| 00:14.27 | *** join/#asterisk CaTtleyA (n=DDoS@bas1-montrealak-1128581558.dsl.bell.ca) |
| 00:15.48 | mackes | Where is everyone? |
| 00:20.15 | *** join/#asterisk d00gster (n=doughant@bas1-cooksville01-1279552398.dsl.bell.ca) |
| 00:20.52 | phix | mackes: I am here now! |
| 00:21.10 | phix | I slept in |
| 00:22.47 | maqr | how greedy are the pattern matches with exten? if i have _XX vs _XXX, does asterisk wait for 3 numbers to be dialed before deciding to follow the XX extension? |
| 00:23.42 | phix | it waits a few secs I think |
| 00:23.45 | phix | you could try it :) |
| 00:26.17 | maqr | true :) |
| 00:26.37 | mackes | yes |
| 00:26.45 | mackes | XXX means 3 numbers |
| 00:27.07 | mackes | _X. means 1 or more numbers |
| 00:27.23 | mackes | _XX. means two or more numbers |
| 00:27.54 | mackes | without the period, the X means one letter/number placeholder. |
| 00:28.04 | mackes | The book has a good section on this |
| 00:30.09 | maqr | mackes: yeah, i knew what that meant, i just wasn't sure how long it waited to figure out which one the user meant |
| 00:30.16 | maqr | i guess i can assume it's long enough |
| 00:30.35 | mackes | Oh.... Instantly |
| 00:30.54 | outtolunc | depends what you are doing, i've had instances that took 8 seconds |
| 00:31.20 | mackes | no no, Your SIP phone might wait that long before it sends the digits |
| 00:31.30 | mackes | However Asterisk is instant |
| 00:31.47 | mackes | So order matters in your extensions.conf |
| 00:32.03 | outtolunc | be rather hard as there was no sip involved, but you go right on thinking that <G> |
| 00:32.28 | mackes | Then its that long before your Zap sends it |
| 00:32.42 | mackes | Really, asterisk is instant |
| 00:33.50 | outtolunc | he was talking when the digit input was less than the pattern |
| 00:34.12 | mackes | If there was a delay, if asterisk waited, then order would not matter. But all the documentaton discusses how you need to order your dialing plan in such a way to not step on smaller extensions. |
| 00:35.09 | mackes | If it is less then a pattern, then it would not trigger an extension. |
| 00:35.22 | outtolunc | until a timeout.. sheesh |
| 00:35.48 | mackes | ok. I guess I don't understand |
| 00:37.26 | maqr | ohh, i see how it works |
| 00:37.28 | maqr | that's interesting |
| 00:39.31 | maqr | is there any way to dial DTMF style from a sip phone? |
| 00:40.54 | mackes | I guess what I am saying is that Asterisk does not compare all of your exten lines first and then pic the right one. It picks the first line that matches the pattern, and executes it. So, As soon as your device, ZAP, or SIP sends extension, it executes. The only delay is in your client device, while your client device waits for a pattern to be matched before it sends to Asterisk |
| 00:41.12 | maqr | well, SIP sends all at once, right? |
| 00:41.13 | maqr | like, always? |
| 00:41.14 | Strom_C | mackes: you're wrong in at least two ways |
| 00:41.23 | mackes | ok, How so |
| 00:41.27 | Strom_C | first, . matches one or more digits or characters, while ! matches zero or more |
| 00:41.55 | Strom_C | second, pattern matching within a context is always on a "most specific match first" basis with the exception of _. |
| 00:43.04 | mackes | OK, in rebuttal- I did say _X. means 1 or more numbers |
| 00:43.13 | Strom_C | _X. means two or more |
| 00:43.20 | Strom_C | _X! is one or more |
| 00:43.54 | mackes | No, _X. means one or more. I have it in my plans |
| 00:44.13 | Strom_C | sigh |
| 00:44.16 | mackes | sigh |
| 00:44.20 | Strom_C | _X. isn't going to match "3" |
| 00:44.24 | Strom_C | _X! will |
| 00:44.49 | mackes | <PROTECTED> |
| 00:45.04 | mackes | I use it for my long distance |
| 00:45.14 | Strom_C | _X. will match a three-digit string, yes, but it's not going to match any single-digit string |
| 00:45.33 | outtolunc | and you are dialing more that '1' to dial longdistance <G> |
| 00:45.55 | [TK]D-Fender | mackes> No, _X. means one or more. I have it in my plans <- No, it will match TWO or more. |
| 00:45.57 | mackes | ok.... whatever, and the pattern matching |
| 00:46.25 | [TK]D-Fender | And super wildcard matches like this are generally a silly idea. |
| 00:46.33 | mackes | fine |
| 00:46.50 | mackes | <PROTECTED> |
| 00:47.03 | Strom_C | what about the pattern matching? |
| 00:47.05 | mackes | Order does not matter? |
| 00:47.06 | seanbright | and :-) matches happy |
| 00:47.27 | mackes | You feel that Asterisk reads the whole plan and picks the best match? |
| 00:47.32 | Strom_C | order within the configuration file within a single context does not matter |
| 00:48.15 | mackes | So, if I have _1. on say line one of my extension.conf |
| 00:48.19 | Strom_C | if you have _XXXX followed by _23XX followed by _236X and you dial 2368, asterisk will match _236X |
| 00:48.25 | mackes | and _1xxxx on line 30 |
| 00:48.42 | mackes | and I dial 12345 |
| 00:48.50 | seanbright | the _1XXXX rule gets called |
| 00:48.55 | mackes | the system will pick 40 |
| 00:48.56 | mackes | 30 |
| 00:48.58 | maqr | if you dial from a DTMF phone vs a SIP phone, the DTMF will early-match and the SIP won't (since it gets it all at once), right? |
| 00:49.00 | Strom_C | mackes: if they're in the same context, asterisk will match _1XXXX |
| 00:49.11 | seanbright | its the most specific match |
| 00:49.17 | mackes | Over the _1. |
| 00:49.21 | Strom_C | yes |
| 00:49.38 | Strom_C | but having your dialplan with overlapping matches like that is a bad idea |
| 00:50.10 | Strom_C | the only time you should ever have a match like _1. is when you expect a varying number of digits to be dialed (i.e. international calling) |
| 00:50.25 | mackes | Why? From what you are saying, Asterisk always finds the best match? |
| 00:50.38 | seanbright | the best match is _1XXXX |
| 00:50.56 | seanbright | yes, it always finds the best match |
| 00:51.07 | Strom_C | there's a difference between what the software will do given a set of conditions and what you should do to ensure that your dialplan remains clean, comprehensible, and easy to maintain |
| 00:51.10 | seanbright | where best = most specific |
| 00:51.34 | mackes | I have found that I have to put the large open variables at the end of my contexts, or the override the smaller strings that they conflict with. |
| 00:51.37 | seanbright | if you dial 123456 or 1234 it will match against _1. |
| 00:52.03 | seanbright | mackes: what version of asterisk? |
| 00:52.06 | mackes | 1.2 |
| 00:52.09 | seanbright | oh |
| 00:52.22 | seanbright | 1.2 sucks |
| 00:52.23 | seanbright | upgrade |
| 00:52.25 | seanbright | (heh) |
| 00:52.45 | mackes | Well, I have heard the same about 1.4 |
| 00:52.59 | seanbright | you spend too much time in this channel or on the -users mailing list than |
| 00:53.08 | mackes | So, with 1.2, does anyone agree with what I am saying |
| 00:53.19 | mackes | I think they discuss this in the book |
| 00:54.30 | mackes | So, do you think 1.4 is ready for prime time? |
| 00:54.40 | seanbright | is that a serious question? |
| 00:54.45 | mackes | yep |
| 00:54.49 | seanbright | sigh |
| 00:54.52 | [TK]D-Fender | mackes, "Procrastination : the art of keeping up with yesterday" |
| 00:54.59 | mackes | Hmmmm |
| 00:55.02 | mackes | Maybe |
| 00:55.09 | [TK]D-Fender | mackes, Have you considered upgrading to Windows 3.11 yet? |
| 00:55.31 | mackes | However I am the person my users will call if the system flips out. Right now it is very stable |
| 00:55.46 | seanbright | mackes: don't fix what isn't broken |
| 00:55.57 | mackes | Wow fender. That is quite clever. |
| 00:56.00 | mackes | Thank you. |
| 00:56.02 | seanbright | mackes: but yes, no matter what the trolls say, 1.4 is just fine for production. |
| 00:56.09 | mackes | Ok, that is fair |
| 00:56.24 | mackes | I dont really have an opinion |
| 00:56.46 | mackes | I would move to 1.4, however I am not sure it has any features I need? |
| 00:56.53 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 00:56.56 | mackes | Why is it better then 1.2? |
| 00:58.59 | [TK]D-Fender | mackes, 1.2 does not get bugfixes other than security only. 1.4 fixed a lot of bugs, major changes to zaptel EC, etc. |
| 00:59.39 | mackes | I do have channel locks on My PRI card... for unknown reasons, once every few weeks |
| 00:59.53 | mackes | I have to reboot the whole system to clear them |
| 01:00.07 | seanbright | "Right now it is very stable" |
| 01:00.10 | mackes | And they are always on channels 2, or 3 stopping all other calls |
| 01:00.39 | mackes | besides the bug fixes, and the GUI (Which I dont want) what other neat things can it do |
| 01:00.47 | [TK]D-Fender | mackes, Any other questions you have that you'd like to answer in front of us? |
| 01:01.05 | [TK]D-Fender | mackes, How about you go read the changelogs and all the otehr wonderful docs out there. |
| 01:01.29 | mackes | Hmmmm... Or.. Or.. How about this, I ask in the Asterisk IRC channel? |
| 01:01.37 | mackes | Nannnnaaaa |
| 01:01.41 | mackes | That is crazy |
| 01:01.47 | seanbright | support vampirism |
| 01:02.11 | mackes | What else is there to discuss and debate in this room but these things? |
| 01:02.13 | [TK]D-Fender | mackes, Yeah you could wait around for the guy who's going to read it all off to you but then I guess the one who does deserves to be someones proxy. |
| 01:02.45 | adeel | well, what's the benefit of anyone 'reading it off'...he'll just be copying/pasting the changelog anyway.. |
| 01:03.03 | mackes | Ok. fine |
| 01:03.24 | adeel | mackes, the general rule of thumb for most things is, if it ain't broke, don't fix it |
| 01:03.24 | seanbright | mackes: you just said you were stable. there is no reason to upgrade. |
| 01:03.45 | mackes | Thank you. |
| 01:03.59 | seanbright | i deployed 1.4.19 3 months ago and have yet to touch the box again |
| 01:04.19 | seanbright | for the record. |
| 01:04.20 | [TK]D-Fender | mackes, Except for that "bug that lock all of my channels out" bit of course. |
| 01:04.38 | seanbright | [TK]D-Fender: right, but its "very stable" so no worries. |
| 01:04.53 | [TK]D-Fender | seanbright, Nothing more stable than a dead halt ;) |
| 01:05.00 | seanbright | agreed. |
| 01:05.07 | mackes | Yeah. I am not sure what causes that . Some times that happens with my Link to Verizon, and sometimes the D-Channel flips out with my link to a Nortel. |
| 01:05.19 | outtolunc | finds kernel panics very stable also |
| 01:05.47 | mackes | OK. |
| 01:06.19 | *** join/#asterisk bingnet922 (n=ken@70.99.220.206) |
| 01:07.00 | mackes | I will sit mute now for a while. I am interested to see what advanced conversation occurs that is not covered in written materials that could be reviewed offline. Fender, please start us off. |
| 01:07.02 | bingnet922 | Hello World. |
| 01:07.42 | [TK]D-Fender | mackes, yeah.. and did you know the word "gullible" isn't in the dictionary? |
| 01:08.30 | mackes | Very nice |
| 01:08.30 | *** part/#asterisk RoyK (n=roy@ip-26-13-149-91.dialup.ice.no) |
| 01:08.46 | *** join/#asterisk hsv-al (n=ding@user-24-214-126-81.knology.net) |
| 01:11.06 | bingnet922 | General question: What determines the number of simultaneous inbound calls an asterisk PBX can accomodate? IDID lines, truck ports? |
| 01:12.25 | outtolunc | your imagination (followed by system hardware, tech type, codec, recording, etc etc etc) |
| 01:12.59 | seanbright | i thought it was my imagination |
| 01:13.55 | outtolunc | only on tuesdays |
| 01:14.00 | outtolunc | sheesh |
| 01:14.02 | [TK]D-Fender | seanbright, you shouldn't let it wander... its too little to be let out alone ;) |
| 01:14.20 | seanbright | zing |
| 01:14.25 | mackes | So, I checked Voipinfo.org |
| 01:14.49 | mackes | I think I was right about the variable order (At least in 1.2) |
| 01:14.53 | mackes | http://www.voip-info.org/wiki/index.php?page=Asterisk+config+extensions.conf+sorting |
| 01:15.31 | mackes | 1.4 might have been different, however it would have been great if I was asked what version I was using before I was told I was wrong |
| 01:16.30 | *** join/#asterisk inv_arp (n=junya@c-76-26-24-197.hsd1.fl.comcast.net) |
| 01:16.55 | bingnet922 | outtolunc: it is necessary to subscribe to an PSTN -> IAX termination service, correct? In such a scenario, the number of simultaneous incoming calls is determined by...? |
| 01:17.09 | [TK]D-Fender | bingnet922, No it isn't |
| 01:17.28 | bingnet922 | [TK]D-Fender: oh? Do tell, please. |
| 01:17.32 | [TK]D-Fender | bingnet922, You can use * to interface with whatever form of connectivity it supports and you want. |
| 01:17.55 | [TK]D-Fender | bingnet922, thats like saying that just because I have a shopping cart, that I have to put GROCERIES in it. |
| 01:18.30 | [TK]D-Fender | bingnet922, * supports many VoIP protocols, various hardware interfaces, etc. You can do whatever you want with it. |
| 01:19.03 | bingnet922 | [TK]D-Fender: in the event I require a real-world telephone number to terminate at my * gateway, and I take the advice I have read to use the IAX protocol rather than SIP, etc... then it is necessary to have PSTN -> IAX termination service? |
| 01:19.05 | [TK]D-Fender | bingnet922, I could use it as a CRON replcaement if I felt particularly masochistic. |
| 01:19.27 | [TK]D-Fender | bingnet922, SIP is typically advisable over IAX for quality and stability reasons. |
| 01:19.59 | bingnet922 | [TK]D-Fender: in which case I would be required to obtain PSTN -> SIP termination? |
| 01:20.06 | [TK]D-Fender | bingnet922, And it depends how you want to acquire that "number". Considered using a real land-line? POTS? PRI? ITSP? |
| 01:20.20 | [TK]D-Fender | bingnet922, You require it if its your intention to use it. |
