| 00:00.44 | *** join/#asterisk JMAFOU (n=crypt@c-68-54-210-115.hsd1.in.comcast.net) |
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| 00:11.33 | \malex\ | is there a comparative review of modern versions of elastix, asterisknow and trixbox? i've been playing with them all, but i'd like a more experienced critique of each, if possible |
| 00:16.55 | *** join/#asterisk trnzmeta (n=bleh@secure27.lnk.telstra.net) |
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| 00:25.00 | joobie | hey guys.. anyone know of a good headset unit to get for a polycom 320? |
| 00:26.05 | MaliutaLap | don't plantronics do one? |
| 00:26.12 | trnzmeta | guys: when sip gateways start the phone call, does the rtp side of things still proxy through the PBX or is it client to client |
| 00:26.17 | MaliutaLap | plantronics are good headsets |
| 00:26.53 | adeel | trnzmeta, if you have the directrtp or canreinvite set to yes, then it's client to client |
| 00:28.03 | joobie | got a model MaliutaLap ? |
| 00:28.37 | MaliutaLap | joobie: no, I know the models for the Cisco 79XX models |
| 00:28.52 | joobie | what's that model? |
| 00:28.56 | joobie | might look into it? |
| 00:29.02 | juanIMP | [TK]D-Fender: are you busy, can I ask? |
| 00:29.04 | MaliutaLap | joobie: most sites with plantronics gear should have them list |
| 00:29.12 | MaliutaLap | joobie: google for it |
| 00:29.17 | joobie | yea there are just different models |
| 00:29.19 | joobie | a lot of models |
| 00:29.21 | joobie | trying to find a good one |
| 00:29.32 | trnzmeta | adeel: is directrtp an option in asterisk or is it in protocol space? |
| 00:30.14 | *** part/#asterisk tainted_ (n=Administ@67.43.165.100) |
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| 00:32.06 | [TK]D-Fender | joobie: Plantronics M22 + H261N |
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| 00:37.12 | pmhaddad | i have another dCAP exam question: anyone know what format the questions for the written are in? multiple choice is my assumption... |
| 00:39.45 | trnzmeta | choose D |
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| 01:28.12 | adeel | trnzmeta, asterisk option...found in sip.conf as of * 1.4 |
| 01:28.43 | trnzmeta | cheers :) |
| 01:29.55 | adeel | i'm running * 1.4.18, and it seems that i'm no longer getting call progress indications (and hence no ring backs) when dialing out, so my users keep thinking the line has dropped and hang up, even though it's going through...any way to figure out why this is happening? it used to work with the same provider |
| 01:36.38 | *** join/#asterisk Yourname` (n=yourname@unaffiliated/yourname/x-837320) |
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| 01:44.23 | russellb | adeel: you could try an up to date asterisk version first :-) |
| 01:45.30 | pmhaddad | is pretty sure he found a bug in 1.6.0.9 |
| 01:45.41 | pmhaddad | doing some more testing... |
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| 02:10.19 | adeel | russellb, heh, yeah i think that's my next step |
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| 02:21.36 | *** join/#asterisk BeeBuu (n=beebuu@121.9.84.11) |
| 02:21.59 | BeeBuu | hi,all |
| 02:22.25 | adeel | hm, the update to 1.4.21.2 didn't help either |
| 02:22.59 | BeeBuu | anyone teache what's the 'T' mean come with cmd meetme? 'T' â set talker detection (sent to manager interface and meetme list) |
| 02:23.58 | *** join/#asterisk Defraz (n=T0tal@24-117-236-174.cpe.cableone.net) |
| 02:24.52 | pmhaddad | BeeBuu, T basically identifies who is talking on which channel |
| 02:25.19 | pmhaddad | with it enabled you can use meetme list in the asterisk CLI to see them |
| 02:25.58 | BeeBuu | pmhaddad: and where can i get that messages? |
| 02:26.11 | pmhaddad | BeeBuu, the asterisk CLI |
| 02:26.15 | pmhaddad | asterisk -rvv |
| 02:26.25 | pmhaddad | assuming asterisk is running |
| 02:26.40 | BeeBuu | o,let me try |
| 02:26.47 | *** part/#asterisk etfonhomey (n=etfonhom@74-143-192-75.static.insightbb.com) |
| 02:27.09 | BeeBuu | set in a var? |
| 02:27.21 | pmhaddad | hm? |
| 02:27.45 | BeeBuu | store in a variable? |
| 02:28.14 | KyleK | what would I use to Play a random file? |
| 02:28.19 | pmhaddad | store what in a variable? |
| 02:29.00 | BeeBuu | who is talking in meet room |
| 02:29.13 | pmhaddad | BeeBuu, sure i guess you could do that |
| 02:29.33 | *** part/#asterisk juanIMP (n=Juancho@200.26.152.222) |
| 02:29.59 | BeeBuu | i got you,thanks. it's "meetme list roomnum" |
| 02:30.20 | pmhaddad | oh. i thought you meant store the result of the command into a variable... |
| 02:30.32 | BeeBuu | ya |
| 02:31.39 | [TK]D-Fender | KyleK: "core show application playback |
| 02:31.45 | pmhaddad | that would be a bit tougher because the MeetMe app in the dialplan doesn't have the list option like that so you can't just store that in a variable like you would an extension |
| 02:31.59 | pmhaddad | actually i'm not even sure if you can |
| 02:32.03 | *** join/#asterisk jer (n=jtregunn@unaffiliated/jer) |
| 02:33.40 | KyleK | thx |
| 02:34.20 | BeeBuu | pmhaddad: thanks |
| 02:35.06 | *** join/#asterisk alancio (n=Alancio@p1119-ipngn904hodogaya.kanagawa.ocn.ne.jp) |
| 02:35.06 | pmhaddad | np |
| 02:35.10 | [TK]D-Fender | pmhaddadPlenty of was |
| 02:35.19 | [TK]D-Fender | pmhaddad: Plenty of was |
| 02:35.21 | [TK]D-Fender | ways |
| 02:35.23 | [TK]D-Fender | dammit |
| 02:35.49 | [TK]D-Fender | Then again... whats the point of it being set in a channel variable anyway? |
| 02:36.04 | pmhaddad | [TK]D-Fender, right |
| 02:36.15 | pmhaddad | i may have misunderstood his question totally :S |
| 02:36.44 | [TK]D-Fender | pmhaddad: He brought it up, and as a statement ending with a question mark. |
| 02:36.52 | [TK]D-Fender | GRAMMAR FAIL |
| 02:36.54 | [TK]D-Fender | and |
| 02:36.56 | [TK]D-Fender | POINT FAIL |
| 02:37.00 | pmhaddad | heh |
| 02:37.05 | pmhaddad | you love the caps key |
| 02:37.23 | BeeBuu | pmhaddad: how can i play a file into a meet room? |
| 02:37.29 | [TK]D-Fender | BeeBuu: Why did you even mention channel variables for this? You have not been clear what you want to do. |
| 02:37.49 | pmhaddad | lets [TK]D-Fender take it from here and goes back to studying |
| 02:37.52 | [TK]D-Fender | BeeBuu: How are you deciding to play this file? |
| 02:38.32 | BeeBuu | [TK]D-Fender: emm, A auto play mechine~~~~ |
| 02:38.50 | [TK]D-Fender | BeeBuu: huh? |
| 02:39.04 | alancio | anybody knows of a cheap PCIe card with a fxo port? |
| 02:39.08 | [TK]D-Fender | BeeBuu: What will TRIGGER it being played? |
| 02:39.37 | BeeBuu | [TK]D-Fender: when someone in room press a key,then a bot will play a song |
| 02:39.53 | [TK]D-Fender | alancio: http://www.digium.com/en/products/analog/aex410.php |
| 02:40.20 | alancio | [TK]D-Fender: that costs 500$ |
| 02:40.58 | alancio | maybe its too new? |
| 02:41.06 | pmhaddad | alancio, $500 is a great price! |
| 02:41.17 | [TK]D-Fender | alancio: 284$ actually |
| 02:41.34 | alancio | where can I get it for 284$? |
| 02:41.36 | pmhaddad | hasn't bought a fxo card for under $500 |
| 02:41.56 | alancio | I bought several TDM400 cards for an average of 150$ (used) |
| 02:42.11 | alancio | in ebay there is only one publication, for 500$ |
| 02:42.13 | [TK]D-Fender | alancio: http://www.telephonydepot.com/ |
| 02:42.32 | alancio | thanks [TK]D-Fender |
| 02:42.45 | [TK]D-Fender | alancio: And you trust ebay as a friggen store? Yeah, I can SEE how hard you looked especially since I jsut gave you the model # |
| 02:43.23 | pmhaddad | would never buy a used fxo card |
| 02:43.36 | alancio | 500$ is the cheapest I found, I found more expensive options |
