WIMPy: I got SIP user defined in sip.conf and exten=> mysipuser,1,Dial(SIP/mysipuser, 10) ? |
| 18:57.49 | x1user | should it me something like these? |
| 18:58.20 | ChannelZ | 'mysipuser' as an exten isnt necessarily what you want |
| 18:59.17 | ChannelZ | Probably why it's saying 'extension not found'. What were you actually trying to dial? (and from what? looks cyclical) |
| 18:59.22 | p3nguin | It'll be hard to dial a name. |
| 18:59.43 | p3nguin | And that space after the comma before the 10 will surely break Dial(). |
| 19:00.14 | x1user | it is not a name it is real number |
| 19:00.27 | p3nguin | Then call it what it is instead of "mysipuser" |
| 19:00.31 | x1user | i've dialed it from another asterisk, trying to set up another now |
| 19:00.32 | ChannelZ | but you have exten => mysipuser |
| 19:00.49 | p3nguin | That creates an extension mysipuser. |
| 19:00.58 | p3nguin | If you want a number, write a number. |
| 19:00.59 | ChannelZ | which again isn't necessarily wrong but isn't necessarily right either |
| 19:01.20 | p3nguin | It'll work if you can dial it from the phone, but that rarely is the case. |
| 19:01.22 | p3nguin | I know I can't do it. |
| 19:02.37 | p3nguin | Anyone here have an iPod Touch second or third gen they'd like to sell so they can get a newer model? |
| 19:05.43 | ChannelZ | never! |
| 19:06.50 | WIMPy | MI1: And what happens if you dial from your mobile? |
| 19:07.41 | MI1 | WIMPy, i got something that is not allowed to dial this |
| 19:08.02 | WIMPy | Who says so? |
| 19:08.40 | MI1 | same error like you dont put extensions for imsi, something is missing there |
| 19:08.57 | MI1 | not sure what |
| 19:09.20 | WIMPy | What does the console say? Do you have the right context? |
| 19:10.36 | MI1 | console say nothing, cell phone will not dial, all i got is in that pastie, it works but only one way |
| 19:11.18 | WIMPy | That was only config, no errors or logs. |
| 19:12.53 | MI1 | hmm, you are right, i thought i forgot something obvious what will be visible on first sight |
| 19:13.05 | MI1 | will try to get some |
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| 19:46.37 | jeffspeff | i'm needing to get my hands on a featurespolicydefault.xml file for the cisco 7945g phones. does anybody have it or know where i can find one? |
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| 23:34.13 | SVLD | hi2all, is anybody can help me? How I can assign incoming call (SIP) from provider with my registration information (asterisk connected to provider by few accounts)? |
| 23:35.39 | ChannelZ | You make a peer in sip.conf as such that it matches the calls coming from your provider |
| 23:35.56 | ChannelZ | Usually by IP address but it depends on what/how they send you calls |
| 23:38.50 | SVLD | 1. provider uses my asterisk as gateway and send INVITEs like SIP/accountname_from_few/number |
| 23:39.21 | SVLD | all accounts registers at same IP |
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| 23:41.15 | SVLD | I cant route incoming calls bi DID (DID in my case - number for I have to dial), and I cant route by peer (IP) - because all accounts on one IP |
| 23:41.25 | ChannelZ | well I can tell you that type=user matches by the From: header of the invite and type=peer matches by IP |
| 23:42.12 | ChannelZ | Why can't you route by DID |
| 23:42.44 | SVLD | I know, but in header From: provider sends some internally stuff, not match with my register information |
| 23:42.58 | ChannelZ | you can try fromuser=xxx |
| 23:43.50 | ChannelZ | I forget if that only applies to outgoing or not. |
| 23:43.53 | SVLD | I cant force provider to change his dialplan :( |
| 23:44.49 | ChannelZ | Can you pastebin a sip debug of one of their invites? |
| 23:45.03 | SVLD | I wanna know, can I assign incoming call with register account elsewhere |
| 23:45.55 | ChannelZ | I dont really know what you're asking. You're telling me your provider doesn't tell you what DID the call came in on? |
| 23:46.21 | ChannelZ | They only send some username and then you are left to guess? |
| 23:46.38 | sunfone | I can't believe that... |
| 23:46.52 | sunfone | pretty crappy provider :) |
| 23:50.02 | ChannelZ | that's why I'd like to see one of the sips |
| 23:50.48 | SVLD | provider tell me: "dial XXX number through YYY gate" |
| 23:51.17 | SVLD | From: "some stuff", To: XXX |
| 23:52.26 | SVLD | http://pastebin.com/a4nDtA7E |
| 23:53.23 | SVLD | I have to guess which gate to use for dial this number |
| 23:55.10 | ChannelZ | So "929" comes from them and is just made up or what? |
| 23:57.01 | ChannelZ | I mean, there is no other information in the invite that seems unique - I assume 786874236776 is the number of the phone you made the test call from |
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| 23:59.46 | SVLD | 786874236776 - is internal number of provider, it changed every call |
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