| 01:21.17 | [TK]D-Fender | bingnet922, if you want to drive a car, of course you'll need a car to satisfy that want. Careful how you reverse all of your questions. |
| 01:21.25 | bingnet922 | [TK]D-Fender: Well, I am trying to understand the options, but the need is very basic: to route a call from what I assume to be the "real world" of PSTN telephones to my own little "virtual" world of *. |
| 01:21.30 | [TK]D-Fender | bingnet922, You are setting up the answer just in the way you ask. |
| 01:21.55 | [TK]D-Fender | bingnet922, well is VoIP how you want to get a PSTN number to arrive to * for processing? |
| 01:22.40 | maqr | do you guys think there's any security issue from loading all these extra modules that i probably won't ever use? |
| 01:23.27 | [TK]D-Fender | maqr, I would suggest unloading channel drivers you don't intend on using. Dialplan app modules, etc, are fine. |
| 01:23.27 | bingnet922 | [TK]D-Fender: Ah, so I could alternately use existing analog or channelized T1 telephone lines as points of termination with asterisk, but in the absence of that possibility I would need VoIP transport? |
| 01:23.34 | maqr | [TK]D-Fender: ok, ty |
| 01:23.57 | [TK]D-Fender | bingnet922, thats largely whats left once you take traditional PSTN connectivity out of the picture, yes. |
| 01:24.23 | [TK]D-Fender | maqr, basically only disable stuff that if it had a bug really opens an avenue for attack. |
| 01:24.39 | bingnet922 | [TK]D-Fender: And in that scenario where I decide to use VoIP transport to my * gateway, what determines the maximum number of concurrent incoming calls? |
| 01:24.55 | [TK]D-Fender | commandeers Qwell's chan_skinny bot-net for great justice... |
| 01:25.18 | hsv-al | jeez |
| 01:25.25 | hsv-al | fender is still plugging away here |
| 01:25.26 | hsv-al | heh |
| 01:25.28 | [TK]D-Fender | bingnet922, bandwidth, codec conversion load (if applicable), and what your provider agrees to allow you. |
| 01:26.27 | [TK]D-Fender | bingnet922, Depending on your need, chances are there's someone out there who'll offer you the service. Its just a question of cost-effectiveness, quality, etc. |
| 01:26.39 | maqr | [TK]D-Fender: i was wondering what that chan_skinny one was for, something about cisco? |
| 01:26.40 | bingnet922 | [TK]D-Fender: OK, IP Communications offers choices of the number of telephone lines and the number of "ports". I am trying to understand the relationship between those options and the number of concurrent calls. |
| 01:26.42 | [TK]D-Fender | bingnet922, key term with VoIP is "YMMV" |
| 01:28.15 | [TK]D-Fender | bingnet922, ports typically refers to simultaneous channels. DID's are PSTN inbound #'s. You can have a DID (#) for which your provider will allow you up to X simultaneous calls for, etc. You can have providers who allow any number of outgoing calls where you can set the # you want to appear as. |
| 01:29.14 | [TK]D-Fender | bingnet922, And of course you'll see plans that look like the VoIP equivalent of a standard analog line. max of 2 calls, single DID associated, appear only as the DID associated with your inbound, etc. |
| 01:29.20 | [TK]D-Fender | ~itsp |
| 01:29.20 | jbot | [~itsp] An ITSP is an Internet telephony Service Provider or "VoIP Telephone company". The allow you to either PLACE calls to the PSTN (called Termination), RECEIVE calls from the PSTN (called Origination), or both. Some offer fixed rates, others $/min. Enter ~itsplist-us (USA) or ~itsplist-ca (Canada) for a listing of popular ITSPs |
| 01:30.11 | bingnet922 | whoa, instant help. |
| 01:31.55 | bingnet922 | [TK]D-Fender: Thanks. One more, what is the piece of hardware called that would allow me to connect an analog phone line to *? Would it be a PCI card sold by Digium, Inc? |
| 01:32.30 | tzanger | [TK]D-Fender: you need to be paid by digium for being their unofficial level one support. you are *always* Here helping out |
| 01:33.22 | [TK]D-Fender | bingnet922, Yes, Digium makes all sort of cards for this. There are other gateway devices that will take in your "line" and IT will speak VoIP to * (usually you might do this on a local LAN), etc |
| 01:33.33 | [TK]D-Fender | tzanger, That'd be nice :) |
| 01:34.50 | coppice | tzanger: I think he just needs a girl :-) |
| 01:35.15 | [TK]D-Fender | coppice, got a girlfriend, thanks :) |
| 01:35.21 | tzanger | heh |
| 01:35.52 | coppice | then why do you spend so much time here? Is she *that* ugly? |
| 01:36.06 | tzanger | coughs |
| 01:36.07 | tzanger | hahaha |
| 01:36.15 | [TK]D-Fender | coppice, She works in a call center with WAY too many night shifts..... |
| 01:37.04 | maqr | oh, burnnn |
| 01:37.08 | [TK]D-Fender | coppice, its 9:30pm here and she finishes in an hour. sucks sometimes, and its not just on weekdays either. Stupid screwy random schedule |
| 01:37.16 | maqr | [TK]D-Fender: did you impress her with your phone knowledges? :p |
| 01:37.57 | [TK]D-Fender | maqr, she loves tt-weasels :) But she's much more impressed with me on the guitar :) |
| 01:38.00 | coppice | oh, if she sucks she can' be all bad |
| 01:38.31 | [TK]D-Fender | coppice, "Life sucks, but rarely swallows" |
| 01:38.34 | *** part/#asterisk bingnet922 (n=ken@70.99.220.206) |
| 01:39.46 | maqr | rofl |
| 01:40.08 | coppice | I made a deep impression on my wife the first time I sung to her in Cantonese, and I sing really really badly. Your guitar comment means nothing :-) |
| 01:40.16 | maqr | [TK]D-Fender: did she know you were a phreak before she was your gf? |
| 01:40.18 | maqr | stops the bad puns |
| 01:40.24 | maqr | last one, promise |
| 01:40.24 | maqr | lol |
| 01:40.34 | *** join/#asterisk moy (n=moyhu@189.169.69.205) |
| 01:40.53 | [TK]D-Fender | coppice, Guess for you love is blind AND deaf. How fortunate! |
| 01:41.39 | coppice | "love is deaf" seems good when you have a mother in law suffering from verbal diahorrea |
| 01:41.54 | [TK]D-Fender | maqr, ahhh the good 'old 2600 days..... |
| 01:42.01 | maqr | lol |
| 01:42.03 | *** part/#asterisk unstable (i=unstable@tor/regular/sid) |
| 01:42.13 | coppice | the good old 2280 days, for those outside the US |
| 01:42.22 | [TK]D-Fender | coppice, funny you seem to be the one suffering from it. |
| 01:42.38 | coppice | I'm just having a bored 5 minutes |
| 01:42.39 | tzanger | I like "light is faster than sound. that is why some people appear bright until they speak" |
| 01:43.05 | seanbright | resents that |
| 01:43.06 | [TK]D-Fender | tzanger, very nice... think I'll recycle that one later |
| 01:43.24 | coppice | "better to remain silent, and be thought a fool, than to open your mouth and remove all doubt". |
| 01:43.36 | maqr | if i'm just using asterisk for SIP, i shouldnt' need chan_agent, iax2, mgcp, phone, or proxy, right? |
| 01:44.00 | [TK]D-Fender | maqr, agent if you need queues maybe, the rest can go. |
| 01:44.14 | *** join/#asterisk techie (n=techie@adsl-76-214-30-46.dsl.lsan03.sbcglobal.net) |
| 01:44.27 | tzanger | coppice: have I thanked you lately for sliptest.c? |
| 01:45.11 | maqr | [TK]D-Fender: 'queues' are how i'd queue up callers? like for support reps or something? |
| 01:45.14 | wwalker | 10:57 < srogers> sure - I expected that, but I didn't expect it to be so drastically out of whack |
| 01:45.35 | [TK]D-Fender | maqr, and FYI my phone knowledge pales to those of people like tzanger, coppice, Strom_M, and several others here. |
| 01:45.39 | wwalker | oops, IRC is hard (damn touchpad) |
| 01:46.03 | [TK]D-Fender | maqr, I am just relatively competant at *, and not experienced with all of it. |
| 01:46.17 | [TK]D-Fender | maqr, yup. |
| 01:46.26 | maqr | [TK]D-Fender: you've been yelling correct directions at me for like 3 days now, so i'm impressed anyway :) |
| 01:46.40 | maqr | "you're doing it wrong, go read a book" seems to be the correct solution to most of my problems |
| 01:46.55 | E-bola | [TK]D-Fender is a great asset for asterisk support, no doubt there :) |
| 01:47.21 | *** join/#asterisk hijacked (n=argh@cerebus.clandestineresearch.com) |
| 01:47.32 | [TK]D-Fender | maqr, if you lack the basics that the book was written for then I point you to the book. Come in with a specific little thing and can back up your situation and I tend to answer specific. |
| 01:48.19 | coppice | a great asset? put a sticker on him, and account him on a four year depreciation cycle |
| 01:49.54 | maqr | heh, very true |
| 02:04.18 | *** join/#asterisk s0lid (n=s0lid@58.69.2.239) |
| 02:10.22 | *** join/#asterisk [gquit]bombadil (n=dana@CPE-70-94-44-157.wi.res.rr.com) |
| 02:12.51 | maqr | is there anything wrong with using Goto? |
| 02:13.44 | [TK]D-Fender | maqr, Sorry, could you make you question a little more vague and open-ended? |
| 02:13.55 | maqr | [TK]D-Fender: how do i do the thing i'm trying to do? |
| 02:14.08 | *** join/#asterisk CaTtleyA (n=DDoS@bas1-montrealak-1128581558.dsl.bell.ca) |
| 02:14.17 | [TK]D-Fender | maqr, load res_psychic.so |
| 02:14.21 | maqr | touche |
| 02:15.03 | maqr | [TK]D-Fender: i'd like to construct a subroutine for dialing my extension, so that whether the call comes in and gets Dial()'d to me, or someone really dials my extension, they end up at the same place (which will be a follow-me, or perhaps just monkeys and voicemail) |
| 02:15.56 | [TK]D-Fender | maqr, you'd have to shwo your current implementation. The answer is variable based on exactly how you did things. |
| 02:16.24 | maqr | [TK]D-Fender: i don't really have an example yet, i'm trying to figure out how to write it... but i could easily do it with the same code in two or three places |
| 02:16.36 | maqr | [TK]D-Fender: rather than copy/paste, i'd typically write a function, but because this is a dialplan... maybe i should write a goto? |
| 02:17.16 | [TK]D-Fender | maqr, go read up on Macros |
| 02:21.11 | *** part/#asterisk war59312 (n=war59312@unaffiliated/war59312) |
| 02:21.42 | maqr | excellent |
| 02:22.04 | maqr | this is the thing i wanted to do |
| 02:22.30 | *** join/#asterisk drehlecom (i=ircbnc@unaffiliated/drehlecom) |
| 02:23.15 | *** join/#asterisk Koshatul (n=evangeli@ppp188-33.static.internode.on.net) |
| 02:28.04 | maqr | [TK]D-Fender: what's the right way to Set() myself a variable to be used inside of my macro? |
| 02:28.50 | [TK]D-Fender | maqr, there is no concept of scope in the dialplan. |
| 02:29.34 | [TK]D-Fender | maqr, All of the macro samples show you how you can use ${ARG1} , etc, to pass "parameters". Go get your hands dirty and see what happens |
| 02:32.09 | maqr | alright, thanks |
| 02:39.04 | drmessano | write it for me? |
| 02:39.29 | SteveTotaro | Qwell: |
| 02:39.32 | maqr | [TK]D-Fender: when they say "local channel variable" that should only ever apply to one "call" as it moves through the config, right? |
| 02:39.48 | Qwell | SteveTotaro: |
| 02:40.04 | SteveTotaro | word has it you are familiar with MGCP codebase |
| 02:40.10 | SteveTotaro | is that word correct? |
| 02:40.40 | [TK]D-Fender | maqr, they call them "channel variables" for a reason. ${X} in one call need not be the same as in another |
| 02:42.25 | *** join/#asterisk isamar (i=1000@voice.maxirede.net) |
| 02:42.29 | isamar | hi folks |
| 02:42.39 | SteveTotaro | Qwell: I planning on implementing chan_megaco and want to know if you have anything already to work off of |
| 02:43.47 | coppice | I'd like to know what the people behind MGCP were on at the time. it would be a huge hit in the right market :-) |
| 02:43.48 | SteveTotaro | i understand MGCP is very similar but not compatible |
| 02:44.51 | coppice | the last time I looked at chan_mgcp (which was quite a while ago) it was only a very elementary implementation of enough of the switch side of MGCP needed to solve its developer's needs |
| 02:45.17 | isamar | I have an issue here with "attended transfer " |
| 02:45.19 | SteveTotaro | i looked at it yesterday, it supports a few kinds of phones |
| 02:45.33 | SteveTotaro | one you have to hack the source to make it work |
| 02:45.57 | isamar | when I receive a call from the PSTN and Dial(SIP/blah|60|TrT) |
| 02:46.00 | coppice | but it is far from a complete MGCP, and its only the switch side |
| 02:46.23 | isamar | I can, at the SIP UA side make an attended transfer pressing ** |
| 02:46.32 | isamar | if it times out... |
| 02:46.41 | SteveTotaro | right, there is some silly $40 bounty for trunking to MGCP system |
| 02:46.47 | isamar | the transfer message is played to the PSTN's counterpart... |
| 02:46.56 | isamar | is that the right behaviour? |
| 02:47.27 | SteveTotaro | if they hit # they should get a transfer prompt |
| 02:47.29 | coppice | the only bounty worth a damn is made from chocolate and coconut |
| 02:47.57 | isamar | then, It should be a damn bug.. |
| 02:47.59 | [TK]D-Fender | coppice, Mounds don't |
| 02:48.00 | SteveTotaro | i was thinking a few grand but i would be happy to supply chocolate and coconut |
| 02:48.57 | SteveTotaro | i want three chan_, chan_nbx, chan_mgcp (complete) and chan_megaco |
| 02:49.12 | *** join/#asterisk apollonx (i=kit@193.19.189.38.STATIC.ISP.KZ) |
| 02:49.19 | coppice | what about chan_charlie? |
| 02:49.36 | SteveTotaro | i already have the proprietary NBX protocol broken |
| 02:49.46 | SteveTotaro | the others are open specs |
| 02:50.10 | isamar | apollonx: what is kz? |
| 02:50.14 | Qwell | SteveTotaro: that word is not correct |
| 02:50.30 | SteveTotaro | reverse engineered |
| 02:50.45 | Qwell | <SteveTotaro> is that word correct? |
| 02:51.05 | SteveTotaro | oh, guess my source was wrong |
| 02:51.13 | Qwell | as is often the case |
| 02:51.24 | SteveTotaro | time will tell |
| 02:51.30 | drmessano | Qwell is working on the Asterisk <> ICQ gateway code |
| 02:51.44 | drmessano | 1996 meet 2006.. 2006, 1996.. |