| 02:43.49 | BeeBuu | [TK]D-Fender: can i play a file to a meet room? |
| 02:44.09 | [TK]D-Fender | beeThere are ways |
| 02:45.04 | BeeBuu | would you teache me how? |
| 02:45.48 | [TK]D-Fender | BeeBuu: features.conf. Stare at it for a while |
| 02:46.16 | alancio | [TK]D-Fender: are the fxo and fxs modules the same as for the TDM cards? |
| 02:46.18 | [TK]D-Fender | alancio: and I found it at HALF that in about 15 seconds flat |
| 02:46.40 | [TK]D-Fender | alancio: Maybe you should actually read the specs on the card |
| 02:48.29 | BeeBuu | [TK]D-Fender: features.conf can work in meet room too? |
| 02:48.54 | [TK]D-Fender | BeeBuu: Please use your brain for a while and THINK of the ways you get to use it. |
| 02:49.21 | BeeBuu | got a tiny brain~~~~ |
| 02:51.09 | *** join/#asterisk voxter (n=voxter@190.10.13.241) |
| 02:51.26 | BeeBuu | the user in room as a caller ? |
| 03:08.02 | *** join/#asterisk ecret (n=ecret@CPE001e9002348e-CM001225d8ab30.cpe.net.cable.rogers.com) |
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| 03:22.45 | *** join/#asterisk aaroneous (n=aaroneou@p86-64.acedsl.com) |
| 03:23.10 | aaroneous | can anyone point me to some dialplan best-practices? |
| 03:23.53 | aaroneous | I want my PBX to function in a manner consistent with what users have experienced in other (IP)PBXs |
| 03:25.12 | aaroneous | (for instance we are considering eliminating the need to dial 9 for PSTN calls, but then we need a workaround for dialing internal extensions/applications) |
| 03:29.31 | [TK]D-Fender | aaroneous: No such woraround needed. Dial-9 is so 80's.... gah |
| 03:29.43 | [TK]D-Fender | aaroneous: I stopped doing that 5 years ago |
| 03:30.09 | styelz | use pattern matching yea? |
| 03:30.18 | aaroneous | [TK]D-Fender: how shall I have my users dial internal extensions without colliding with external calls? |
| 03:30.29 | [TK]D-Fender | No reason you can't have 8000 & 8001234567 in a dialplan |
| 03:31.16 | aaroneous | what if I have an extension 8001? |
| 03:31.23 | [TK]D-Fender | aaroneous: Because you are assuming that whatever is listening to digits HAS to stop instantly at 4 if its starts one way vs another. |
| 03:31.36 | [TK]D-Fender | aaroneous: 8001 is no different |
| 03:32.00 | aaroneous | how will it know then? |
| 03:32.20 | aaroneous | whether to wait for more digits or to connect to 8001.. |
| 03:32.35 | styelz | _800X. |
| 03:33.44 | [TK]D-Fender | aaroneous: First thing is to put a NAME to "it" |
| 03:34.04 | [TK]D-Fender | aaroneous: You need to consider what is doing the thinking and understand its logic |
| 03:35.42 | aaroneous | I have a lot of different "its".. some are cutting-edge polycom 650s and some are analog phones and other devices on FXS ports.. I want them all to behave the same way from the user's perspective so as to reduce confusion |
| 03:36.13 | [TK]D-Fender | aaroneous: And another idea is the remove the association of this "not so miraculous" ability from "IP-IPBX's". IP-PBX is a broken and worthless term really... |
| 03:36.24 | [TK]D-Fender | aaroneous: All doable |
| 03:37.41 | aaroneous | hmm.. I guess I'm having trouble understanding how I can distinguish between a call to x1800 and a call to 18005551212 |
| 03:38.05 | [TK]D-Fender | aaroneous: Tell me a way you can imagine it. Describe what you would do. |
| 03:38.46 | thehar | russellb: ? |
| 03:38.54 | aaroneous | I would require either * or # either before or after an extension |
| 03:39.11 | [TK]D-Fender | aaroneous: Ok, thats one way. Imagine another |
| 03:40.13 | aaroneous | I might try to see if there is some country code reserved for private use (an RFC1918 of telephone numbers) |
| 03:41.10 | [TK]D-Fender | aaroneous: No, your sample was perfect. 1800 + 7 NANPA dial clashing with the desire to use 1800 specifically for another purpose |
| 03:41.34 | [TK]D-Fender | aaroneous: Imagine the act of dialing and when you think something could think to take things one way vs another |
| 03:42.31 | aaroneous | well, what is it that you suggest? |
| 03:44.54 | [TK]D-Fender | aaroneous: Out of ideas as to how "the system" would know to differentiate besides specifying a terminating char? |
| 03:46.12 | aaroneous | yes.. out of ideas, tho I am not an asterisk or telephony expert.. just trying to gather some information so that I can direct my asterisk guy.. |
| 03:47.36 | aaroneous | and I'd probably go with a prefix character rather than a terminating character to avoid issues like 911, 311, etc |
| 03:47.56 | [TK]D-Fender | aaroneous: Heres the though... dial 1800 and STOP <- |
| 03:47.56 | *** join/#asterisk lanning (n=lanning@173.8.187.197) |
| 03:48.36 | [TK]D-Fender | aaroneous: Huh? 3 seconds passed and no more digits? Guess they must be DONE or something... |
| 03:48.40 | aaroneous | what if the user is just taking a long time to dial tho? |
| 03:48.41 | [TK]D-Fender | ;) |
| 03:49.00 | aaroneous | or what if they want to be connected quickly to the internal extension? |
| 03:49.10 | [TK]D-Fender | aaroneous: What happens if you pick up a regular analog phone on a regular analog like and dial 4 digits and just sit there? |
| 03:49.37 | aaroneous | sometimes I am dialing a number and need to pause to reference a piece of paper for the rest.. |
| 03:50.07 | aaroneous | [TK]D-Fender: the phone company annoyingly tells you to try again |
| 03:50.24 | [TK]D-Fender | aaroneous: get the whole # before you dial like the rest of the planet, and live with the very nominal possibilty of an overlap depending on what you shoose to have as your non-PSTN internal range |
| 03:50.25 | aaroneous | or whatever.. I can't remember.. |
| 03:50.37 | [TK]D-Fender | aaroneous: Yes, they tell you to move along quietly. |
| 03:50.49 | [TK]D-Fender | aaroneous: So dial & wait. all you need |
| 03:51.24 | [TK]D-Fender | aaroneous: not that depends on the concept of dialing OFF-hook. ON-hook dialing can allow you to wait around forever, but this is regardless of dialplans anway |
| 03:51.36 | aaroneous | I think I am going to go with #+extension |
| 03:52.09 | aaroneous | yeah it's just that my users are so stupid that it's easier to enforce that dialing be done in the same manner on and off-hook |
| 03:52.33 | aaroneous | otherwise they get confused when off-hook dialing doesn't work the same way as on-hook |
| 03:52.35 | [TK]D-Fender | aaroneous: that will be a little tricky o learn how to set up, and may impat your choices of hardware a little depending. |
| 03:52.42 | aaroneous | and then they come to me and tell me that the system is broken |
| 03:52.58 | [TK]D-Fender | aaroneous: well ON-hook dialing doesn't have timout OR cut-off rules. No dodging that bullet |
| 03:54.18 | aaroneous | eh I don't want such rules for off-hook dialing either.. aren't such rules a legacy/relic of phone companies trying to minimize the use of finite switch resources? |
| 03:55.02 | [TK]D-Fender | aaroneous: NO dodging this bullet |
| 03:55.06 | aaroneous | plus if extension 1800 is the CEO for instance, I don't want a lot of people accidentally bothering him |
| 03:55.29 | [TK]D-Fender | aaroneous: Analog phones that can dial on-hook don't have dialplans. other digital phones BYPASS them. |
| 03:55.57 | [TK]D-Fender | arrWell I guess you'd probably NOT want to pick an extension rang that overlaps in a terminally silly way then. |
| 03:56.18 | aaroneous | or if some popular person is at extension 9115 such rules wouldn't be cool either |
| 03:56.44 | aaroneous | nor do I want people having to wait an extra 3 seconds to get their 911 call connected in an emergency |