| 02:52.02 | Qwell | I have a 6 digit icq uin :p |
| 02:52.03 | SteveTotaro | 2006? you drinking? |
| 02:52.12 | Strom_C | Qwell: I do too, somewhere |
| 02:52.20 | Qwell | Strom_C: yeah.."somewhere" |
| 02:52.42 | SteveTotaro | so in the interest of expanding asterisk, who was working on MGCP? |
| 02:52.42 | Qwell | I managed to find mine one day, but really didn't care to remember it, heh |
| 02:52.48 | Strom_C | why the jizz won't this adtran ta608 respond to the craft port? |
| 02:52.55 | Corydon76-dig | Does anybody actually use ICQ anymore? |
| 02:53.06 | SteveTotaro | not since aol bought them |
| 02:53.06 | drmessano | Corydon76-dig: Asians |
| 02:53.08 | jbeez | because adtran is teh suck |
| 02:53.10 | Qwell | Corydon76-dig: sure |
| 02:53.15 | Qwell | it's AIM now |
| 02:53.16 | SteveTotaro | adtran rulez |
| 02:53.23 | Corydon76-dig | Yeah, AIM |
| 02:53.39 | drmessano | ICQ has a huge non-english userbase |
| 02:53.42 | Qwell | I still use my icq occasionally |
| 02:53.54 | SteveTotaro | i used my icq with subseven |
| 02:53.55 | outtolunc | still uses his old number from time to time |
| 02:54.14 | Strom_C | is it possible to disable the craft port on these things? |
| 02:54.28 | SteveTotaro | i don't know about that adtran unit, sorry |
| 02:54.36 | coppice | ICQ even has another Steve Underwood who can write Chinese :-) |
| 02:55.02 | drmessano | It's not going to matter much when XMPP takes over |
| 02:55.53 | coppice | will that happen before or after IPV6 takes over? :-) |
| 02:56.05 | drmessano | Far before |
| 02:56.13 | drmessano | It's well on it's way |
| 02:56.26 | SteveTotaro | so here is the deal, i want and will have chan_nbx and chan_megaco developed |
| 02:56.43 | coppice | well, I've been using XMPP for maybe 9 years, and I still don't see a massive uptake |
| 02:57.02 | SteveTotaro | wondering if Digium wants in? |
| 02:57.07 | drmessano | GTALK? |
| 02:57.09 | coppice | who many nbx users are there? I thought it didn't sell well |
| 02:57.23 | SteveTotaro | sells very well |
| 02:57.35 | SteveTotaro | v3000 and nbx |
| 02:57.40 | drmessano | Facebook chat will be XMPP based |
| 02:57.45 | drmessano | AIM is going XMPP |
| 02:57.48 | Qwell | and it'll steal your identity |
| 02:57.53 | Qwell | (again) |
| 02:58.08 | SteveTotaro | i have installed a hundred 3coms at least |
| 02:58.14 | coppice | everyone's going IPv6 too, but it never actually happens |
| 02:58.25 | drmessano | These things are actually happening |
| 02:58.55 | mackes | What do you all think of SER? |
| 02:59.07 | SteveTotaro | OpenSER has merits |
| 02:59.12 | drmessano | AIM XMPP exists |
| 02:59.15 | SteveTotaro | depends on application |
| 02:59.26 | drmessano | Facebook XMPP is withing a few weeks of being developed |
| 02:59.35 | SteveTotaro | load balancing and failover |
| 02:59.37 | drmessano | or fully developed, I should say |
| 02:59.40 | mackes | Does SER simply connect clients? |
| 02:59.54 | SteveTotaro | read jerjer's tutorials |
| 03:00.36 | mackes | Oh, ok- Were might I find those? |
| 03:01.04 | SteveTotaro | http://www.openser.org/docs/modules/1.2.x/dispatcher.html |
| 03:01.14 | mackes | and do you know of a Bible for openser? |
| 03:01.20 | SteveTotaro | if you google "openser howto" |
| 03:01.27 | mackes | Thanks for the link |
| 03:01.33 | SteveTotaro | jerjer's tutorial is right up top |
| 03:01.56 | SteveTotaro | follow his tutorial and then read about the dispatcher module |
| 03:02.36 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 03:02.57 | mackes | Neat, I will. Thanks |
| 03:04.27 | maqr | what variable would store the inbound callerd ID from my ITSP (sip)? |
| 03:04.38 | SteveTotaro | exten |
| 03:04.44 | Strom_C | no |
| 03:04.47 | Strom_C | CALLERID(num) |
| 03:04.51 | SteveTotaro | oh sorry to quick to reply |
| 03:04.58 | Strom_C | DOOF |
| 03:05.02 | SteveTotaro | what about name? |
| 03:05.07 | SteveTotaro | you left that out |
| 03:05.09 | Strom_C | CALLERID(name) |
| 03:05.33 | Strom_C | and dont forget the all-important CALLERID(burrito) |
| 03:05.47 | maqr | and that'd be in ${} because it's a channel variable, right? |
| 03:05.59 | Strom_C | actually, it's a function, but you can treat it like a variable |
| 03:06.24 | maqr | ${CALLERID(num)} though? since it's in the channel scope? |
| 03:06.36 | Strom_C | yes |
| 03:07.04 | SteveTotaro | isn't there a ${CALLERID(all)} |
| 03:07.15 | Strom_C | yeah, but it's harder to parse |
| 03:07.30 | SteveTotaro | but easy to farce |
| 03:07.49 | Strom_C | with data that's sparse |
| 03:07.54 | *** join/#asterisk anthm (n=anthm@CPE-72-131-113-50.wi.res.rr.com) |
| 03:09.39 | SteveTotaro | so i should check with BKW about chan_nbx and chan_megaco? |
| 03:09.45 | maqr | at what point does CALLERID() turn from an incoming thing into an outgoing thing? |
| 03:10.01 | maqr | since anything inbound should have a callerid set my by ITSP, but then it needs to send some caller id to my sip phone as well |
| 03:10.04 | maqr | when i Dial() it |
| 03:10.05 | SteveTotaro | depends on your dialplan |
| 03:10.26 | SteveTotaro | it should do that by default |
| 03:10.33 | maqr | oh wait, i'm retarded |
| 03:10.38 | maqr | ignore that |
| 03:10.38 | Strom_C | maqr: the caller ID is always the caller ID of the calling party |
| 03:10.44 | maqr | Strom_C: yeah, i just realized that |
| 03:10.52 | Strom_C | draw circle...bang head here |
| 03:10.52 | drmessano | Personally, i'd like to see asterisk development focused on making ASTERISK work, not "lets create a chan_everything" |
| 03:11.00 | SteveTotaro | not always, i have changed it many times |
| 03:11.06 | SteveTotaro | asterisk works |
| 03:11.21 | drmessano | Maybe some perfection of SIP and IAX2.. and less on chan_toshiba |
| 03:11.31 | SteveTotaro | i would like to eliminate IP based phone systems from having to integrate via T1 |
| 03:11.53 | [TK]D-Fender | drmessano, take your pick of the 3 other chan_sip "replacements" stalled in the works. |
| 03:12.04 | drmessano | I'd like them all to use Asterisk and be done with it |
| 03:12.22 | SteveTotaro | again, dreams are nice |
| 03:12.26 | SteveTotaro | reality is messy |
| 03:13.00 | SteveTotaro | sip works just fine for me |
| 03:13.10 | SteveTotaro | iax2 is problematic at best |
| 03:13.13 | drmessano | Focusing development time on integrating Asterisk with some other PBX's standards is counterproductive as shit |
| 03:13.16 | drmessano | Plain and simple |
| 03:13.28 | SteveTotaro | to you maybe |
| 03:13.37 | [TK]D-Fender | IAX2 is by and large a waste. |
| 03:13.37 | SteveTotaro | it aids in adoption |
| 03:13.39 | drmessano | Asterisk is Asterisk, not a plugin for every other PBX |
| 03:13.54 | SteveTotaro | asterisk definition? |
| 03:14.01 | SteveTotaro | why was it named asterisk?!? |
| 03:14.04 | drmessano | If you're gonna use Asterisk, use it.. |
| 03:14.09 | SteveTotaro | wilcard.... |
| 03:14.37 | SteveTotaro | wildcard, there is a reason why the cards were called that |
| 03:14.37 | mackes | So, I would like to create a hot failover server for my production Asterisk server- Many things I have read suggest a SER server infront of two Asterisk Servers. My clients are all SIP Polycom/ Asterisk and my PSTN access inbound/ Outbound is all SIP (Vitelity) |
| 03:15.06 | drmessano | "Asterisk is pretty cool.. I could make it work with my Cisco PBX" <-- or just, you know.. call me crazy, use Asterisk FOR your PBX |
| 03:15.19 | SteveTotaro | it does work with skinny |
| 03:15.23 | drmessano | I guess that's a silly idea |
| 03:15.31 | drmessano | Using asterisk AS a PBX |
| 03:15.38 | [TK]D-Fender | SteveTotaro, Where by Skinny we only support phones. |
| 03:15.47 | SteveTotaro | that's fine |
| 03:16.01 | SteveTotaro | i want to support 3com and NEC Dterm phones |
| 03:16.10 | drmessano | Asterisk support of Skinny is purely for those that want to use Cisco phones to impress |
| 03:16.19 | SteveTotaro | then flash to sip |
| 03:16.26 | drmessano | 3com and NEC phone's don't impress.. they're just part of the native systems |
| 03:16.27 | drmessano | FAIL |
| 03:16.30 | mackes | Is it better to use Cisco phones with SIP firmware or with Skinny |
| 03:16.37 | drmessano | Cisco phones are not feature rich with SIP |
| 03:16.38 | SteveTotaro | sip |
| 03:16.44 | drmessano | Negative |
| 03:16.57 | drmessano | SIP on Cisco phones is flaky and doesn't support all the capabilities |
| 03:17.00 | SteveTotaro | 3com and NEC phones do impress |
| 03:17.28 | SteveTotaro | if the SLA and BLF lamps could work in Asterisk |
| 03:17.38 | drmessano | I see lots of people clamoring for 3com phones.. all the time |
| 03:17.39 | SteveTotaro | it would be my phone of choice |
| 03:17.44 | drmessano | They ask for them by name |
| 03:17.45 | mackes | I have noticed that the Lamp does not work for Asterisk on the Cisco 7961/41 with SIP |
| 03:17.51 | [TK]D-Fender | * is missing a proper SIP stack supporting SIP-B, etc. |
| 03:17.52 | mackes | does it with Skinny |
| 03:18.13 | SteveTotaro | probably not working the helpdesk |
| 03:18.18 | drmessano | Cisco SIP firmware is feature lacking |
| 03:19.02 | SteveTotaro | but if you were an independant contractor selling phone systems you would |
| 03:19.07 | *** join/#asterisk jlgaddis (n=jlgaddis@fedora/jlgaddis) |
| 03:19.13 | *** join/#asterisk nighty^ (n=nighty@x122091.ppp.asahi-net.or.jp) |
| 03:19.16 | drmessano | you would what? |
| 03:19.35 | Strom_C | "you would learn to spell independent" is my guess |
| 03:19.37 | SteveTotaro | see people wanting 3com and asking for it by name |
| 03:19.43 | drmessano | HAH |
| 03:19.44 | drmessano | ok |
| 03:19.57 | SteveTotaro | spelling police can kiss it |
| 03:20.23 | drmessano | Newsflash.. most people don't know 3com makes phone systems or phones |
| 03:20.32 | SteveTotaro | ok guy |
| 03:20.33 | drmessano | They know the Cisco name.. even if in passing |
| 03:20.48 | SteveTotaro | guess you nevver worked for an interconnect |
| 03:20.49 | drmessano | or seeing it on a Linksys box |
| 03:21.03 | drmessano | I guess you have never worked with users |
| 03:21.15 | SteveTotaro | for many many years |
| 03:21.20 | drmessano | or business owners |
| 03:21.25 | drmessano | "customers" |
| 03:21.26 | SteveTotaro | for many many years |
| 03:21.32 | SteveTotaro | for many many years |
| 03:21.37 | [TK]D-Fender | 3com NBX is a top-tier hybrid PBX. |
| 03:21.38 | *** join/#asterisk Frogzoo (n=Frogzoo@124.184.33.9) |
| 03:21.42 | SteveTotaro | helpdesk, can i help you |
| 03:21.54 | SteveTotaro | i can't print |
| 03:22.03 | mackes | Are 3Com phones really that cool? |
| 03:22.03 | drmessano | Your delusion about the average person asking for 3com phones by name is somewhat amusing |
| 03:22.15 | SteveTotaro | average person no |
| 03:22.33 | SteveTotaro | but average business person buying a phone system, shopping around |
| 03:22.45 | SteveTotaro | you know, the guys that control the purse strings.... |
| 03:22.54 | maqr | what's the right way to say "If Busy (like my sip phone rejects the call), Do ____"? |
| 03:23.07 | mackes | Polycom Baby |
| 03:23.07 | drmessano | The average business person hires a phone contractor and asks them "Show me what you have".. |
| 03:23.13 | [TK]D-Fender | drmessano, the average person is a total moron. Managers call up telecom interconnectors saying "Hey we want a PBX", and whatever product they're pushing, thats what the manager will hear about. That and word of mouth from other companes |
| 03:23.17 | drmessano | The average business person doesn't "shop" for a PBX |
| 03:23.19 | Strom_C | maqr: do a GotoIf() based on DIALSTATUS |
| 03:23.20 | drmessano | Give me a break |
| 03:23.27 | SteveTotaro | if they are smart the shop around |
| 03:23.28 | drmessano | Do they shop for file servers too? |
| 03:23.35 | drmessano | "Oh, I want a DL385.. they're hot" |
| 03:23.38 | SteveTotaro | probably get four or five quotes |
| 03:23.46 | maqr | Strom_C: would letting it fall through to an 'i' extension be wrong? |
| 03:23.56 | Strom_C | maqr: that would be very wrong |
| 03:23.57 | jlgaddis | i like dl385s =) |
| 03:24.02 | Strom_C | you never want to let it fall through to anything |
| 03:24.20 | SteveTotaro | have you ever been in the market for a real PBX and not aware of asterisk? |
| 03:24.30 | SteveTotaro | then i guess you don't know how it works |
| 03:25.26 | Strom_C | all this arrogance feels great, guys! keep it up |
| 03:25.27 | drmessano | If I mentioned a 3com PBX to the average business owner, it would generate no more of a response than if I mentioned a Toshiba, NEC, or a lot of the others.. |
| 03:25.50 | SteveTotaro | doubtfully |
| 03:25.53 | drmessano | If they even remember 3com when there were relevent |
| 03:25.56 | SteveTotaro | have you tried it? |
| 03:26.25 | jlgaddis | we have a pbx that no one we talk to has ever heard of |
| 03:26.28 | drmessano | Actually, I have discussed phone systems with a lot of business folks |
| 03:26.45 | SteveTotaro | probably talking only about asterisk |
| 03:26.50 | maqr | Strom_C: what kind of variable is DIALSTATUS? i don't see that in the book |
| 03:26.50 | drmessano | Nope |
| 03:27.01 | SteveTotaro | helpdesk, can i help you |
| 03:27.05 | SteveTotaro | yeah, i can't print |
| 03:27.07 | drmessano | But your assumptions are amusing |
| 03:27.09 | SteveTotaro | did you reboot? |
| 03:27.15 | drmessano | What are you on about? |
| 03:27.20 | [TK]D-Fender | maqr, "core show application dial" |
| 03:27.27 | Strom_C | SteveTotaro: perhaps you should take a hint and shut up already |
| 03:27.39 | SteveTotaro | nope sorry |
| 03:27.40 | [TK]D-Fender | maqr, and stop right now and read EVERYTHING in you your source tarball's docs folder. |
| 03:27.52 | SteveTotaro | TK knows i am right |
| 03:28.06 | drmessano | Hmmm.. |
| 03:28.06 | SteveTotaro | DRM is speaking from inexperience |
| 03:28.20 | jlgaddis | you guys heard of tadiran? |
| 03:28.25 | drmessano | You have no clue what I am speaking about.. and I have yet to see you show much experience |
| 03:28.33 | SteveTotaro | i have heard of telrad |
| 03:28.43 | drmessano | Since you can't seem to make an argument based on fact, you're getting personal.. which is amusing |
| 03:29.11 | SteveTotaro | ok let me find marketshare data |
| 03:29.21 | SteveTotaro | nah, no need, i know what it is |
| 03:29.23 | maqr | [TK]D-Fender: oh, i didn't know that stuff was there, good call :) |
| 03:29.26 | drmessano | Of course you do |
| 03:29.26 | SteveTotaro | maybe you can find it |
| 03:29.32 | drmessano | You know it all |
| 03:29.35 | drmessano | Just ask you |
| 03:29.49 | SteveTotaro | yup |
| 03:29.59 | maqr | [TK]D-Fender: this is extremely useful |
| 03:30.51 | *** join/#asterisk JenniferAkemi (n=akemi@76-10-182-237.dsl.teksavvy.com) |
| 03:31.17 | [TK]D-Fender | drmessano, I've had to LOOK for interconnects, and I was contracted by one to bring THEM up to speed on VoIP & *. |
| 03:31.59 | [TK]D-Fender | drmessano, these are the places companies go when they say "we need a phone system" and are barely even comprehending what it means let along the options out there. |
| 03:32.10 | drmessano | Indeed |
| 03:32.16 | [TK]D-Fender | drmessano, You are being very near sighted. |
| 03:32.20 | drmessano | How so? |
| 03:32.46 | drmessano | By saying that the average business owner has no idea what PBX brands exist, the quality, etc |
| 03:32.49 | [TK]D-Fender | drmessano, So yeah, 3com IS a big name actually, as are NEC toshiba and the rest. I've seen so many different systems out there its a little scary. |
| 03:32.50 | SteveTotaro | Cisco leads the IP phone market, with 39% unit market share; the next closest competitors are 3Com and NEC, who are tied for 2nd |
| 03:33.13 | [TK]D-Fender | SteveTotaro, Cisco took #1? IIRC 3com held that for some time. |
| 03:33.16 | drmessano | Ask Joe Businessman if he's ever heard of a 3com PBX |
| 03:33.17 | SteveTotaro | i mean, just google it |
| 03:33.42 | SteveTotaro | ask joe businessman that has been in the purchasing cycle... |
| 03:33.58 | [TK]D-Fender | drmessano, My head office was looking at InterTel. talk about "less than popular". |
| 03:34.14 | [TK]D-Fender | then of course there's Avaya..... which too many businesses know of... |
| 03:35.08 | drmessano | In case you haven't realized this, Joe Businessman PAYS SOMEONE ELSE to tell him what is going to fit his need and to tell him what exactly his own needs are, the things he doesnt know exist.. Joe Businessman doesn't say "I want a 3com IP PBX, get me Ed's Telephone on the phone" |
| 03:35.38 | [TK]D-Fender | drmessano, like I said, they call an interconnector who pimps whatever flots his boat and makes him money. |
| 03:35.44 | [TK]D-Fender | floats* |
| 03:35.47 | jlgaddis | anyone have a recommendation for a voip provider that works well w/ asterisk? |
| 03:35.56 | SteveTotaro | NEC is more profitable than 3com |
| 03:36.05 | drmessano | Joe Businessman is concerned with building houses, or mining, or selling penis pumps.. not whether not he's got a 3com PBX |
| 03:36.10 | SteveTotaro | vitelity works well with asterisk |
| 03:36.31 | SteveTotaro | then he is not taking the buying cycle seriously |
| 03:36.40 | SteveTotaro | my customers shop around |
| 03:37.31 | SteveTotaro | anyways, i cited the market share and why having chan_X to support them would help asterisk's uptake |
| 03:37.38 | drmessano | All your customers are up to speed on IP PBX features and protocols? Wow.. you work with a very special class of customer |
| 03:37.53 | SteveTotaro | they are because i educate them |
| 03:38.02 | drmessano | Most I know focus on making money in their core business, not learning PBX tech for a once-in-ten-year purchase |
| 03:38.11 | SteveTotaro | it is part of my sales cycle |
| 03:38.17 | drmessano | They normally PAY someone to do that for them |
| 03:38.26 | SteveTotaro | yeah, they pay me |
| 03:38.36 | SteveTotaro | phone system is the core of most businesses |
| 03:38.52 | SteveTotaro | EOL |
| 03:38.55 | drmessano | [23:37] <SteveTotaro> they are because i educate them <-- Oh, I thought they didn't need educating? THat they were well aware of the product out there, because they are involved in the buying cycle |
| 03:39.10 | drmessano | That they ask for 3com by name |
| 03:39.18 | drmessano | Not needing to be "educated" |
| 03:39.24 | SteveTotaro | EOL |
| 03:39.34 | SteveTotaro | /dev/null |
| 03:47.50 | NovceGuru | so i've been getting mixed reviews on the asterisk appliance |
| 03:52.12 | [TK]D-Fender | NovceGuru, Its a toaster. |
| 03:52.56 | NovceGuru | so it runs NetBSD? :P |
| 03:57.00 | maqr | "WARNING[20041]: file.c:682 ast_readaudio_callback: Failed to write frame" <-- what would you take this to mean? |
| 03:58.09 | *** join/#asterisk techie (n=techie@adsl-76-214-30-46.dsl.lsan03.sbcglobal.net) |
| 03:58.54 | *** join/#asterisk Kalianyia (n=firewnyd@c-68-35-189-103.hsd1.nm.comcast.net) |
| 04:00.25 | maqr | oh, nevermind, my fault |
| 04:01.47 | maqr | wait, no, not my fault |
| 04:01.50 | maqr | it just does that and dies |
| 04:01.53 | maqr | that's not very good at all |
| 04:06.34 | *** join/#asterisk stkn__ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 04:08.20 | maqr | it was those damn monkeys |
| 04:23.52 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 04:32.16 | sack | hi guys , i update to 1.4.20 and 1.4.20.1 and i stated to have zombies processes for agi , someone else notice about = |
| 04:32.33 | sack | 1.4.19 works perfect though |
| 04:36.18 | sack | working with SIP , so could be a thread lock issue in chan_sip |
| 04:45.14 | *** join/#asterisk watchy (n=watchy@h200.176.255.206.cable.cmdn.cablelynx.com) |
| 04:45.25 | watchy | man i finnaly got XM radio to stream from the command line |
| 04:45.53 | watchy | now i gues to get it to stream to a mp3 |
| 04:50.35 | watchy | hrm |
| 04:50.49 | watchy | i think after i get it streaming |
| 04:51.08 | watchy | you think it would be possible to make it so individuals on hold could change channels? |
| 04:51.15 | watchy | incase they hate the on hold music? |
| 04:51.51 | brookshire | watch, of course it's possible, but probably load intensive.. how many lines are you looking at having? |
| 04:51.58 | watchy | like 3 |
| 04:52.08 | brookshire | yeah.. shouldn't be a big deal |
| 04:52.30 | watchy | i think it would be an intresting feature |
| 04:52.45 | watchy | if not just to do it to learn something new |
| 04:54.40 | watchy | im still trying to figure out how to pipe this to * |
| 04:59.00 | watchy | i wonder if i could do it by playing it to a audio cards output |
| 04:59.11 | watchy | then bringing it abck in through loopback in the input |
| 04:59.18 | watchy | but that would only give me 1 stream |
| 05:03.28 | TJNII | streaming moh isn't a new idea.... see the docs. |
| 05:04.34 | watchy | yea i know |
| 05:04.42 | watchy | but theres nothing about streaming XM |
| 05:05.24 | TJNII | No, but it does tell you exactly what format it expects sound in |
| 05:05.36 | TJNII | So you an pipe it through, say, sox |
| 05:05.43 | watchy | yea thats what i'm looking at now |
| 05:05.55 | watchy | trying to get mplayer to output in the right format |
| 05:06.19 | watchy | awe yea |
| 05:06.32 | watchy | looks like someone put it in the wiki about mplayer |
| 05:07.41 | TJNII | Oh, I'm sure it's in the mplayer man page too. Pity it's what, 250 pages long? |
| 05:07.45 | TJNII | I hate that damn man page |
| 05:07.49 | watchy | haha |
| 05:07.52 | watchy | im reading it now |
| 05:08.07 | TJNII | It needs to be about 15 man pages, not 1 |
| 05:08.08 | watchy | what took me forever to find was the servers of XM |
| 05:08.23 | watchy | since it requires a l/p |
| 05:13.41 | sack | ummm after a lot of testing no way to use 1.4.20.x with agi :-( |
| 05:13.51 | drmessano | Most man pages were written by folks selling the use of their software, not "instructing" about it |
| 05:14.16 | drmessano | "This thing has 217,952 functions" |
| 05:14.22 | drmessano | "Name 1..." |
| 05:14.26 | watchy | haha |
| 05:14.32 | drmessano | "Uh, why just one, when you get SOO MANY MORE!" |
| 05:14.41 | drmessano | 3 words |
| 05:14.45 | drmessano | Used Car Salesmen |
| 05:15.23 | drmessano | "So, how can I use WGET to download a file?" |
| 05:15.51 | drmessano | "Hey, did you know you can spoof user-agent with WGET??!! and.. and.." |
| 05:16.03 | drmessano | "Ok, this thing is for download files, right?" |
| 05:16.10 | watchy | hahahaha |
| 05:16.11 | sack | drehlecom, man wget ? ... hey ... who is that man ?! |
| 05:16.32 | drmessano | lol |
| 05:16.34 | sack | ups ... drmessano :-) |
| 05:17.10 | drmessano | I guess with anything, you can't know what a person is looking for in the manual.. |
| 05:17.34 | sack | funny ... still there's a lot of companies using http referer protection .... [offtopic] |
| 05:17.36 | drmessano | However, going over the stuff YOU consider EASY would be a good start.. Since that's 90% of the questions |
| 05:18.04 | sack | BIG COMPANIES |
| 05:18.10 | sack | O:-) |
| 05:18.43 | drmessano | There's still a lot of big companies using 16 bit software to store valuable parts of your data, unencrypted... BIG ONES |
| 05:18.48 | drmessano | Scared yet? lol |
| 05:18.55 | sack | yah ! |
| 05:19.20 | *** join/#asterisk murdock_ut (n=chatzill@c-98-202-154-226.hsd1.ut.comcast.net) |
| 05:19.54 | drmessano | I find it comforting to know some big company that I do business with in some way has my info stored in an unstable, easily cracked app |
| 05:21.15 | sack | i don't want to keep doing offtopic here ... so --- anyone using 1.4.20.x with agi issues ? |
| 05:21.34 | sack | i'm really stucked there :-( |
| 05:22.42 | sack | i followed svn changes , so only i can do is testeing 1.4.20-rcX |
| 05:22.47 | sack | _testing_ |
| 05:23.25 | sack | 1.4.20-rc1 looks good , but not others |
| 05:24.00 | drmessano | File a bug report |
| 05:24.37 | sack | drmessano, yes .... sure i will |
| 05:27.34 | sack | btw silly question how should i run asterisk to get a proper backtrace / usefull info for file a full |
| 05:27.48 | sack | s/full/bug/ |
| 05:28.00 | sack | LOL |
| 05:31.11 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 05:31.11 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.0-beta9 (2008/05/14) Asterisk 1.4.20.1, 1.2.28.1 (2008/05/21), *-Addons 1.4.6 (2008/02/21) 1.6.0-beta3 (2008/04/02), Zaptel 1.4.9.2 (2008/02/28), Libpri 1.4.3 (2007/12/13) -=- #asterisknow or #asterisk-gui for AsteriskNOW (asterisknow.org) -=- #switchvox for Switchvox (switchvox.com) -=- #freepbx for FreePBX/trixbox |
| 05:41.51 | watchy | is moh run for each person on hold? |
| 05:41.53 | watchy | or is it global |
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| 06:10.35 | disposable | can i use templates in extensions.conf? |
| 06:11.58 | phix | hii |
| 06:13.16 | disposable | this wasn't a trick question |
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| 06:46.33 | lmadsen | I know this is OT, but is anyone good with spreadsheet formulas? I have a spreadsheet with a column with the client name, and another column with hours worked. Each row is a task. I want to add up the hours for all tasks performed for that client in another cell. |
| 06:48.42 | Strom_C | are the hours all intermingled between different clients within the same column? |
| 06:49.19 | lmadsen | } Client | Task Description | Hrs Worked | |
| 06:49.43 | lmadsen | each row is a different client potentially |
| 06:50.10 | lmadsen | I want to sum all 'hrs worked' for each 'client' |
| 06:50.20 | lmadsen | or for a specific client |
| 06:51.00 | Strom_C | now, the simple question: why not do this in accounting software which is specialized for this very thing? |
| 06:51.15 | lmadsen | because I have all my hours tracked in a google spreadsheet |
| 06:51.24 | lmadsen | it's worked very well for 2 years now :) |
| 06:51.32 | lmadsen | I've just decided I wanted to do something complex now :) |
| 06:52.55 | Strom_C | Unless each entry for the client has the exact same string, I think you're better off keeping them in separate spreadsheets for each client |
| 06:53.56 | lmadsen | Each | Client| field has the exact same string to the case |
| 06:55.03 | lmadsen | ooo, I know how to do it now |
| 06:55.20 | *** join/#asterisk jivco (n=jivco@85.187.217.6) |
| 06:56.14 | lmadsen | =SUMIF(B7:B64,"=ClientName",H7:H64) |
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| 08:38.21 | sticko | anybody alive? |
| 08:38.28 | Strom_C | all dead here |
| 08:39.02 | sticko | I am a noob and have some questions about this asterisk pbx |
| 08:39.34 | sticko | I say I am a noob, I am a telecom technician new to asterisk |
| 08:39.46 | Strom_C | well, stop with the preamble and get on with the questions :) |
| 08:41.59 | sticko | I want to set up a voip switch to work with vonage. I also run an online business, and want my customers to be able to call in to an autoattendant and have the attendant fetch tracking numbers from a database and read those off to the customer. |
| 08:42.21 | sticko | I guess some sort of text to speech is necessary |
| 08:42.35 | Strom_M | why must it work with vonage? |
| 08:42.47 | Strom_M | and in what capacity do you expect it to "work with vonage:? |
| 08:43.21 | sticko | It doesn't really, I have been a vonage customer for years now and like thier service. |
| 08:43.41 | sticko | Supposedly asterisk can trunk with the softphone credentials |