| 03:57.09 | [TK]D-Fender | aaroneous: While you hadn't come up with a bunch of ways this could work, you come up with the ways that can fail. So don't do those :p |
| 03:57.22 | [TK]D-Fender | "Doctor, doctor! It hurts when I raise my arm like this!" |
| 03:57.30 | aaroneous | nor do I revoking extensions in the future when I learn that they contain some reserved sequence that I didn't know of initially |
| 03:57.57 | aaroneous | nor do I want to be, that is |
| 03:58.50 | [TK]D-Fender | aaroneous: But you don't want "dial 9". How the hell did you think they did it? :) |
| 03:59.41 | [TK]D-Fender | aaroneous: Welcome to "the obvious rules of physics". You want these options, choose for yourself how best to avoid and there are still rules to follow |
| 04:00.00 | aaroneous | well it's more that most calls placed are outbound, so I'd rather have "dial # before dialing extensions" than "dial 9 before outbound calls" |
| 04:00.25 | [TK]D-Fender | aaroneous: Do any of these otehr PBX's you've see force that? |
| 04:00.34 | [TK]D-Fender | aaroneous: Believe be you're making an Everest out of a mole-hill |
| 04:00.53 | aaroneous | well I was just hoping for a more foolproof and clever common solution |
| 04:01.09 | aaroneous | but it appears that there isn't one (at least there isn't one in common usage) |
| 04:01.19 | [TK]D-Fender | aaroneous: Nothing is foolproof, because we all know how gosh-darned clever fools can be. |
| 04:01.35 | [TK]D-Fender | aaroneous: Or worse yet... you work with IDIOTS. Those guys are dangerous |
| 04:01.58 | aaroneous | [TK]D-Fender: I wouldn't hesitate to describe a substantial number of our employees as such |
| 04:02.29 | [TK]D-Fender | aaroneous: Been there, done that. You'd be amazed at how effective pain-therapy is :) |
| 04:02.44 | ltd_wk | Anyone know much about the semantics of SIP 302 redirects with * 1.4? specifically when dialling a peer handset that sends back a 302. I'm observing that * 1.4 seems to create a local channel in the dialout context of the peer... |
| 04:03.26 | [TK]D-Fender | ltdIndeed rather natural |
| 04:03.46 | ltd_wk | Except, something seems to go wrong when that happens. It dials the redirected number, then hangs up shortly after |
| 04:04.05 | ltd_wk | after approx 2 rings. |
| 04:04.16 | aaroneous | [TK]D-Fender: pain-therapy doesn't work on these idiots.. they are very stubborn and will let business grind to a halt before you make things work in a manner compliant with their pea-brained expectations |
| 04:04.57 | [TK]D-Fender | aaroneous: then you'll have to get approval for use of "enhanced-training" techniques. |
| 04:05.14 | aaroneous | that, and I want a simple configuration such that I don't need to consider every extension or set of extensions to reserve |
| 04:05.50 | aaroneous | [TK]D-Fender: when business grinds to a halt, I take the heat, not the idiots grinding it to a halt because of their refusal to learn |
| 04:06.11 | [TK]D-Fender | aaroneous: remember that the only point of contention is an idiot picking something like 911X clash or jsut sitting around |
| 04:06.46 | [TK]D-Fender | aaroneous: Taht being said, SOMETHING is about to have a fixed prefix by your definition. |
| 04:06.47 | aaroneous | we're talking about a company where nobody has seemed able to figure out their voicemail yet because it "doesn't work like my cell phone voicemail does" |
| 04:07.07 | [TK]D-Fender | aaroneous: You're jsut stealing the "9" off an "outbound number" and moving it around |
| 04:07.30 | [TK]D-Fender | aaroneous: Oh.. and in what way is your current VM different just out of curiosity? |
| 04:08.08 | ltd_wk | [TK]: any idea what might cause the hangup with the 302? |
| 04:08.47 | aaroneous | well, for one, the asterisk voicemail we're using has this nice feature of being able to navigate in real time between back and forth between messages in the index.. verizon wireless et al. just make you wait for the message to end or delete it |
| 04:09.17 | aaroneous | two, they don't understand the concept of VM folders, so they think that if they go through all of their new messages that they'll start to hear their old messages |
| 04:09.32 | [TK]D-Fender | aaroneous: Guess what, they'll bitch at ANYTHING different then, and EVERYTHING is different. They can officially "fuck off" and deal with it :) |
| 04:09.55 | [TK]D-Fender | aaroneous: You mean Panasonic VM is different than Norstar NAM2?!?! OMG Riot!!!!!!!1 |
| 04:10.11 | *** part/#asterisk manipura (n=Mike@S01060022b0d49327.cg.shawcable.net) |
| 04:10.28 | aaroneous | [TK]D-Fender: no.. what they'll do is "fuck off" and not check their messages as a result.. and then sales inquiries and orders will be ignored, and then the company will go out of business |
| 04:10.54 | aaroneous | this is a seriously backward and bumbling company that I am trying to salvage here |
| 04:11.20 | *** join/#asterisk yo-mama (n=bsumrall@ftnco.com) |
| 04:11.23 | aaroneous | but we have this culture of people not being challenged to learn new tricks, and this culture has very deep roots |
| 04:11.34 | [TK]D-Fender | aaroneous: You know what. You've said that in a way that really signs off as "You're already DOA". Why bother? Guess nothing will make them happy. Let them use nothing then. Merry Christmas |
| 04:11.52 | yo-mama | does anyone know of software or a feature for sms blasting? |
| 04:12.20 | [TK]D-Fender | aaroneous: Think the boss will let them get away with it when it costs business? You'd be amazed how fast people learn when their ass is on the line for neglecting important calls. |
| 04:12.35 | aaroneous | [TK]D-Fender: the boss is the same way |
| 04:12.49 | [TK]D-Fender | aaroneous: its HI money. Good luck with that. |
| 04:12.52 | [TK]D-Fender | HIS* |
| 04:13.13 | aaroneous | this is not a normal company we're talking about here |
| 04:13.25 | denon | aaroneous: obama contract? |
| 04:13.43 | aaroneous | nor is the nature of my job normal |
| 04:13.55 | denon | I knew it .. west wing staffer |
| 04:13.58 | [TK]D-Fender | aaroneous: Seriously. This is a completely unneeded conversations. You seem to have already given up. Any company that stupid is going to fail. God luck with all that |
| 04:13.58 | aaroneous | denon: nah.. family business |
| 04:14.00 | [TK]D-Fender | good* |
| 04:14.30 | *** join/#asterisk CunningPike (n=CunningP@S01060014bf81366b.vc.shawcable.net) |
| 04:15.29 | aaroneous | I haven't given up.. I'm just not going with the annoying/inconvenient/possibly_dangerous approach of using timeouts to accomplish this goal |
| 04:16.01 | [TK]D-Fender | aaroneous: Well then you're shuffling reserved prefixs around |
| 04:16.09 | [TK]D-Fender | aaroneous: Which you just said you didn't want. |
| 04:17.05 | yo-mama | does anyone know of software or a feature for sms blasting? |
| 04:17.21 | aaroneous | #extension should work, right? that'll never be in the international numbering plan |
| 04:18.07 | [TK]D-Fender | aaroneous: Well according to other telecom standards, "#" is suposed to mark an end-of-dial |
| 04:19.18 | [TK]D-Fender | aaroneous: Certain devices won't appreciate that. Like Analog Zap/DAHDI channels, and many ATA's |
| 04:19.24 | aaroneous | ick |
| 04:20.12 | aaroneous | longs for the day when we are done with the antiquated notion of telephone numbers |
| 04:21.16 | aaroneous | I'm still annoyed that I couldn't find IP phones with qwerty input for SIP identities |
| 04:21.37 | [TK]D-Fender | longs for the day when you can freely beat people whothink they can go through life being idiots. |
| 04:21.56 | [TK]D-Fender | Cope, or get your sorry ass RUN OVER by the rest of the planet. |