| 08:43.53 | Strom_M | heh, so you want to use vonage as your ITSP |
| 08:45.30 | Strom_M | let me ask you this: in what capacity are you a telecom technician? |
| 08:45.35 | sticko | but its not set in stone, i guess if i could find a provider that could give me a 800-888 number and still be affordable, that would be good too. |
| 08:47.08 | sticko | I am nortel BMC 50 certified, telecenter ICS, telecenter VI certified as well |
| 08:47.39 | sticko | I do a lot of educational intercom/classrom phones for schools |
| 08:48.01 | Strom_M | ok, so your experience is pretty much all PBX installs |
| 08:48.10 | sticko | BCM 50 |
| 08:49.06 | Strom_M | vonage really blows as an ITSP. it's a consumer grade service. I would stay away from it. |
| 08:56.06 | Strom_M | so....do you have questions, like you said you did? |
| 08:57.25 | sticko | can the system use autoattendant to fetch data and text to speech back to the inbound caller? |
| 08:57.32 | Strom_M | easily |
| 08:58.12 | Strom_M | though unless you have a specific reason to use text-to-speech, I'd recommend using recordings of real human beings. I find text-to-speech to be a bit unnervingly artificial. |
| 08:59.39 | sticko | I just need it to read off tracking alphanumerics |
| 09:00.11 | Strom_M | asterisk has recordings of a very pleasant-sounding woman reading all those :) |
| 09:02.25 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 09:02.58 | sticko | being new to asterisk, should I be use asterisknow? I am not familiar with linux either. |
| 09:03.35 | gr0mit | sticko, my STRONG recommendation is to not use any web gui |
| 09:03.44 | gr0mit | it causes sooo much grief |
| 09:03.50 | *** join/#asterisk kai4711 (i=psybnc@h1395155.stratoserver.net) |
| 09:04.05 | gr0mit | much better to start with a VERY simple set of files |
| 09:04.25 | gr0mit | then you will get to understand the architecture of asterisk |
| 09:04.51 | gr0mit | once you understand the arctitecture, then by all means use a GUI if that;s what floats your boat |
| 09:05.38 | *** join/#asterisk oej (n=olle@ns.webway.se) |
| 09:05.45 | Strom_M | right, olle? |
| 09:06.07 | sticko | great |
| 09:07.34 | sticko | if the system is pure voip, I don't need any other hardware than network switch gear and phones right? |
| 09:08.08 | sticko | and the asterisk box |
| 09:08.10 | Strom_M | what do you mean "network switch gear"? |
| 09:08.56 | sticko | switches/routers |
| 09:09.36 | Strom_M | correct |
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| 10:35.05 | A500mg | hi |
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| 12:59.45 | SteveTotaro | i want to base a call center off a cluste of magicjacks and asterisk |
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| 13:02.12 | philip76 | question, im newbie with this card but i want to make sure if it will work, I have TDM400P with 4 FXO cards |
| 13:02.14 | philip76 | anyone |
| 13:02.39 | *** join/#asterisk mukudo (n=jgreen@58.251.97.17) |
| 13:03.04 | mukudo | Hi, has anyone tried CDRTool? I can't login to the CDR WebUI with admin/admin |
| 13:03.25 | mukudo | don't know how to troubleshoot it since there is so little documentation |
| 13:05.36 | *** join/#asterisk bootc (n=bootc@arcadia.prv.bootc.net) |
| 13:06.20 | bootc | hey folks |
| 13:06.32 | bootc | I'm trying to set up ldirector to do SIP failover with Asterisk |
| 13:06.47 | bootc | but I can't get Asterisk to authenticate its requests |
| 13:06.48 | bootc | http://pastebin.com/m25b003d7 |
| 13:06.57 | bootc | any clues on what should go in sip.conf? |
| 13:23.31 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 13:36.18 | SteveTotaro | <PROTECTED> |
| 13:37.48 | SteveTotaro | philip, you want to know if your tmd400p will work |
| 13:37.53 | SteveTotaro | why wouldn't it? |
| 13:42.48 | philip76 | yes SteveTotaro |
| 13:43.42 | philip76 | earlier i tried to install but when i tried to ztcfg âvv all the channels are FXS |
| 13:44.09 | philip76 | but I have the card TDM400P with 4FXO channels |
| 13:44.16 | philip76 | anyone |
| 13:46.57 | *** join/#asterisk oej_ (n=olle@ns.webway.se) |
| 13:47.12 | rob0 | Phil should start at ~book |
| 13:47.17 | rob0 | ~book |
| 13:47.18 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 13:47.56 | philip76 | ok thanks |
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| 13:55.04 | hsv-al | interesting |
| 13:55.07 | hsv-al | its quiet |
| 13:55.14 | hsv-al | d-fender isn't lecturing already :) |
| 13:57.44 | hsv-al | dont tell me im the only one up at 8:57am on a saturday morning, heh |
| 13:58.31 | tzafrir_home | philip76, all channels are "FXS" because you try to configure them as such |
| 13:59.16 | tzafrir_home | hmm...naturally, you have FXSKS signnaling, that's an FXO module |
| 13:59.37 | tzafrir_home | (yeah, why not confuse users if you get the chance) |
| 14:00.54 | philip76 | tzafrir_home>ok so if im going to use only 1port how should i define this and to have an outbound call, im newbie but im trying to google everything but i have no success. all of my extensions are working except for outbound using POTS or PSTN |
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| 14:31.03 | SteveTotaro | philip |
| 14:31.50 | SteveTotaro | you can use groups or you can define the specific interface in your dial statement |
| 14:31.52 | philip76 | yes SteveTotaro |
| 14:32.06 | philip76 | ok i'll try that |
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| 14:50.50 | hsv-al | im in shock |
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| 14:56.26 | seanbright | i'm in baltimore |
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| 15:14.30 | hsv-al | nice |
| 15:14.33 | hsv-al | d-fender is actually up |
| 15:14.40 | hsv-al | im dissapointed in you |
| 15:14.44 | hsv-al | you didnt get here earlier :) |
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| 15:34.02 | *** mode/#asterisk [+o lmadsen] by ChanServ |
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| 15:50.50 | maqr | if i have sln files, will asterisk automagically transcode them into ulaw/alaw/whatever as needed? |
| 15:51.30 | *** join/#asterisk excAliBuR (n=sales@207.134.8.33) |
| 15:53.31 | seanbright | yes |
| 15:54.33 | seanbright | it will try to pick the best format for the channel format, but resort to transcoding if necessary |
| 15:55.45 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 15:55.57 | excAliBuR | i'm trying to get my asterisk server to be seen from outside my network, so i set up a forward from port 5060 to my server's ip.. is that the correct port? |
| 15:56.30 | seanbright | ~sipnat |
| 15:56.30 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 15:56.43 | seanbright | excAliBuR: ^^^ |
| 16:03.37 | maqr | seanbright: so ideally, i should have my desired codec format and also slns in my sounds directory? |
| 16:04.29 | Qwell | maqr: exactly |
| 16:04.51 | drmessano | I just load them all up.. but that's idea if you're going to be sticking with a specific codec |
| 16:05.16 | maqr | ok, i can work with that |
| 16:05.27 | maqr | Qwell: does it also know how to transcode wavs? or just sln? |
| 16:05.44 | Qwell | wav too, if in the correct format |
| 16:06.05 | drmessano | "just sln" isn't accurate |
| 16:06.36 | drmessano | Asterisk transcodes a lot of formats.. sln is no different to wav than g711 or gsm is |
| 16:07.06 | maqr | oh, ok |
| 16:07.34 | drmessano | Don't confuse how the world has beat everyone into being so flippant about audio formats |
| 16:07.50 | excAliBuR | umm |
| 16:08.08 | excAliBuR | could someone here with a soft-phone help me for a few mins just to test my box ?? |
| 16:08.28 | seanbright | can i call home (china)? |
| 16:08.52 | drmessano | Can I call that dudes mom from last weekend? |
| 16:08.56 | excAliBuR | ip: 207.134.8.33 username: 2000 password: password <-- call extension 1000 :) |
| 16:09.14 | drmessano | or china |
| 16:09.15 | *** join/#asterisk ManxPower (n=manxpowe@162.sub-75-201-60.myvzw.com) |
| 16:09.16 | maqr | speaking of audio formats... is ulaw considered 'lossy'? |
| 16:09.24 | jblack | seanbright: If you have a provider, yes. I have seen china on many rate cards |
| 16:09.50 | seanbright | jblack: i meant while helping excAliBuR "test" |
| 16:09.58 | seanbright | isn't really from china |
| 16:10.01 | drmessano | maqr: That statement alone tells me you need to read up on formats |
| 16:10.07 | ManxPower | maqr: ulaw and alaw are not "lossy" from the standpoint of most telecom stuff |
| 16:10.08 | maqr | drmessano: agreed |
| 16:10.30 | maqr | ManxPower: that's what i got out of reading it, but it's clearly not the same quality output that i'm speaking into my microphone |
| 16:10.38 | ManxPower | maqr: correct. |
| 16:10.53 | ManxPower | in telecom "lossy" usually means "screws up DTMF and music" |
| 16:10.56 | drmessano | Your microphone is 100% uncompressed and lossless |
| 16:11.04 | maqr | drmessano: it's not even digital :) |
| 16:11.14 | drmessano | ugh |
| 16:11.15 | jblack | maqr: There's almost always a loss when converting between formats. |
| 16:11.47 | excAliBuR | i don't see user 2000 signing in :( |
| 16:11.53 | ManxPower | if you are an audio geek, then anything except uncompressed raw adio is "lossy" |
| 16:11.54 | drmessano | maqr: You can have the crappiest electret microphone ever and not have loss or compression |
| 16:12.05 | maqr | drmessano: yeah, i got that much |
| 16:12.54 | drmessano | maqr: You don't create loss or compression until you start modifying that audio after gets past the microphone.. so that's just obvious |
| 16:13.06 | maqr | ManxPower: by that definition, shouldn't G711 be capable of transmitting faxes? (i know this channel says it's a bad idea, i'm just curious) |
| 16:14.06 | Qwell | it's "capable", sure |
| 16:14.27 | excAliBuR | has anyone tried to register on my server? |
| 16:14.41 | drmessano | G711 isn't the issue |
| 16:15.06 | drmessano | If it were the case, G722 would fix it |
| 16:15.17 | drmessano | or make it suck less |
| 16:16.00 | maqr | so, what is the fax issue, exactly? |
| 16:16.25 | drmessano | VoIP is not meant to transmit data |
| 16:16.31 | drmessano | It's meant to transmit voice |
| 16:16.56 | drmessano | The protocols do a great job for what they were designed for.. |
| 16:17.07 | maqr | yeah, but phones transmit data, and i read that phone switches use G711 too... so, i must be missing something |
| 16:17.23 | drmessano | Voice over IP phones do not transmit data |
| 16:17.29 | excAliBuR | ok.. who in here has a softphone ? |
| 16:17.39 | drmessano | over their voice channel anyway |
| 16:17.49 | seanbright | excAliBuR: i just tried |
| 16:17.52 | excAliBuR | and? |
| 16:17.52 | seanbright | excAliBuR: timed out |
| 16:17.56 | excAliBuR | :( |
| 16:17.59 | excAliBuR | ummm ... |
| 16:18.00 | excAliBuR | ok |
| 16:18.06 | excAliBuR | maybe i need more ports open |
| 16:18.14 | seanbright | ~sipnat |
| 16:18.15 | jbot | [~sipnat] Quick guide on configuring * + SIP behind NAT : http://www.aocomputing.net/?p=3 , otherwise check the WIKI at http://www.voip-info.org/wiki/view/Asterisk+SIP+NAT+solutions |
| 16:18.20 | seanbright | excAliBuR: ^^^ |
| 16:18.25 | excAliBuR | i already read them |
| 16:18.28 | drmessano | Did you read seanbrights paste? |
| 16:18.29 | seanbright | deja vu much? |
| 16:18.31 | drmessano | What did you open? |
| 16:19.01 | drmessano | and what did you config? |
| 16:19.23 | excAliBuR | 5000-20000 goes to my server on both tcp/udp |
| 16:19.44 | seanbright | [sbright@elixer ~]$ telnet 207.134.8.33 5060 |
| 16:19.45 | seanbright | Trying 207.134.8.33... |
| 16:19.45 | Nugget | telnet is eeeeeeevil! |
| 16:19.45 | seanbright | telnet: connect to address 207.134.8.33: Connection refused |
| 16:19.55 | seanbright | no they don't :-) |
| 16:20.08 | drmessano | wow |
| 16:20.13 | drmessano | 5000 to 20000? WTF |
| 16:20.28 | excAliBuR | hmmm |
| 16:20.29 | drmessano | How about 5060 UDP and 10000-20000 UDP for starters |
| 16:20.30 | seanbright | that does seem like a lot. |
| 16:20.33 | seanbright | heh |
| 16:20.40 | maqr | drmessano: but g711 is from the 70s... and faxes are from the 80s... it's the same codec, isn't it? how can it not transmit that kind of data? |
| 16:20.55 | drmessano | Well |
| 16:21.09 | drmessano | I already said it's not G711 |
| 16:21.18 | drmessano | G711 is not the problem |
| 16:21.22 | maqr | yeah, but what *is* the problem? |
| 16:21.28 | maqr | the loss must happen somewhere to make it not possible |
| 16:21.35 | rob0 | Being TCP-only, a telnet to port 5060 won't connect to * |
| 16:21.50 | drmessano | Protocols, maqr |
| 16:22.09 | excAliBuR | ok i changed it to this 5060 20000 192.168.1.200 udp |
| 16:22.10 | seanbright | rob0: good point |
| 16:22.20 | excAliBuR | 5060-20000 |
| 16:22.21 | excAliBuR | :) |
| 16:22.26 | excAliBuR | to to be safe... |
| 16:22.27 | maqr | drmessano: have some protocol names i could look up? i'm still not sure i understand |
| 16:22.31 | excAliBuR | just to be safe++ |
| 16:22.48 | seanbright | still times out for me, might be my softphone |
| 16:22.53 | drmessano | maqr: try SIP and IAX |
| 16:23.11 | excAliBuR | can someone else try? |
| 16:23.15 | excAliBuR | ip: 207.134.8.33 username: 2000 password: password <-- call extension 1000 :) |
| 16:24.02 | excAliBuR | in my console i see this .. 1000/1000 192.168.1.131 D 56766 Unmonitored |
| 16:24.09 | excAliBuR | why port 56766? |
| 16:24.21 | excAliBuR | i don't have anything that high going to my server |
| 16:25.47 | drmessano | Thats the source port |
| 16:26.09 | excAliBuR | i'll put my server in the DMZ for now |
| 16:26.11 | seanbright | is registered now |
| 16:26.12 | drmessano | lol |
| 16:26.23 | drmessano | DMZ.. |
| 16:26.26 | seanbright | but can't dial :-/ |
| 16:26.29 | seanbright | heh |
| 16:26.31 | drmessano | Read the effin guides, excAliBuR |
| 16:26.37 | drmessano | DMZ is NOT and NEVER the answer |
| 16:26.58 | excAliBuR | for testing it is |
| 16:27.00 | Qwell | drmessano: what if the question is "What is the one thing I should never do?" |
| 16:27.06 | drmessano | lol |
| 16:27.25 | drmessano | excAliBuR: You are testing NOTHING.. in the end, you need a working SIP/NAT setup with proper port forwards |
| 16:27.31 | drmessano | You are doing NOTHING by putting it in a DMZ |
| 16:27.45 | drmessano | Qwell: Indeed |
| 16:28.06 | drmessano | As a matter of fact, putting your box in a DMZ is only going to screw with it |
| 16:28.22 | excAliBuR | so turn off dmz ?? |
| 16:28.27 | excAliBuR | never use dmz ?? |
| 16:29.11 | maqr | drmessano: oh, now i get it |
| 16:29.50 | maqr | drmessano: and T.38 would require a special provider which does PSTN<->T.38, right? |
| 16:30.18 | [TK]D-Fender | maqr, yes |
| 16:30.26 | excAliBuR | ummm Qwell i'm guessing u have a softphone.. could you try to register wit me? |
| 16:31.03 | [TK]D-Fender | maqr, And the problem isn't code, its that IP is PACKET based and packets get lost, delayed, etc. One little screwup and *BAM* carrier lost. |
| 16:31.14 | maqr | [TK]D-Fender: apparently RTP is a big part of the problem |
| 16:31.15 | [TK]D-Fender | excAliBuR, and READ THE GUIDE <------ |
| 16:31.17 | maqr | which makes sense |
| 16:31.24 | [TK]D-Fender | maqr, RTP = your voice packets. |
| 16:31.43 | excAliBuR | the guide won't help me test |
| 16:31.44 | [TK]D-Fender | maqr, And yes, all sorts of bad things can happen to them. |
| 16:32.01 | maqr | [TK]D-Fender: it's the same RTP that's use for things like RealAudio and Windows Media streams, right? |
| 16:32.07 | excAliBuR | it works on my internal network.. but i need to find out external if it will work |
| 16:32.28 | [TK]D-Fender | maqr, no idea what wrapper those 2 protocols use. |
| 16:32.44 | [TK]D-Fender | excAliBuR, go test with FWD then. |
| 16:35.25 | excAliBuR | i need a drink |
| 16:36.20 | excAliBuR | thanks anyways |
| 16:36.22 | *** part/#asterisk excAliBuR (n=sales@207.134.8.33) |
| 16:36.25 | seanbright | ? |
| 16:36.29 | seanbright | douchebag. |
| 16:36.38 | maqr | :o |
| 16:36.44 | jbeez | douche canoe |
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| 17:03.05 | za3toor | hey guys... i have been looking for the past 3 days on how to set my dnsmgr.conf file |
| 17:03.18 | za3toor | can anyone help |
| 17:03.30 | za3toor | a link would be great... thanks |
| 17:07.09 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 17:07.48 | maqr | if i built from source, what's the right way to get format_mp3? |
| 17:09.52 | maqr | oh, i guess i still need asterisk-addons |
| 17:15.49 | maqr | aha, success |
| 17:16.09 | maqr | i'm getting good at this asterisk thing :p |
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| 17:30.09 | WildPikachu | reads the ilbc license ... it seems to be a bit vague regarding binary distribution |
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| 18:36.34 | WildPikachu | wonders what happened between ilbc & asterisk |
| 18:36.51 | WildPikachu | asterisk is still listed on their site along with other opensource projects |
| 18:38.38 | drmessano | Nothing happened |
| 18:38.52 | drmessano | I believe it was a licensing issue |
| 18:39.35 | drmessano | You can still pull down iLbc and drop it in the asterisk source with a shell script that's included in Asterisk |
| 18:52.13 | *** join/#asterisk SteveTotaro (n=Administ@pool-151-196-246-236.balt.east.verizon.net) |
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| 19:00.17 | WildPikachu | i saw that drmessano |
| 19:00.30 | WildPikachu | was wondering why its listed on the ilbc site then as including ilbc |
| 19:00.51 | WildPikachu | also a thanks on the ilbc site that it was included a few years back |
| 19:00.54 | WildPikachu | sounds a bit weird |
| 19:01.05 | Qwell | WildPikachu: link? |
| 19:01.32 | WildPikachu | http://www.ilbcfreeware.org/software.html <= listed there |
| 19:01.48 | WildPikachu | thanks here => http://www.ilbcfreeware.org/news.html |
| 19:01.59 | WildPikachu | "Asterisk has introduced iLBC as a codec for its soft PBX. Big thanks to Mark Spencer (Digium) and Michael Haberler (Eunet)! " |
| 19:02.03 | WildPikachu | *shrug* |
| 19:02.31 | WildPikachu | maybe I misunderstand |
| 19:04.03 | mackes | What is better gsm or iLBC |
| 19:04.54 | WildPikachu | iLBC is the only codec which all of our phones support and our clients phones & pbx systems :( |
| 19:05.09 | WildPikachu | makes it hard to distribute asterisk with ilbc now to them for free |
| 19:05.19 | WildPikachu | i need to get a lawyer to look over the ilbc license |
| 19:08.29 | *** join/#asterisk techie (n=techie@adsl-76-214-30-46.dsl.lsan03.sbcglobal.net) |
| 19:08.51 | philip76 | has anyone here tried to configure outbound calls using freepbx |
| 19:09.07 | philip76 | need help i cant make it happen |
| 19:09.13 | philip76 | anyone |
| 19:09.52 | Juggie | ~freepbx |
| 19:09.53 | jbot | [~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 |
| 19:10.18 | philip76 | ok thanks |
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| 19:18.06 | drmessano | WildPikachu |
| 19:18.14 | WildPikachu | um? |
| 19:18.22 | drmessano | It has to do with DIGIUM DISTRIBUTING ILBC IN THE ASTERISK TARBALL |
| 19:18.25 | drmessano | Not USING ILBC |
| 19:18.31 | drmessano | How can that be any clearer? |
| 19:19.12 | drmessano | If there was a GAFFE, do you think it would be SO VERY EASY to install ilbc within asterisk right now? |
| 19:19.39 | WildPikachu | i read the commits and all it said was "licensing issue" |
| 19:19.45 | WildPikachu | would be nice to of known what issue |
| 19:19.47 | WildPikachu | thats all |
| 19:19.59 | mackes | ~pbxinaflash |
| 19:19.59 | jbot | [pbxinaflash] Ward Mundy's toy assembled by joe roper visit pbxinaflash.org or #pbxinaflash |
| 19:20.11 | drmessano | I would think it's wildly obvious |
| 19:20.32 | drmessano | and if you've any other open source software before, surely you have run into something like this |
| 19:20.51 | WildPikachu | pleaes enlighten me? |
| 19:20.54 | WildPikachu | *please |
| 19:21.03 | drmessano | Good god |
| 19:21.09 | drmessano | Ok, |
| 19:21.14 | drmessano | Joe has an app |
| 19:21.20 | drmessano | He uses the GPL license |
| 19:22.05 | drmessano | Steve has an app, he uses the "Lesser Don't Distruibte This With Any GPL Apps Because GPL Sucks" License |
| 19:22.15 | drmessano | Joe uses Steves app within his app |
| 19:22.32 | drmessano | But can't distribute it with his app because Steve has conflicting licensing |
| 19:22.49 | WildPikachu | ah |
| 19:22.49 | drmessano | So Joe adds a shell script to his app to download Steve's app on install so the dependency is there |
| 19:22.55 | *** join/#asterisk mackes (n=root@cpe-24-198-43-238.buffalo.res.rr.com) |
| 19:23.06 | drmessano | Now Joe's app works, and he's not violated Steves license |
| 19:23.35 | WildPikachu | so the issue was it being distributed under GPL |
| 19:23.40 | drmessano | No |
| 19:23.58 | drmessano | There was a licensing issue.. Who cares what the specifics were |
| 19:24.01 | drmessano | Fact is.. |
| 19:24.21 | drmessano | There wasn't a Gaffe, or else you wouldnt have a handy script in Asterisk to download and extract the sources |
| 19:24.29 | drmessano | So why worry about it |
| 19:24.55 | WildPikachu | I'd like to distribute binary ilbc modules and was curious about the specifics |
| 19:25.15 | drmessano | Then you need to read up on iLBC's licensing |
| 19:25.38 | drmessano | The conflict with Asterisks licensing is irrelevant in your case |
| 19:25.53 | WildPikachu | would of liked to know what it was :), thats all :) |
| 19:25.53 | drmessano | You need to be concerned with iLBC's distrubtion licensing |
| 19:26.02 | drmessano | distribution |
| 19:26.26 | drmessano | Go read iLBCs licensing |
| 19:26.31 | WildPikachu | already did |
| 19:27.26 | drmessano | Then you should have your answer |
| 19:30.06 | drmessano | Your rights to distrubution are very clearly spelled out in their licensing PDF |
| 19:30.31 | drmessano | Which again, regardless of the issue with Asterisk.. your situation applies to you |
| 19:33.27 | *** join/#asterisk emist (n=emist@unaffiliated/emist) |
| 19:35.22 | drmessano | Not trying to be difficult, but licensing based on someone else's conflict is dangerous.. and making assumptions based on someone else's licensing conflict are silly.. Best to read up on the licensing of the app in issue and figure out how it applies to you.. Most of the time you can do it without a lawyer :) |
| 19:36.06 | WildPikachu | agreed |
| 19:38.09 | drmessano | If iLBC was off limits for asterisk, trust me, the devs wouldn't make it so easy to pull the code down and compile.. I would expect more of a "iLBC is no longer supported. If you want to patch for it, whatever.. but we don't want to hear about it... iLBC is dead to us" approach.. |
| 19:38.24 | drmessano | and that's not the case |
| 19:38.47 | drmessano | But your distributing binaries may be an issue from the little I read of the iLBC license |
| 19:38.52 | drmessano | But it is all spelled out |
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| 19:40.37 | WildPikachu | well |
| 19:40.46 | WildPikachu | 3.1 says one must basically include the license |
| 19:40.56 | WildPikachu | but yea. .. this is offtopic here :) |
| 19:41.20 | drmessano | Yes and no.. |
| 19:41.58 | WildPikachu | and " a notice stating that the Source Code version of the Original Code is |
| 19:41.58 | WildPikachu | available under the terms of this License." |
| 19:42.08 | WildPikachu | plus satisfy 2.1 |
| 19:45.42 | WildPikachu | back to my ael :) |
| 19:52.21 | delparnel | hey all |
| 19:53.06 | WildPikachu | heya delparnel |
| 19:53.20 | delparnel | how goes |
| 19:54.29 | WildPikachu | i got one thing to say ... backup data :) |
| 19:54.44 | WildPikachu | our pabx crashed our offices in 3 continents :( |
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| 20:06.08 | *** join/#asterisk jbot (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 20:06.08 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.6.0-beta9 (2008/05/14) Asterisk 1.4.20.1, 1.2.28.1 (2008/05/21), *-Addons 1.4.6 (2008/02/21) 1.6.0-beta3 (2008/04/02), Zaptel 1.4.9.2 (2008/02/28), Libpri 1.4.3 (2007/12/13) -=- #asterisknow or #asterisk-gui for AsteriskNOW (asterisknow.org) -=- #switchvox for Switchvox (switchvox.com) -=- #freepbx for FreePBX/trixbox |
| 20:09.35 | *** join/#asterisk RoyK (n=roy@ip-106-28-149-91.dialup.ice.no) |
| 20:13.02 | WildPikachu | would Queue() return to the scope of the caller? |
| 20:13.07 | WildPikachu | i guess it wouldn't |
| 20:21.16 | *** join/#asterisk grandpapadot (n=no@adsl-074-185-089-046.sip.bhm.bellsouth.net) |
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| 20:22.04 | *** join/#asterisk JT (n=j@unaffiliated/jt) |
| 20:29.21 | maqr | this is so cool, asterisk actually makes sense to me now, i'm writing dial plans like a champ |
| 20:29.24 | maqr | thanks guys |
| 20:30.27 | WildPikachu | yea! |
| 20:30.35 | WildPikachu | t is timeout, what is i and o? |
| 20:31.30 | Strom_M | i is invalid |
| 20:31.40 | Strom_M | o is "operator" |
| 20:31.50 | Strom_M | look up "asterisk special extensions" |
| 20:31.58 | WildPikachu | ah, excellent |
| 20:32.04 | WildPikachu | googles on voip-info |
| 20:32.17 | WildPikachu | how would one get to the o? |
| 20:32.21 | WildPikachu | special key? |
| 20:32.29 | WildPikachu | reads |
| 20:33.11 | maqr | i totally knew that :) |
| 20:33.29 | maqr | WildPikachu: http://www.the-asterisk-book.com/unstable/besondere-extensions.html |
| 20:33.35 | maqr | i couldn't make much sense out of the pdf book |
| 20:33.37 | maqr | this one is way better |
| 20:34.31 | WildPikachu | ta |
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| 20:36.40 | *** join/#asterisk revengervn (n=test_tes@cpe-76-184-3-239.tx.res.rr.com) |
| 20:37.04 | *** join/#asterisk UngaMan (n=jvannini@dynamic53-150.MASAYA.cablenet.com.ni) |
| 20:37.16 | UngaMan | hello |
| 20:37.53 | revengervn | hi everyone |
| 20:37.54 | revengervn | I |
| 20:38.00 | revengervn | I'm having a problem |
| 20:38.04 | revengervn | with asterisk AGI |
| 20:38.12 | revengervn | can you guys help me out? |
| 20:38.37 | Strom_M | ~ask |
| 20:38.37 | jbot | ask is, like, Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 20:39.11 | revengervn | MP3Player('http://amber.streamguys.com:4290/listen.pls') |
| 20:39.18 | revengervn | it does not work |
| 20:39.30 | revengervn | can someone explain it to me? |
| 20:39.41 | UngaMan | hello |
| 20:40.25 | revengervn | hello |
| 20:40.31 | revengervn | can anybody help me out? |
| 20:40.39 | Strom_M | revengervn: jeez, calm down |
| 20:40.57 | revengervn | yes |
| 20:41.02 | Strom_M | i'm looking |
| 20:41.19 | revengervn | thanks |
| 20:41.44 | revengervn | it seems that MP3player does not work with some kinds of online streaming media |
| 20:41.49 | revengervn | is that true? |
| 20:42.03 | UngaMan | I am studying the option of using Asterisk for connecting several branches and a central office using VoIP over Internet... I would like to find an example or a guide that may help in my research |
| 20:42.10 | Strom_M | will you just shut up for thirty seconds so I can research your problem, revengervn? |
| 20:42.34 | UngaMan | at the digium site I found a case that says it could be done... but I need more information... |
| 20:42.46 | UngaMan | do you know where can I found that? |