| 04:22.22 | [TK]D-Fender | Coddle idiots and you encourage them to remain such. |
| 04:22.39 | [TK]D-Fender | points to KerryG & FreePBX |
| 04:22.41 | [TK]D-Fender | :p |
| 04:23.05 | aaroneous | [TK]D-Fender: be nice now.. I appease idiots because that's how I stay employed here.. |
| 04:24.11 | aaroneous | I am slowly trying to retrain them, but there's at least a 15-year history to why they are so stupid, so change isn't going to happen overnight.. its going to be incremental, and I can't just fire everyone and hire/train replacements overnight |
| 04:27.16 | drmessano | aaroneous: Qwerty inputs is the wrong line of thinking |
| 04:27.23 | drmessano | Same thinking that will never get us past this |
| 04:28.29 | drmessano | We need to be using some interchangable form of contact data.. Send me your contact, I put it in my PIM, which Syncs to my phone, IM client, Smartphone, etc |
| 04:28.39 | drmessano | Then I click your name, done |
| 04:29.03 | aaroneous | why should email address and phone identities be different? |
| 04:29.50 | aaroneous | why not have both mailto:aaroneous@acmecorp.com and sip:aaroneous@acmecorp.com? |
| 04:30.04 | [TK]D-Fender | drmessano: Tied to a single identity? How antiquated. I want revolving identities like James Bond had revoloving license plates! |
| 04:30.26 | [TK]D-Fender | aaroneous: You can |
| 04:30.31 | aaroneous | and why not have telephones capable of "dialing" SIP identities without convoluted t9/etc nonsense? |
| 04:31.02 | aaroneous | [TK]D-Fender: yeah.. I know you can.. my point is that I was complaining about the lack of QWERTY input in IP desk phones |
| 04:31.05 | [TK]D-Fender | aaroneous: Think your family business is hard to change? Try changing the PLANET :) Oh wait... thats what you've just suggested |
| 04:31.08 | drmessano | aaroneous: Never said they were |
| 04:31.17 | [TK]D-Fender | arrTell you what... start with your OWN back yard |
| 04:31.27 | drmessano | But you dont need input from the phone |
| 04:31.56 | [TK]D-Fender | INPUT? You mean the phone won;'t just pull numbers out my ass?!?! err... head!@ |
| 04:32.00 | drmessano | Dialing a SIP ID defeats the purpose? |
| 04:32.24 | [TK]D-Fender | drmessano: Kinetic energy dialing? EW! |
| 04:32.27 | aaroneous | if my phone doesn't have QWERTY it might as well not have 0-9 either, as far as I see it |
| 04:32.40 | drmessano | No one is gonna dial a URI.. the contact should be there from a previous exchange |
| 04:32.45 | [TK]D-Fender | aaroneous: BRILLIANT! Do away with "numbers"! |
| 04:32.51 | drmessano | Qwerty input is going BACKWARDS |
| 04:33.32 | aaroneous | drmessano: and in your world, how exactly will the initial exchange be established, if there is no means of entering user@domain.tld or telephone number? |
| 04:34.00 | [TK]D-Fender | aaroneous: Memory-engram relational databe exchange of course |
| 04:34.11 | aaroneous | [TK]D-Fender: yes. telephone numbers would not exist if we were reinventing the teelphone |
| 04:34.16 | drmessano | This is unified messaging.. The phone gets its info from a PC or other device.. I shouldnt even be initiating a call from the handset.. I should be clicking to dial from an application |
| 04:34.22 | [TK]D-Fender | aaroneous: Now please insert your head into the MRI so we can swap contact details! |
| 04:35.33 | aaroneous | drmessano: well, of course you might as well ditch the desk phone in the first place (not that I disagree with this radical notion, tho I'm sure my users would) if that's your thinking (but I don't think it is) |
| 04:36.07 | [TK]D-Fender | hordes his pre-release copies of res_telepathy.so and res_fluxcapacitor.so |
| 04:36.20 | drmessano | If the phone is going to have any "dialing" at all, it should be in the form of a mini contact list synced from your unified messaging account, an up/down scroll key, and a dial button |
| 04:36.54 | aaroneous | what is so hard or radical about dialing user@domain? you type this every time you want to email someone for whom you have a business card.. |
| 04:37.00 | drmessano | I hear the SNOM's have pretty slick client that sync's with Outlook like that |
| 04:37.01 | [TK]D-Fender | drmessano: NEURAL dammit! You and your damn infernal kinetic-energy input controls! |
| 04:38.13 | *** join/#asterisk LeddyHM (n=NONE@mail.artica.net) |
| 04:38.30 | drmessano | aaroneous: Inputting qwerty on a phone is a step backwards.. You're gonna convince me going from 10 digit dialing to entering a full fucking SIP address is gonna be BETTER? Im an ignorant user, HELL NO.. The way SIP URI's are going to work, and IMO, should work, is their integration into unified contact info which is best dialing with menu interfaces and simple controls |
| 04:38.44 | drmessano | The "entry" uglyness, if needed, can be done from a PC.. |
| 04:39.43 | aaroneous | drmessano: which is more difficult to remember, an "email address" which can be used as a unified identifier, or some arbitrary string of numerals, which can only be used for phone calls? |
| 04:40.30 | [TK]D-Fender | drmessano: Reminds me of some people how I set a PC up for that kept trying to send e-mails to addresses like www@john:shimt/hiscompany. |
| 04:40.44 | aaroneous | frankly I find alice@acemecorp.com a lot more memorable than whatever the hell her phone number and extension are |
| 04:41.08 | [TK]D-Fender | aaroneous: Yes, and you're describing idiots who can't handle FUCKING VOICEMAIL. |
| 04:41.38 | [TK]D-Fender | aaroneous: this is like the the pot vs the entire kettl-producing industry. |
| 04:41.46 | drmessano | aaroneous: Has nothing to do with remembering.. You're still working under this notion that I want to enter a URI everytime I want to make a call.. Who the hell even enters phone numbers anymore? We work off contact lists.. |
| 04:41.57 | [TK]D-Fender | aaroneous: Oh.. an FYI... they're both black :0 |
| 04:42.04 | drmessano | It should be the same as your cell phone.. Working out of a contact list.. click, dial |
| 04:42.12 | aaroneous | can we tone down the language? |
| 04:42.35 | drmessano | What language? |
| 04:42.57 | aaroneous | I just want to keep the debate civil and polite, that's all.. |
| 04:43.02 | [TK]D-Fender | aaroneous: Figured I'd drive the point home, no serious harm intended. But you've just described working the very people who will never survive your ideads nor your ability to change them. |
| 04:43.08 | drmessano | Who said it wasnt? Oversensitive? |
| 04:43.11 | [TK]D-Fender | aaroneous: Will tone down. |
| 04:43.25 | [TK]D-Fender | ideals* |
| 04:43.52 | drmessano | wanders off |
| 04:43.53 | aaroneous | tnx.. I know this is a very heated subject :> |
| 04:44.09 | [TK]D-Fender | aaroneous: Try remembering how to spell everybody's names. Espeically foreign ones. Or ones with minor writing differences. |
| 04:44.38 | [TK]D-Fender | aaroneous: Alpha dialing is a land-mine waiting for a walk in the park |
| 04:45.05 | aaroneous | I actually find having to remember some information about people helpful to my overall memory.. |
| 04:45.22 | drmessano | Using ONE address for everything UNIFIES the process of contacting someone.. It doesnt SIMPLIFY the exchange process.. |
| 04:45.44 | [TK]D-Fender | aaroneous: Consider the massive dyslexia in America alone. Oh? Sorry, you DON'T come from a western culter? Where are my Chinese charaters to dial up for take-out? I'll STARVE! :) |
| 04:46.07 | [TK]D-Fender | culture. WOW, my typing skills have gone right down the toilet |
| 04:46.14 | drmessano | user@host addresses are NOT easier 99% of the time.. if you dont believe, call john.rashimahama@fandoopharmcologicals.com |
| 04:46.51 | [TK]D-Fender | drmessano: The "p" is silent.... like in swimming ;) |
| 04:46.57 | drmessano | The idea is to create unified address objects and be to use that as IM contacts, Mail contacts, and be able to click a button and dial that user just the same |
| 04:47.02 | [TK]D-Fender | starts a spelling bee |
| 04:47.05 | drmessano | lol |
| 04:47.26 | aaroneous | I think we'll deal with non-western characters on telephones the same way we deal with non-western characters in email addresses on PCs.. |
| 04:47.45 | drmessano | Yes, by telling someone to send their contact info and putting it in outlook |
| 04:47.55 | drmessano | So all I do is click and send |
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| 04:48.32 | [TK]D-Fender | drmessano: Real curse happens when neither side uses the others charater set and can't type it in :) |
| 04:48.40 | drmessano | Where I will take great issue is the implication users actually manage email addresses any better than phone numbers.. Forget SIP URIs, the very notion of management of email addresses with MOST users is a Unicorn.. a fairy |
| 04:49.05 | Qwell | SO... who's got an Ubuntu 9.04 box that can confirm some behavior for me really quick? |
| 04:49.06 | drmessano | But entering the info for "John Johnson" ONE time and being able to IM, phone, and email him from that same info is very powerful |
| 04:49.55 | drmessano | Being able to click a DIAL button in Outlook or your IM client to continue the conversation on the phone is very slick |
| 04:50.14 | aaroneous | I don't know.. I can remember the email addresses of a lot of my friends and business contacts/colleagues, whereas I can't remember their phone numbers.. |
| 04:50.18 | drmessano | But I dont think SIP URI's are going to simplify the exchange.. They're just not less complicated |
| 04:50.23 | drmessano | YOU can |
| 04:50.33 | drmessano | I know hundreds of users that CANT |
| 04:50.35 | aaroneous | and you're talking to someone who has a fairly good unified contact management system.. |
| 04:50.37 | drmessano | Average people |
| 04:50.51 | drmessano | Youre not being objective and basing this simply on your limited personal experience |
| 04:52.24 | drmessano | People _do not_ remember email addresses.. I could argue better/worse than phone numbers, but thats moot here.. If I handed out my email address to 100 average people, I would be surprised if 5% had retention |
| 04:53.07 | drmessano | People still email JOEUNDERSCORESMITH@yahoo.com and wonder why it didnt go through |
| 04:53.14 | drmessano | or STEVEPERIODJOHNSON@gmail.com |
| 04:53.24 | aaroneous | drmessano: so I guess my brain must be radically abnormal then? |
| 04:53.40 | drmessano | aaroneous: No, you're a tech.. we dont think like other people |
| 04:54.11 | drmessano | aaroneous: We're abnormally biased towards things average realtors, soccer moms, and construction workers would not be |
| 04:54.32 | aaroneous | I see |
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| 04:55.35 | drmessano | Now, if I sent you a contact "thingo" in Outlook and had a fairly intelligent IT person who trained the users how to drag that into their contact list, I could get you to click that to IM, phone, or email me, all day long |
| 04:55.53 | drmessano | No number to remember or lose, no "Oh crap, he got a new phone.. ummm" |
| 04:56.04 | [TK]D-Fender | aaroneous: Yes, but now not only do you have to remember their full name, but also a domain? Wait, that was yahoo.fr not gmail.com? |
| 04:56.18 | [TK]D-Fender | aaroneous: What about subdomains? |
| 04:56.42 | [TK]D-Fender | aaroneous: Save that secret decoder ring at the bottom of your box of Froot-Loops, you're gonna need it ;) |
| 04:57.00 | aaroneous | [TK]D-Fender: I've seen subdomains falling out of fashion for use in email addresses.. |
| 04:57.05 | drmessano | I think the SIP URI dialing is a great BACKEND for the process... it makes the NUTS and BOLTS a lot easier.. it does NOT simplify it for the user.. Unifying the contact info would somewhat |
| 04:58.10 | aaroneous | and is jane.doe@ibm.com really more difficult than 914-499-1973? |
| 04:58.31 | drmessano | john.rashmatanata@countlesstechnologies.com is |
| 04:58.44 | drmessano | Most people do NOT have quick and short email addresses |
| 04:59.03 | drmessano | ESPECIALLY for personal use |
| 04:59.10 | aaroneous | drmessano: yeah, in the same way that most americans don't know how to place an international call or indicate a country code.. |
| 04:59.27 | drmessano | Cowgirlwithatruckfulloflove1976@yahoo.com does not make a good SIP URI |
| 04:59.57 | drmessano | But that will be their address, and you will have to deal with it |
| 04:59.58 | aaroneous | drmessano: well should we really have any sympathy for that kind of n00bery? |
| 05:00.31 | *** join/#asterisk fakhir_ (n=fakhir@unaffiliated/fakhir) |
| 05:00.43 | drmessano | aaroneous: Its a techs job to engineer solutions around it.. we can only apply so much to the fix |
| 05:01.35 | drmessano | If the business owner decides giantbigsolutionsfromalittleoldhouse.com is a GREAT domain name, what are you gonna tell them? "Fuck you, thats too long.. change it" |
| 05:01.41 | drmessano | No, youre stuck.. |
| 05:01.54 | aaroneous | I'm happy to let phone numbers become pointers to SIP URIs as a transitional system, but I think given the choice between remembering a string of numbers or an "email address", in the long term users will choose the email address |
| 05:02.07 | drmessano | Then he says "Now I want SIP dialing.. I saw it on CNET.. make it happen" |
| 05:02.34 | drmessano | People wont remember his domain.. they wont want to enter it into ANYTHING more than once |
| 05:02.45 | drmessano | Which is were unified messaging shines |
| 05:03.03 | drmessano | You enter it once and you can contact him 10 different ways, same address, with a CLICK |
| 05:03.08 | drmessano | Never have to deal with it again |
| 05:03.18 | aaroneous | his choice of domain is doomed for many more reasons than just SIP dialing |
| 05:03.22 | drmessano | Thats where it makes a good BACKEND |
| 05:03.34 | drmessano | aaroneous: Moot.. He pays the bills.. |
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| 05:04.26 | aaroneous | drmessano: I basically have the ideal unified contact management system that you describe.. it is a reality in my life.. |
| 05:04.33 | drmessano | aaroneous: You cant vary your argument based on the users domain choice.. if you're gonna apply this en masse, you must account for all iterations of stupidity and whims of users |
| 05:05.05 | drmessano | and in the case of crappy, hard to remember addresses, entering a URI into a phone manually is nuts |
| 05:05.09 | aaroneous | drmessano: and you know what, it really sucks when someone changes their phone number, IM identity, or email address(es) without bothering to notify me.. |
| 05:05.11 | yo-mama | smsq question. I am following the instructions to the letter but all attempts fail and test message is just sitting in the spooler |
| 05:05.38 | drmessano | aaroneous: EXACTLY why they need ONE address.. one you never have to remember or worry about |
| 05:06.36 | aaroneous | you haven't described how that "outlook contact" is going to be automagically updated every time their cell phone number changes.. |
| 05:06.43 | drmessano | aaroneous: Im with you 100% on the use of the SIP URI's as the backend for calling... I just completely disagree about it being a better user experience.. I dont see it that way at all.. I think the address is exchanged, goes into a contact, and from there, is used however.. never to be remembered again |