| 20:42.49 | Strom_M | UngaMan: it's easy to do |
| 20:42.51 | UngaMan | thank you |
| 20:43.19 | Strom_M | UngaMan: but the answer is highly dependent on the specific needs of your installation |
| 20:43.22 | [TK]D-Fender | UngaMan, go lookup "asterisk dual servers" on the WIKI |
| 20:43.25 | [TK]D-Fender | ~wikis |
| 20:43.25 | jbot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 20:43.43 | [TK]D-Fender | UngaMan, and the BOOK. |
| 20:43.44 | [TK]D-Fender | ~book |
| 20:43.45 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 20:44.00 | Strom_M | revengervn: is the playlist you're linking to an MP3 stream? |
| 20:44.11 | revengervn | yes |
| 20:44.15 | [TK]D-Fender | UngaMan, Connecting * servers together isn't really any different than connecting a SIP phone to *. |
| 20:44.16 | revengervn | i want my asterisk |
| 20:44.22 | revengervn | to play streaming media |
| 20:44.25 | revengervn | like online radio |
| 20:44.27 | Strom_M | revengervn: try linking to the mp3 stream directly and not to the playlist |
| 20:44.31 | Strom_M | revengervn: yes, i know |
| 20:44.39 | Strom_M | you need to be patient |
| 20:44.46 | revengervn | i see |
| 20:44.51 | Strom_M | or you'll irritate me to the point where i don't want to help you. |
| 20:45.02 | revengervn | thanks in advance |
| 20:45.30 | UngaMan | Strom and Fender: thank you so much... will review those sites |
| 20:45.45 | UngaMan | it seems I have a lot to read... hehe as usual |
| 20:45.47 | UngaMan | :) |
| 20:51.15 | *** join/#asterisk asdx (n=diego@adsl-129-35.click.com.py) |
| 20:51.23 | asdx | hi, anyone knows a good h323 softphone? |
| 20:51.33 | maqr | is it possible to Goto a whole new context? |
| 20:52.23 | ManxPower | maqr: "core show application goto" |
| 20:52.53 | maqr | ohh, i see |
| 20:53.08 | ManxPower | "core show applications" will tell you what applications are available. |
| 20:53.17 | ManxPower | This is the first place you should look for application docs |
| 20:53.28 | asdx | my f******* telco is blocking sip/iax now |
| 20:53.50 | maqr | ManxPower: i didn't know about that, ty |
| 20:54.12 | ManxPower | Where "This" == "Asterisk CLI" |
| 20:54.28 | drmessano | Your telco is smart enough to block SIP and IAX? |
| 20:55.02 | drmessano | What are you basing this on? |
| 20:56.57 | revengervn | hello |
| 20:57.01 | revengervn | i have a question |
| 20:57.09 | asdx | drmessano: yes, they are blocking it |
| 20:57.20 | revengervn | can MP3Application plays other kind of extensions like '.rm' or |
| 20:57.24 | drmessano | asdx: How do you know this? What are you basing it on? |
| 20:57.27 | *** join/#asterisk atis_work (n=atis_wor@193.238.212.171) |
| 20:57.27 | revengervn | '.mov' |
| 20:57.32 | asdx | drmessano: i been using sip for 2 months, they blocked it before too |
| 20:57.44 | asdx | drmessano: then i tried iax and it did work, when sip was already blocked... now iax doesn't work too |
| 20:57.50 | asdx | drmessano: after my complains in forums, etc |
| 20:57.57 | drmessano | asdx: Who is your telco? |
| 20:58.53 | asdx | drmessano: http://www.copaco.com.py/ |
| 20:58.58 | asdx | drmessano: they have the reputation to be incompetent |
| 20:59.02 | Strom_M | revengervn: I'm pretty sure it's just for playing MP3 |
| 20:59.10 | asdx | drmessano: i know they are blocking it, i don't even have to do tests to prove it |
| 20:59.30 | drmessano | I find it unlikely that any telco is aware enough of IAX to block it |
| 20:59.30 | revengervn | thanks |
| 20:59.53 | asdx | drmessano: they probably looked at my posts in the forums |
| 21:00.21 | asdx | drmessano: i said "SIP doesn't work, IAX works" and a few days later, it wasn't working anymore |
| 21:00.31 | revengervn | so do you know what application can play .rm and .mov or how we can play that kind of streaming media in asterisk? |
| 21:01.12 | asdx | drmessano: how can i know if it's really blocked |
| 21:01.13 | Strom_M | why in god's name do you want to play movies over the telephone |
| 21:01.52 | drmessano | asdx: What do you mean BLOCKED? You are overusing overgeneral terms here |
| 21:01.57 | SteveTotaro | why does anyone want to do anything but talk on their phone? |
| 21:02.04 | drmessano | asdx: Can you register? |
| 21:02.11 | asdx | drmessano: NO |
| 21:02.12 | revengervn | I'm just asking |
| 21:02.14 | SteveTotaro | laptop is good for internet, chat, movies |
| 21:02.18 | asdx | drmessano: i can't |
| 21:02.21 | asdx | drmessano: i could 2 days ago |
| 21:02.23 | revengervn | yes |
| 21:03.10 | SteveTotaro | http://www.youtube.com/watch?v=td2Kvu1O3YE |
| 21:03.12 | revengervn | does anyone have any idea? |
| 21:03.30 | SteveTotaro | about what, i just got here |
| 21:04.06 | revengervn | do you know what application can play .rm and .mov or how we can play that kind of streaming media in asterisk? |
| 21:04.12 | revengervn | thanks so much |
| 21:04.34 | SteveTotaro | all i know for streaming is app_ices |
| 21:04.48 | SteveTotaro | but it is probably just limited to audio |
| 21:04.52 | Strom_M | revengervn: AFAIK there is no way to play real media or quicktime anything in asterisk |
| 21:05.14 | Strom_M | revengervn: figure out how to stream mp3 or play the music locally from the server |
| 21:05.36 | SteveTotaro | app_ices will allow you to stream audio |
| 21:06.32 | revengervn | thank you |
| 21:06.45 | revengervn | i'll try to figure about |
| 21:06.52 | drmessano | asdx: Is making internet phone calls illegal in Paraguay? |
| 21:06.53 | revengervn | i'll try to figure out |
| 21:07.05 | revengervn | thansk Strom_M and Steve for ur time |
| 21:07.17 | asdx | drmessano: yes |
| 21:07.25 | drmessano | Then there is your answer |
| 21:07.31 | drmessano | I won't help you break the law |
| 21:07.44 | asdx | drmessano: skype works though |
| 21:07.50 | SteveTotaro | i will give info on breaking the law |
| 21:07.58 | drmessano | "Works" and "Legal" are not the same |
| 21:07.59 | SteveTotaro | if you do it, it is your problem, not mine |
| 21:08.17 | asdx | drmessano: you are in the corruption side then |
| 21:08.38 | asdx | drmessano: don't help me, i will figure it myself |
| 21:08.38 | drmessano | asdx: No, its about right and wrong |
| 21:08.38 | SteveTotaro | making thermite is pretty fun |
| 21:08.51 | SteveTotaro | law is not about right and wrong |
| 21:08.52 | asdx | drmessano: THEY are wrong |
| 21:09.07 | asdx | they are a fucking corrupted monopoly |
| 21:09.10 | SteveTotaro | illegal does not = wrong |
| 21:09.17 | asdx | with full of corrupted idiots |
| 21:09.18 | SteveTotaro | it used to be illegal to help slaves escape |
| 21:09.31 | drmessano | asdx: There's a lot of things that do not make sense legally, but laws are set to be abided by, whether you like them or not |
| 21:09.46 | SteveTotaro | laws are meant to be broken |
| 21:10.02 | SteveTotaro | otherwise there would be no need for the law |
| 21:10.07 | SteveTotaro | in the first place |
| 21:10.23 | asdx | ok |
| 21:10.35 | SteveTotaro | change pots |
| 21:10.44 | SteveTotaro | sorry change ports |
| 21:10.48 | SteveTotaro | use openvpn |
| 21:10.55 | drmessano | asdx: If your country says VoIP is illegal, don't you think you'll get any sort of trouble if they're going to those lengths to enforce it? |
| 21:11.13 | SteveTotaro | nah, just use openvpn on a nonstandard port |
| 21:11.18 | SteveTotaro | they will never catch you |
| 21:11.38 | asdx | drmessano: sure, i will get into troubles, but at least i will be doing the right thing |
| 21:11.42 | maqr | wait, voip is illegal where? |
| 21:11.56 | drmessano | Paraguay |
| 21:11.57 | SteveTotaro | voip is illegal in many parts of the world |
| 21:11.58 | asdx | drmessano: there's no reason for a country to stop technology, just because 100 idiots wants to do more money |
| 21:12.12 | maqr | that's pretty ridiculous |
| 21:12.18 | SteveTotaro | you won't get caught unless you want to |
| 21:12.25 | maqr | good thing america isn't that bad (yet) |
| 21:12.51 | UngaMan | if I may |
| 21:12.57 | SteveTotaro | please |
| 21:13.00 | UngaMan | VoIP per se is not illegal |
| 21:13.17 | asdx | my country sucks |
| 21:13.18 | SteveTotaro | bypassing the pstn is illegal |
| 21:13.19 | UngaMan | but using a hidden port for making cheap international calls |
| 21:13.22 | UngaMan | is illegal |
| 21:13.23 | asdx | this is like dictatorship |
| 21:13.36 | SteveTotaro | no hidden port |
| 21:13.37 | drmessano | asdx: There's also no reason Marijuana should be illegal, since alcohol kills more people each year than weed does, doesn't mean I am going show my ass with civil disobedience to make a point.. Having an opinion is great. and a great way to get throw in jail by those same corrupt assholes that make it illegal in the first place |
| 21:13.42 | SteveTotaro | a non standard port |
| 21:13.47 | UngaMan | Steve: that is |
| 21:13.51 | SteveTotaro | and openvpn encryption |
| 21:13.53 | asdx | drmessano: yeah i guess |
| 21:14.11 | UngaMan | bypassing PSTN is illegal... |
| 21:14.15 | SteveTotaro | every year in |
| 21:14.26 | SteveTotaro | DC they have a "smokeout" |
| 21:14.36 | SteveTotaro | cops don't arrest for weed |
| 21:15.06 | UngaMan | unless u use VoIP for making private calls between offices of a same enterprise or bussiness in different cities or countries |
| 21:15.14 | asdx | drmessano: in that telco (copaco) there is full of idiots... really corrupted morons, they are really incomptent, we are stuck with <= 1mbps internet connection |
| 21:15.31 | SteveTotaro | they are not morons |
| 21:15.34 | *** join/#asterisk iamhrh (n=iamhrh@74.7.128.162) |
| 21:15.35 | asdx | they are |
| 21:15.39 | SteveTotaro | they are smart and pay off the government |
| 21:15.41 | drmessano | asdx: Trust me, I appreciate how stupid your situation is.. |
| 21:15.43 | asdx | i hope they will go away now, with the new government |
| 21:15.43 | SteveTotaro | to limit you |
| 21:15.54 | SteveTotaro | rebels |
| 21:16.00 | SteveTotaro | that is how the USA became free |
| 21:16.07 | SteveTotaro | pickup guns and start killing |
| 21:16.09 | asdx | they will go away |
| 21:16.30 | SteveTotaro | live free or die |
| 21:16.46 | asdx | they will go away because they are incompetent... be competent or die |
| 21:16.59 | SteveTotaro | pay the government and stay in power |
| 21:17.13 | SteveTotaro | sonatel in Senegal is the same deal |
| 21:17.34 | SteveTotaro | many countries have this same setup, phone company pays off the government |
| 21:18.01 | SteveTotaro | or is owned by the government |
| 21:18.18 | *** part/#asterisk iamhrh (n=iamhrh@74.7.128.162) |
| 21:18.38 | SteveTotaro | sonatel charged $3k U$D/mo for a voice E1 |
| 21:18.53 | SteveTotaro | when they found out it was for the US Embassy, they wanted more |
| 21:19.24 | SteveTotaro | when we said no, service was interrupted "unexplainably" |
| 21:19.39 | SteveTotaro | i know they just pulled the cross connect |
| 21:21.32 | asdx | we got new president, a ex bishop or something, he doesn't have anything to do with that telco shit |
| 21:21.49 | asdx | i think he will free the thing |
| 21:21.57 | asdx | hopefully |
| 21:24.40 | asdx | i will break the laws anyway |
| 21:24.41 | asdx | fuck it |
| 21:26.09 | revengervn | excuse me, can STREAM FILE cmd play MP3 or other types of extensions? |
| 21:27.13 | Strom_M | revengervn: you should read the documentation |
| 21:28.08 | asdx | SteveTotaro: was usa in some kind of dictatorship like this too? |
| 21:29.17 | revengervn | Strom_M: it does not mention about the extensions at all: http://www.voip-info.org/wiki/view/stream+file |
| 21:29.29 | Strom_M | revengervn: i said the documentation, not the wiki |
| 21:30.54 | asdx | i should gtfo from this country |
| 21:31.59 | revengervn | why dont you go elsewhere and curse your country |
| 21:34.29 | *** join/#asterisk aksyn (n=aksyn@78.86.127.226) |
| 21:34.38 | *** join/#asterisk disposable (i=disposab@blackhole.sk) |
| 21:39.02 | asdx | i'll try openvpn |
| 21:43.32 | asdx | bah, i wont do it |
| 21:43.57 | asdx | i don't want to go to jail, and there is probably someone from there here spying me |
| 21:44.20 | asdx | i will better do it in secret |
| 21:46.25 | [TK]D-Fender | asdx, and they NEVER scan IRC for people like you either ;) |
| 21:46.43 | [TK]D-Fender | asdx, can you her the sirens? Run! |
| 21:46.50 | revengervn | asdx, use Tor |
| 21:46.57 | revengervn | you can hide yourself |
| 21:47.03 | revengervn | by using TOR |
| 21:47.08 | revengervn | google it |
| 21:47.21 | asdx | ok |
| 21:47.21 | hsv-al | d-fender |
| 21:47.22 | revengervn | it's based on onion routing |
| 21:47.23 | asdx | [TK]D-Fender: heh |
| 21:47.23 | hsv-al | what days are you off? |
| 21:47.54 | asdx | can they see the encrypted data (openvpn) in the telco? |
| 21:47.57 | [TK]D-Fender | hsv-al, ones I'm not at home/work for. Thats rather few. I'm usually home at some time no matter what. |
| 21:48.08 | hsv-al | i was shocked this morning |
| 21:48.12 | hsv-al | i was in here at 8:45am |
| 21:48.16 | hsv-al | and you werent answering questions |
| 21:48.17 | hsv-al | :) |
| 21:51.08 | *** join/#asterisk emist_ (n=emist@unaffiliated/emist) |
| 21:52.52 | [TK]D-Fender | hsv-al, yeah I needed a nice full nights sleep... haven't been getting enough |
| 21:52.59 | florz | asdx: yes, it is possible to see the encrypted data, of course, and it is in principle possible to recognize it as (probably) encrypted - and in case of voip it should even be rather easy to see that the application you are using is probably voip, due to the large number of small packets |
| 21:53.54 | florz | asdx: ... the latter at least with the usual protocols, which do this for latency reasons, so there is pretty much no way around it, except for inflating the packets with random data |