| 05:07.23 | drmessano | aaroneous: URI on the cell phone.. already being done |
| 05:07.50 | yo-mama | has anyone here ever worked with smsq? |
| 05:08.26 | aaroneous | holy crap it's already 1AM |
| 05:08.36 | aaroneous | I somehow thought it was like 11:30 |
| 05:08.38 | aaroneous | shite |
| 05:08.49 | aaroneous | gotta get out of the office! |
| 05:08.55 | drmessano | heh |
| 05:09.01 | aaroneous | c'ya all |
| 05:09.07 | drmessano | take care |
| 05:09.13 | aaroneous | likewise.. g'night |
| 05:09.17 | aaroneous | out |
| 05:12.51 | yo-mama | does anyone know how to use fastsms or smsq? |
| 05:17.49 | [TK]D-Fender | Ok, checkout time, later all |
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| 05:38.42 | ltd_wk | Here's a verbose log of the 302 redirect gone bad - http://pastebin.ca/1413026 - the call gets dumped just after the redirected call gets answered |
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| 05:40.20 | ltd_wk | -- Local/0423746742@service_19_outdial-2209,1 stopped sounds -- wondering if this is related |
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| 05:51.41 | yo-mama | does anyone know how to use fastsms or smsq? |
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| 06:48.58 | yo-mama | does anyone know how to use fastsms or smsq? |
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| 07:07.09 | cool^tom | Hi. Currently I have a PBX with a PRI Line comming into it. I was wondering if it is possible to have an asterisk box with a 2span PRI. One pri connectes to my telco and the other PRI connects to my PBX. The PBX would act more like a channel bank. Would such a scenario be possibel in Asterisk? |
| 07:12.23 | dpryo | cool^tom: yeah, that works fine. I've got a bunch of asterisks acting as pri-gateways for more stupid expensive calling-equipment |
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| 07:24.24 | redax | hi, |
| 07:25.06 | redax | is there anything changed on the manager interface related to Action: originate, other than I need the 'originate' right as well in the manager.conf ? |
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| 07:37.57 | troy- | is there anything cheaper than the snom 370 w/ builtin vpn client? |
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| 08:17.48 | Stese | Hey all |
| 08:18.48 | Stese | I'm trying to debug a * ISDN issue, and i'm just wondering what "Cannot hangup chan, no ast" means? |
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| 08:29.52 | mkarg | good morning! |
| 08:30.39 | Stese | Morning |
| 08:30.46 | mkarg | hi Stese, |
| 08:31.47 | mkarg | Stese: which linux distro would you recommend for an asterisk server with some more services to run on the same machine (samba, dns, etc.) |
| 08:32.14 | mkarg | I used to run centos but the asterisk packages from atrpms are broken meanwhile, so I'm looking for a new one. |
| 08:32.45 | mkarg | I had a quick look on eisfair, but I'm not yet convinced... |
| 08:35.21 | *** join/#asterisk joobie (n=joobie@124-168-34-12.dyn.iinet.net.au) |
| 08:35.49 | Stese | Well, the only one i've used is CentOs, myself... but I know that many people use Fedora and Debian |
| 08:36.23 | mkarg | Stese: what version of CentOS are you running and do you install asterisk from the sources? |
| 08:37.23 | Stese | Erm, the latest from the website, and yes, but i'm still learning myself, so didn't mange to get it all running.... but that it most likely my inexperience more than anything else |
| 08:37.27 | joobie | sup ladies |
| 08:41.36 | tzafrir_laptop | Stese, which ISDN? (dahdi / misdn? bri / pri?) |
| 08:41.37 | *** join/#asterisk defsdoor (n=andy@82.133.90.135) |
| 08:42.02 | Stese | tzafrir_laptop : mISDN on a b410P BRI |
| 08:45.28 | KyleK | im running asterisk on the packages in jaunty bwahahaha |
| 08:47.57 | tzafrir_laptop | KyleK, those are basically Debian packages |
| 08:48.59 | tzafrir_laptop | Most Universe packages are packages basically copies from Debian Unstable at freeze time, with a minimal Ubuntu-specific patch |
| 08:50.22 | KyleK | i kinda like minimal patching |
| 08:50.45 | KyleK | lots of patches in a distro just screams "we're too cool to submit patches upstream" |
| 08:57.03 | mkarg | does anyone know if there is another repo for centos than atrpms containing asterisk packages? |
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| 09:09.10 | cool^tom | Would it be possible to configure a PRI Line as fxs? |
| 09:12.34 | KyleK | whats the PRI plug into? |
| 09:15.12 | KyleK | if by PRI you mean something like a T1 or bigger then yes |
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| 09:29.01 | *** join/#asterisk possy (n=possy@smtp.theinternet.de) |
| 09:29.11 | possy | good morning |
| 09:30.00 | MaliutaLap | but mornning is still 4 and a half hours away |
| 09:31.07 | possy | he,he |
| 09:31.32 | possy | I have multiple entries in my full log of asterisk that all look like this: http://pastebin.com/d3d2c8a31 |
| 09:32.08 | possy | in a matter of seconds I have thousands of these entries |
| 09:32.22 | possy | http://pastebin.com/d37e7b64d contains the relevant part of the extensions.conf |
| 09:33.27 | Stese | possy > What are your debug and verbose levels set to? |
| 09:33.37 | possy | full |
| 09:33.43 | possy | set verbose 9 |
| 09:33.56 | possy | plus sip debug |
| 09:34.10 | possy | nnected to Asterisk 1.4.21.2~dfsg-3 currently running on asterisk (pid = 30744) |
| 09:34.10 | possy | Verbosity is at least 9 |
| 09:34.25 | Stese | looks like it's just standard output |
| 09:34.41 | possy | stese, probably, but not 25000 times per second |
| 09:35.07 | possy | I assume, my extensions.conf contains an error |
| 09:35.15 | Stese | have you proven the security of your * |
| 09:35.17 | Stese | ? |
| 09:35.32 | possy | It is not connected to the net |
| 09:35.55 | Stese | it's connected to something |
| 09:36.16 | possy | to a small local network |
| 09:36.58 | tzafrir_laptop | cool^tom, PRI is not FXS. PRI is ISDN. You may be asking about using FXS over E1 or T1? |
| 09:37.25 | possy | Stese, why do you ask? |
| 09:37.44 | KyleK | its PBX security week |
| 09:38.06 | KyleK | convince 3 people to secure thier PBX and get a free coffee mug |
| 09:38.20 | tzafrir_laptop | possy, you have a loop in your dialplan? |
| 09:38.32 | Stese | lol... maybe kaii |
| 09:38.34 | Stese | oops |
| 09:38.37 | *** join/#asterisk Andre101 (n=a@123-243-77-135.tpgi.com.au) |
| 09:38.37 | Stese | tab error |
| 09:38.38 | Stese | lol |
| 09:38.41 | possy | he,he |
| 09:39.04 | possy | tzafrir_laptop, i do? yes, if they fall out of the queue, I want them to here the announcement again |
| 09:39.16 | Stese | KyleK > maybe, but since the calls looked normal, I thought it was a fair question |
| 09:39.27 | Andre101 | Hello people.. can someone point me in the right direction to fix this : http://www.pastebin.ca/1413127 |
| 09:39.36 | *** join/#asterisk Great_Anta_Baka (n=tensai@196.33.159.83) |
| 09:39.44 | Andre101 | sip will register fine at first, then it will time out and give these errors |
| 09:40.44 | possy | tzafrir_laptop, is there a better way of doing what I do? Or some settings I should look into? |
| 09:41.11 | tzafrir_laptop | possy, is it the same channel every time? If so, you have some sort of a loop |
| 09:41.26 | possy | it looks identical each time |
| 09:42.10 | *** join/#asterisk joelsolanki (i=joelsola@202.160.161.94) |
| 09:42.14 | joelsolanki | Hi all |
| 09:42.26 | possy | tzafrir_laptop, I will look into the Queue command and see, what I can find |
| 09:42.39 | joelsolanki | i am trying to install dahdi-linux and it gives me error.You do not appear to have the sources for the 2.6.18-128.el5xen kernel installed |
| 09:42.50 | joelsolanki | this is centos5.3 with xen installed |
| 09:42.58 | joelsolanki | will it work with xen kernel ? |
| 09:43.42 | tzafrir_laptop | linux-devel-xen or something similar. That's the package you need |
| 09:44.02 | joat | hmm... meetme speaker detection is borked again in 1.6.1.0 |
| 09:44.20 | possy | tzafrir_laptop, I have remove the n parameter from the queue. Let's see what happens |
| 09:44.29 | joelsolanki | oh k. let me try :) |
| 09:45.02 | Andre101 | Hello people.. can someone point me in the right direction to fix this : http://www.pastebin.ca/1413127 ? |
| 09:47.54 | possy | tzafrir_laptop, thanks so far |
| 09:48.04 | *** part/#asterisk possy (n=possy@smtp.theinternet.de) |
| 09:48.47 | *** join/#asterisk viq (n=viq@unaffiliated/viq) |
| 09:52.05 | *** join/#asterisk Subdolus (n=subby@subby.afraid.org) |
| 09:52.44 | *** join/#asterisk Andre101 (n=a@123-243-77-135.tpgi.com.au) |
| 09:53.10 | Andre101 | Hello, can someone point me in the right direction to fix this? http://www.pastebin.ca/1413127 |
| 09:57.35 | *** join/#asterisk Andre101 (n=a@123-243-77-135.tpgi.com.au) |
| 10:01.13 | *** join/#asterisk lost_soul (i=shawn@cpe-67-241-68-104.twcny.res.rr.com) |
| 10:03.34 | *** join/#asterisk proxium (n=proxium@196.203.51.238) |
| 10:04.02 | lost_soul | gm everyone |
| 10:05.07 | proxium | Hi every body, I use Meetme conference with Asterisk and SipPhone, is it normal to get in CLI: app_meetme.c:778 build_conf: Unable to open pseudo device ??? |
| 10:13.18 | *** join/#asterisk shozaib (n=sunny@61.5.134.34) |
| 10:14.05 | tzafrir_laptop | proxium, that is from chan_dahdi.c . aparantly you don't have a dahdi module or something similar |
| 10:15.40 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 10:18.58 | proxium | tzafrir_laptop: I don't know how to check if I had one but let me know plz, is it necessary to use (Timing Device) as zaptel_dummy or dahdi_dummy and why we use them if we don't have any related Hardware only SoftPhone ?? |
| 10:20.03 | tzafrir_laptop | proxium, what version of asterisk is it? |
| 10:20.09 | proxium | 1.4 |
| 10:21.30 | proxium | tzafrir_laptop: Asterisk (Ver. 1.4.22) with Vicidial (2.0.5-203) |
| 10:22.56 | tzafrir_laptop | do you use it with zaptel or dahdi? |
| 10:25.24 | *** join/#asterisk huye (n=huye@soho2.i-xanadu.com) |
| 10:26.48 | proxium | tzafrir_laptop: Really I don't know how to check if I'm loading Zaptel or Dahdi module, but I remember when I install it I've skipped those steps because I need only testing envirenoment in VM with no Hardware only SoftPhone as Ekiga or X-Lite |
| 10:27.30 | tzafrir_laptop | proxium, ls -d /proc/zaptel /proc/dahdi |
| 10:27.52 | tzafrir_laptop | ah, so you probably have nither |
| 10:28.09 | tzafrir_laptop | so meetme won't work for you |
| 10:28.12 | *** join/#asterisk mikkel (n=mikkel@130.226.36.170) |
| 10:29.42 | proxium | no one so :) okay how to establish conference so with Meetme or other or is there an alternative to all this? |
| 10:32.11 | tzafrir_laptop | look for app_conference :-( |
| 10:32.29 | tzafrir_laptop | actually, try the latest dahdi RC |
| 10:32.39 | tzafrir_laptop | it should run well even in Xen |
| 10:33.16 | tzafrir_laptop | Though the load of Vicidial should prove as a useful load testing |
| 10:35.07 | *** join/#asterisk jtodd (n=jtodd@88.72.251.206) |
| 10:35.07 | *** mode/#asterisk [+o jtodd] by ChanServ |
| 10:37.05 | *** join/#asterisk Andre101 (n=a@123-243-77-135.tpgi.com.au) |
| 10:37.41 | Andre101 | Hello, can someone point me in the right direction to fix this? http://www.pastebin.ca/1413127 ? it's not registering.. i can get it to register with a softphone though |
| 10:40.08 | joelsolanki | Hi all getting error for kernel source while installing dahdi. Please see pastebin. http://pastebin.ca/1413158 |
| 10:40.11 | joelsolanki | any help ? |
| 10:40.29 | joelsolanki | i installed kernel source for centos5.3 of xen. but didnt work. |
| 10:44.24 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 10:44.28 | L|NUX | hello every one |
| 10:44.59 | L|NUX | i am using Monitor to record calls but i have noticed that sound is very slow is there any way to increase volume ? |
| 10:47.22 | *** join/#asterisk ITguru (n=ITGuru@webfax.impactteachers.com) |
| 10:48.37 | tzafrir_laptop | joelsolanki, your running kernel is 2.6.18-128.el5xen . Not 2.6.18-128.1.6.el5xen |
| 10:50.57 | *** join/#asterisk Andre101 (n=a@123-243-77-135.tpgi.com.au) |
| 10:53.57 | *** join/#asterisk Xen^ (n=linux@unaffiliated/lnux/x-10290) |
| 10:59.55 | *** join/#asterisk Jimbo12 (n=jim@i-195-137-121-39.freedom2surf.net) |
| 11:00.32 | Jimbo12 | Hi all - I wonder if anyone can help me with OCS Mediation server integration with Asterisk 1.6 please? |
| 11:05.43 | *** join/#asterisk shinao1 (n=shinao1@78.138.29.146) |
| 11:07.34 | *** join/#asterisk arpu (n=arpu@chello062178159144.10.14.univie.teleweb.at) |
| 11:09.18 | *** join/#asterisk Pazzo (n=ugelt@195.254.225.136) |
| 11:20.58 | *** join/#asterisk maximo (n=maximo@CPE001217b1920e-CM00111ade9528.cpe.net.cable.rogers.com) |
| 11:24.18 | *** join/#asterisk ingenius (n=alektro@host90.190-230-73.telecom.net.ar) |
| 11:27.03 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 11:27.56 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 11:33.29 | *** join/#asterisk L|NUX (n=linux@unaffiliated/lnux/x-10290) |
| 11:33.43 | *** join/#asterisk Andre101 (n=a@123-243-77-135.tpgi.com.au) |
| 11:33.55 | Andre101 | Hello, can someone point me in the right direction to fix this? http://www.pastebin.ca/1413127 ? it's not registering.. i can get it to register with a softphone though. |
| 11:40.11 | *** part/#asterisk mintee (i=1000@72-165-177-67.dia.static.qwest.net) |
| 11:46.48 | *** join/#asterisk Andre101 (n=andre101@123-243-77-135.tpgi.com.au) |
| 11:48.26 | Andre101 | Does anyone know why I always get "401 Unauthorized" when trying to register with a voip provider? i have triple checked passwords, still not working |
| 11:49.47 | Chainsaw | There's more to it then passwords. Is your username correct? |
| 11:50.09 | Andre101 | yep |
| 11:52.02 | Andre101 | Any other suggestions? |
| 11:56.16 | wdoekes | is there an account(code) field instead of username perhaps? |
| 11:56.37 | Andre101 | wdoekes: nope, username is my phone number... |
| 11:56.40 | Andre101 | this has worked in the past |
| 11:57.00 | Andre101 | i put the box behind a NAT and it started happening.. |
| 11:57.16 | Andre101 | but i'm getting 401 messages back so I'm not sure if it's the NAT or not |
| 11:57.32 | wdoekes | it might very well respond with 401 to any error |
| 11:58.20 | Andre101 | ok, well.. i added externip and localnet's to sip.conf, port forwarded.. |
| 11:59.00 | wdoekes | why does the server reply with an 172.16.* address? |
| 11:59.05 | wdoekes | is that your lan? |
| 11:59.27 | Andre101 | yea, that's the LAN |
| 11:59.43 | wdoekes | it shouldn't know that, right? |
| 11:59.50 | *** join/#asterisk joseph__ (n=j@212.98.141.199) |
| 12:00.13 | Andre101 | in the contact part, it has the right IP though.. is this what the provider looks at? |
| 12:01.37 | wdoekes | I have no idea :) .. but if an external peer (the provider) knows/sees your LAN IP, that's often a bad thing |
| 12:02.52 | joseph__ | if i use dial with music on hold dial(sip/${EXTEN}@Trunk,30,m) ,if the called party line is closed user will not hear the message from operatior notifying that line is out of service or tem un available ,what to do to solve this problem |