| 21:55.05 | asdx | hmm i see |
| 21:55.17 | asdx | interesting... |
| 21:55.19 | florz | asdx: plus, make sure you don't use a vulnerable debian openssl for key generation, for both openvpn and tor |
| 21:56.00 | *** join/#asterisk JT (n=j@unaffiliated/jt) |
| 21:56.06 | asdx | ok, thanks |
| 21:57.34 | maqr | ok, i'm stuck... how exactly am i supposed to write a dial plan that answers anything that enters a specific context, plays a Background(), and waits for input? |
| 21:58.35 | [TK]D-Fender | maqr, reconsider your term "anything enters a context". |
| 21:59.26 | [TK]D-Fender | maqr, you don't "enter" a context. Some piece of dialplan sends you to a specific place, or a call comes in and lands on a pattern. Pick a patter that applies |
| 22:00.09 | maqr | hmm |
| 22:00.15 | watchy | i love u tk |
| 22:00.35 | watchy | tk: i think i have figured out how to stream XM to * MOH |
| 22:01.40 | watchy | but i really wanna make it so people on hold can change stations |
| 22:02.01 | [TK]D-Fender | watchy, you've got the source jsut like everyone else. |
| 22:02.26 | watchy | but i'm fat so i have a disability |
| 22:02.47 | maqr | rofl |
| 22:02.50 | maqr | best internet excuse ever |
| 22:03.11 | [TK]D-Fender | watchy, only if all that fat is crammed up in your skull.... |
| 22:03.13 | watchy | i have sausage fingers |
| 22:03.20 | *** join/#asterisk DarnoQ (n=d@chello089076192243.chello.pl) |
| 22:03.28 | maqr | mash the keypad to order a dialing wand |
| 22:04.00 | watchy | haha |
| 22:04.02 | *** join/#asterisk stkn (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 22:04.02 | [TK]D-Fender | watchy, people claiming to be fat shouldn't say they have "sausage fingers". That invites semi-cannibalisation. |
| 22:04.11 | [TK]D-Fender | self* |
| 22:04.30 | watchy | good thing i have no dipping sauce around |
| 22:04.49 | watchy | oh no ranch spilt on my fingers now i have to eat them |
| 22:05.52 | maqr | you definitely crossed the line with that one |
| 22:07.06 | watchy | im actually starving i just woke up |
| 22:07.42 | maqr | [TK]D-Fender: the confusing part is that sometimes the extension is what the user dials, and sometimes it's where the user came from |
| 22:08.00 | drmessano | "watchy fingers" sounds like the sort of food served at a strip club |
| 22:08.43 | maqr | patent pending |
| 22:10.07 | [TK]D-Fender | maqr, go get your head screwed on straight about exactly where the call is coming from. |
| 22:11.45 | maqr | [TK]D-Fender: well, there's only two ways the call can come in... either it's from my ITSP as an inbound call, or it's from my internal SIP phone |
| 22:12.49 | maqr | [TK]D-Fender: so say on my SIP phone, i dial 999, and it sends me to this new context... where i match again on 999 and play a Background... then i press '1' to go to voicemail |
| 22:13.16 | maqr | [TK]D-Fender: first i have to match 999 and goto, then i match 999 and play background, then i match 1 to do voicemail? |
| 22:14.48 | maqr | which actually seems to work |
| 22:14.56 | maqr | what a mindfuck, i guess i still don't quite get it |
| 22:15.51 | [TK]D-Fender | maqr, on your SIP phone is doesn't send you to a "new context". It matches in the context the device is TOLD to look in. |
| 22:16.19 | [TK]D-Fender | maqr, So why are you dialing an exten only to get an IVR prompting you to enter ANOTHER exten to go to? |
| 22:16.53 | maqr | [TK]D-Fender: i was just doing it to test my tree without having to keep calling from my cell phone |
| 22:17.05 | [TK]D-Fender | maqr, ok, fine. |
| 22:17.35 | maqr | that still might be completely silly to do |
| 22:19.58 | *** join/#asterisk harmagent (n=vector@host-90-199-9-69.midco.net) |
| 22:23.38 | maqr | success! |
| 22:24.04 | watchy | i think a big steak sounds tasty |
| 22:26.27 | harmagent | watchy you didn't used to hang out on undernet did you? |
| 22:26.43 | watchy | yes i was an undernet server admin for a few years |
| 22:27.03 | harmagent | did you know vector? |
| 22:27.12 | watchy | yes he was a sexy man |
| 22:30.16 | *** join/#asterisk riddlebox (n=james@75-132-225-75.dhcp.stls.mo.charter.com) |
| 22:41.16 | maqr | is there a way to make 'includes' not take last priority? |
| 22:41.24 | lmadsen | maqr: no, that is intentional |
| 22:41.51 | lmadsen | maqr: if you need to change matching order.... start in a context with only includes |
| 22:43.05 | maqr | hmm, ok |
| 22:47.06 | maqr | http://pastebin.ca/1028309 <-- why would this not play tt-monkeys twice? |
| 22:49.25 | Strom_M | I think there's an error in your gotoif syntax |
| 22:49.33 | Strom_M | replace the | with ? |
| 22:49.53 | maqr | oh, oops |
| 22:49.56 | maqr | i knew that :) |
| 22:52.03 | *** join/#asterisk s0lid (n=s0lid@58.69.3.199) |
| 22:53.19 | *** join/#asterisk adeel (n=adeel@c-24-7-132-23.hsd1.ca.comcast.net) |
| 23:06.04 | *** join/#asterisk eXistenZ (n=existenz@unaffiliated/existenz) |
| 23:09.13 | UngaMan | hello again |
| 23:09.31 | UngaMan | I've been reading the book's pdf |
| 23:10.16 | *** join/#asterisk jm|home (n=jm|home@zen.jamiem.com) |
| 23:10.16 | UngaMan | still looking info about connecting 2 * boxes via internet |
| 23:10.33 | UngaMan | the idea is to place 2 boxes in 2 different places... |
| 23:10.50 | UngaMan | each one managing a number of analog extensions |
| 23:11.27 | UngaMan | and the boxes must communicate via internet and not by PSTN |
| 23:12.43 | UngaMan | I found the example using IAX and SIP |
| 23:13.22 | UngaMan | would any of those setups work using Internet? |
| 23:13.59 | UngaMan | both boxes are visible to each other via the Net |
| 23:15.32 | mackes | Yep. Both will work |
| 23:15.53 | mackes | But the Internet is not a extremely dependable method. |
| 23:16.01 | mackes | Most of the time it will be fine |
| 23:16.23 | mackes | But, once in a while, you are going to have a poor call/ connection |
| 23:19.42 | [TK]D-Fender | UngaMan, SIP & IAX2 are VoIP protocols. Internet = IP |
| 23:23.16 | maqr | if my ITSP is showing 3215551234 for incoming callerid, is there some way i can auto-prefix that with a '1' for my country code? it'd make matching extensions a lot easier |
| 23:23.27 | UngaMan | maCKES: thanks :) |
| 23:24.15 | ManxPower | maqr: I think you need to read The Good Book |
| 23:24.16 | ManxPower | ~book |
| 23:24.17 | jbot | Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://tfot.leifmadsen.com, or see ~buybook |
| 23:24.29 | UngaMan | Fender: Iḿ aware of the poor quality somtimes the client will have... and that arises another question |
| 23:24.40 | ManxPower | maqr: www.voip-info.org also has many examples if dialplan stuff -- just remember sometimes the info is wrong |
| 23:24.47 | maqr | lol |
| 23:24.55 | UngaMan | how much bandwidth per line will be consumed? |
| 23:25.05 | *** join/#asterisk ikevin (n=kevin@kevin.linux-fr.net) |
| 23:25.13 | ManxPower | maqr: /path/to/src/asterisk/doc and /path/to/src/asterisk/configs is also a good place to look. |
| 23:25.21 | ManxPower | ~bandwidth |
| 23:25.21 | jbot | bandwidth is probably This is a measure, in some amount of bits per second, of theamount of data that can be sent over a particular cable, interface, orbus. |
| 23:25.26 | ManxPower | ~codecs |
| 23:25.27 | jbot | from memory, codecs is http://snipurl.com/wiki_codecs. If you have audio/codec problems, first try to 'disallow=all' and 'allow=ulaw' and see if that works. Anyone that tells you to use 'allow=all' is an idiot as it usually causes audio problems, or Number/Name: 1/g723, 2/gsm, 4/ulaw, 8/alaw, 16/g726, 32/adpcm, 64/slin, 128/lpc10, 256/g729, 512/speex, 1024/ilibc |
| 23:25.41 | ManxPower | hmm.. none of those are what I was looking for. |
| 23:25.58 | UngaMan | lol |
| 23:26.10 | UngaMan | but it says interesting tips |
| 23:26.27 | UngaMan | 8kb, 16kb? |
| 23:26.57 | ManxPower | UngaMan: it depends on the codec. You can assume about 16k of overhead over and above the codec bandwidth usage if you are using 20 ms audio packets |
| 23:28.35 | UngaMan | ohhh |
| 23:28.36 | UngaMan | ok |
| 23:29.21 | ManxPower | UngaMan: The answer to almost every question you will have about Asterisk will require you to know several things before the question can be answered. |
| 23:29.25 | maqr | ManxPower: hrm, i still don't get it though... my issue is that incoming calls don't use the country code prefix, even though they should... is that uncommon? |
| 23:29.56 | ManxPower | maqr: maqr: calls from what country to what country using what provider? |
| 23:30.18 | maqr | ManxPower: from the US, to the US, using vitelity |
| 23:30.36 | ManxPower | maqr: internal calls do not normally include the country code. |
| 23:30.39 | maqr | ManxPower: i'd like the numbers to include the '1' at the beginning, so i cna match them easier |
| 23:30.44 | maqr | *can |
| 23:31.04 | ManxPower | maqr: exten => _NXXNXXXXXX,1,Goto(1${EXTEN},1) |
| 23:31.14 | maqr | oh |
| 23:31.21 | maqr | that makes sense |
| 23:31.22 | maqr | heh |
| 23:31.28 | ManxPower | my systems to a fair amount of callerid massaging before the call gets to the end user phone |
| 23:31.40 | ManxPower | addind a 9, a 1, and - in the correct places. |
| 23:32.38 | maqr | yeah, that makes sense |
| 23:33.14 | ManxPower | none of those are really part of the callerid and some ip phones will reject it, but the phones we use accept the extra chars just fine. |
| 23:33.56 | UngaMan | ManxPower: don't worry about that... early today I stated that I'm researching about Asterisk. I actually work with Avaya Systems in my office... so perhaps I know a little about VoIP :p |
| 23:35.05 | ManxPower | UngaMan: then you should know this stuff already. Were you really asking about IAX2? |
| 23:35.28 | ManxPower | UngaMan: Avaya has been playing with Asterisk for several years. |
| 23:36.41 | *** join/#asterisk DagMoller (n=aguirre@unaffiliated/dagmoller) |
| 23:36.47 | UngaMan | yep... but they are not helping me much... I have an IPO ... and the manager is Win based |
| 23:37.13 | ManxPower | UngaMan: There is nothing different about SIP with Asterisk compared to SIP using Avaya |
| 23:37.22 | ManxPower | at least in the bandwidth used |
| 23:37.33 | UngaMan | ok... now that's helpful |
| 23:37.47 | UngaMan | thanks! |
| 23:37.59 | mackes | Vitelity is very good |
| 23:38.06 | ManxPower | IAX2 has a feature called trunking that can massively decrease the protocol overhead. |
| 23:38.22 | ManxPower | ~trunk |
| 23:38.23 | jbot | [trunk] a word with varying definitions. In Asterisk, a trunk is a "stream of UDP packets containing IAX2 frames from more than 1 call"; in telecom, a trunk is a "single voice channel between two pieces of switching equipment."; in Ethernet a trunk carries more than one 802.1q VLAN. There is no such thing as a "SIP Trunk" -- Don't use the term. |
| 23:39.07 | mackes | What do you all think of Avaya SIP phones? |
| 23:39.29 | ManxPower | ~phones |
| 23:39.30 | jbot | hmm... phones is http://bani.anime.net/phones/. While personal preference will dictate which phone works best for you, general consensus on a rough order of quality and suggestibility is as follows: Polycom (any), Aastra 480i, Aastra 5i Series, Cisco 7940+, Linksys SPA-9XX, Snom, and finally everything else. Do not consider Grandstream phones. Ever. places like such as |
| 23:39.41 | ManxPower | mackes: they are not even on our radar |
| 23:39.47 | mackes | ok |
| 23:40.41 | ManxPower | mackes: Many vendors that use SIP lock their phones and servers to only work with their own phones. Nortel is an example of this. I do NOT know of Avaya does this or not. |
| 23:41.25 | ManxPower | Personally I use Polycom phones. |
| 23:41.44 | mackes | Yeah, We do to, |
| 23:42.24 | mackes | I like to experiment. The Avaya 4600 looks sweet |
| 23:44.04 | UngaMan | that's another interesting topic... |
| 23:44.09 | UngaMan | to create a mix |
| 23:45.21 | *** join/#asterisk UnixDog (n=UnixDog@213.161.33.65.cfl.res.rr.com) |
| 23:46.35 | UnixDog | ok the new name for zaptel makes no sense it should have been renamed digi-zap |
| 23:46.41 | UnixDog | much esier |
| 23:48.36 | mackes | Now we need to get the Cisco IP Phone 7985G work with Asterisk |
| 23:48.37 | NovceGuru | anybody using rhino voip hardware? |
| 23:49.05 | UngaMan | Ok... I appreciate your help and comments |
| 23:49.11 | UngaMan | will come back later |
| 23:49.36 | UngaMan | and remeber... tomorro the Phoenix will touch down on Mars! |
| 23:49.43 | UngaMan | g'nite! |
| 23:49.48 | *** part/#asterisk UngaMan (n=jvannini@dynamic53-150.MASAYA.cablenet.com.ni) |
| 23:51.45 | NovceGuru | They are in my town but they seem to use trixbox :( |
| 23:52.03 | NovceGuru | nevermind they just sell hardware |
| 23:52.56 | tzafrir_home | UnixDog, anything with "zap" is probably too close to the problematic trademark (zaptel.com) |
| 23:53.48 | outtolunc | slaptel |
| 23:54.21 | tzafrir_home | UnixDog, think of it this way: nobody will want to claim the name DAHDI, and therefore future costly legal battles would be saved ;-) |
| 23:55.49 | UnixDog | ok ast-zap |
| 23:55.59 | UnixDog | but zap should not matter |
| 23:56.12 | UnixDog | I think all the name bullshit is crap |
| 23:56.20 | UnixDog | a name is a name is a name |
| 23:56.58 | UnixDog | thats like saying I have to change my name because some one else has the name Richard |
| 23:57.05 | UnixDog | its all bullshit |
| 23:57.22 | outtolunc | claimed it first.. better change yours |
| 23:57.38 | UnixDog | lol |
| 23:57.43 | outtolunc | i don't like dick you can use that one <G> |
| 23:57.56 | UnixDog | what ever |
| 23:58.45 | UnixDog | that means I need to change the bsd port |
| 23:58.53 | UnixDog | wich will take time |