| 12:03.07 | joseph__ | i Use* |
| 12:08.30 | *** join/#asterisk Peregrine (n=danielc@unaffiliated/peregrine) |
| 12:09.12 | joseph__ | kindly advice its a high priority for me |
| 12:09.14 | joseph__ | :) |
| 12:10.41 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 12:12.44 | *** join/#asterisk [TK]D-Fender (n=joe@216.191.106.163) |
| 12:13.55 | Peregrine | Hi, I am trying to interface Asterisk with a ShoreTel phone system. Most things work fine, but I am having a situation where ShoreTel sends Asterisk a REFER response with a Referred-By header and the subsequent INVITE does not have the Referred-By header. Asterisk 1.4.13. Could someone give me a hand on understanding what needs done? |
| 12:16.47 | *** join/#asterisk qdk (n=qdk@195.242.194.42) |
| 12:26.09 | joseph__ | [TK]D-Fender can i talk to the admins |
| 12:26.25 | [TK]D-Fender | joseph__: of...? |
| 12:27.57 | *** join/#asterisk gr0mit (n=tim@extrt.txrx.org.uk) |
| 12:29.13 | joseph__ | [TK]D-Fender my question is above |
| 12:29.40 | [TK]D-Fender | joseph__: Look when I JOINED and realize I dont see your question. |
| 12:29.49 | joseph__ | if i use dial with music on hold dial(sip/${EXTEN}@Trunk,30,m) ,if the called party line is closed user will not hear the message from operatior notifying that line is out of service or tem un available ,what to do to solve this problem |
| 12:30.23 | joseph__ | I use dial with music on hold command |
| 12:30.23 | joseph__ | exten => _X.,n,Dial(SIP/Trunk/${EXTEN}|300|m),I am facing a big problem |
| 12:30.23 | joseph__ | if the called party line is closed or number is incorrect or have a voice mail (Early media 183) user will not hear the message from operator notifying that line is out of service , temporary unavailable
, |
| 12:30.23 | joseph__ | what to do to solve this problem |
| 12:30.46 | joseph__ | this is more clear |
| 12:30.54 | joseph__ | [TK]D-Fender got it |
| 12:32.05 | *** join/#asterisk ccesario (n=ccesario@189-19-9-100.dsl.telesp.net.br) |
| 12:33.26 | [TK]D-Fender | joseph__: http://www.zmonkey.org/blog/node/165 |
| 12:33.37 | ncopa | anyone tried 1.6.2.0-beta with uclibc? |
| 12:34.17 | [TK]D-Fender | joseph__: Ah, that appears to be more like outbound early media |
| 12:34.42 | [TK]D-Fender | joseph__: go ask in #asterisk-dev |
| 12:37.05 | joseph__ | [TK]D-Fender thanks i will test that |
| 12:37.18 | *** join/#asterisk WeazelON (n=deazel@mail2.tikalnetworks.com) |
| 12:37.30 | WeazelON | hey guys, could someone please help me with IVR settings in the FreePBX? i'm trying to put a " * " option to dial to another IVR, and the asterisk is ignoring me pressing the " * ", having "#" or anything else, works fine. |
| 12:37.38 | joseph__ | I use dial with music on hold command |
| 12:37.38 | joseph__ | exten => _X.,n,Dial(SIP/Trunk/${EXTEN}|300|m),I am facing a big problem |
| 12:37.38 | joseph__ | if the called party line is closed or number is incorrect or have a voice mail (Early media 183) user will not hear the message from operator notifying that line is out of service , temporary unavailable
, |
| 12:37.38 | joseph__ | what to do to solve this problem |
| 12:37.48 | [TK]D-Fender | WeazelON: ... |
| 12:37.50 | [TK]D-Fender | ~freepbx |
| 12:37.51 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 12:37.52 | [TK]D-Fender | ^^^^ |
| 12:38.07 | WeazelON | dang :/ |
| 12:38.13 | WeazelON | its dead quite there |
| 12:38.23 | [TK]D-Fender | WeazelON: changes nothing. |
| 12:38.34 | WeazelON | bummer |
| 12:38.41 | [TK]D-Fender | just wait a bit |
| 12:38.48 | beek | [TK]D-Fender: Good morning. Your infobot description of freepbx has gotten nicer... |
| 12:38.52 | joseph__ | [TK]D-Fender that will not stop the MOH |
| 12:39.07 | [TK]D-Fender | 4 minutes early in the Northern hemisphere is not a great time to expect fast results |
| 12:39.23 | [TK]D-Fender | beek: Its been like that for several years |
| 12:39.53 | beek | [TK]D-Fender: I seem to remember it being a bit more caustic. Hmmmm |
| 12:40.09 | [TK]D-Fender | beek: Yes, its been toned down, but for a long time now |
| 12:46.55 | *** part/#asterisk Peregrine (n=danielc@unaffiliated/peregrine) |
| 12:48.30 | *** join/#asterisk ariel_ (i=3fd6eca9@gateway/web/ajax/mibbit.com/x-cbc525882f38b175) |
| 12:49.42 | *** join/#asterisk mort_gib (n=mjensen@177.210.244.195.dsl.static.gibconnect.com) |
| 12:50.35 | *** join/#asterisk jaytee (n=jforde@unaffiliated/jaytee) |
| 12:54.47 | tompaw | hello |
| 12:55.53 | tompaw | I might be dumb and blind, but any ideas why my fresh setup it 1.6.1.0 ignores sip messages completely? |
| 12:56.23 | tompaw | my tcpdump shows incoming udp register/sip messages, netstat shows 5060 listening for connections... |
| 12:56.42 | tompaw | yet still it doesn't respond AT ALL, sip debug is empty, no errors, asterisk up and running |
| 12:58.14 | Stese | Hmm, IpTables? |
| 12:59.05 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 12:59.05 | *** join/#asterisk t3hrealadamd (n=t3hreala@65.215.34.250) |
| 12:59.39 | tompaw | Stese: god bless you. |
| 13:00.10 | Stese | huh? |
| 13:00.15 | tompaw | that was it. |
| 13:00.23 | tompaw | I forgot to configure the firewall |
| 13:00.23 | Stese | oh right... *faints* |
| 13:00.29 | Stese | # |
| 13:01.26 | [TK]D-Fender | no-one ever looks at the big print |
| 13:04.27 | *** part/#asterisk cool^tom (n=thomas@122.166.46.215) |
| 13:04.50 | Katty | BOO! |
| 13:05.01 | Stese | Eak |
| 13:05.38 | Katty | :> |
| 13:06.16 | Katty | it's a gorgeous day outside! |
| 13:06.22 | Katty | the sun is shining! the birds are singing! |
| 13:06.31 | Stese | the tank is clean? |
| 13:06.41 | Katty | no, the tank is maxing out his defense. |
| 13:06.59 | file | tackles Katty |
| 13:07.01 | Katty | possibly dodge and parry, if he's defense capped. |
| 13:07.07 | Katty | HAI FILE |
| 13:07.10 | Katty | hugs on file |
| 13:07.15 | Stese | lol... I thought you were quoting Finding Nemo |
| 13:07.23 | file | continues to eat his morning muffin |
| 13:07.30 | Stese | btw... are you able to answer this question... |
| 13:07.35 | Katty | file: i'm having blueberry muffin top, cereal |
| 13:07.38 | Stese | I'm trying to debug a * ISDN issue, and i'm just wondering what "Cannot hangup chan, no ast" means? |
| 13:07.45 | file | Katty: yum |
| 13:08.07 | Katty | Stese: not sure. |
| 13:08.31 | Katty | jaytee: :> |
| 13:08.34 | Katty | pamples jaytee |
| 13:10.16 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 13:10.24 | Stese | ok... no worries |
| 13:11.19 | [TK]D-Fender | Katty: Mew. |
| 13:11.20 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 13:11.29 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 13:12.08 | *** join/#asterisk mikkel (n=mikkel@84-238-113-66.u.parknet.dk) |
| 13:12.36 | tompaw | did anything change in 1.6.1 regarding users.conf? |
| 13:13.00 | tompaw | it might be my unlucky day, but it keeps saying that no matching peer found. |
| 13:13.03 | Katty | hugs jaytee |
| 13:13.06 | Katty | hugs [TK]D-Fender |
| 13:13.12 | tompaw | and the peer is there - right there in users.con |
| 13:13.15 | tompaw | f |
| 13:13.54 | [TK]D-Fender | tompaw: perhaps you could read the lovely docs included with the tarball. And look at "sip show peers", and maybe I dunno... show us your configs and the CLI output of your complete failed attempt |
| 13:14.17 | [TK]D-Fender | tompaw: So far I'd very comfortably write it off as "user error" |
| 13:15.57 | tompaw | [TK]D-Fender: you are right, it is a user error - my users.conf is completely ignored |
| 13:16.15 | tompaw | its entries are not added to peer list. |
| 13:17.03 | *** join/#asterisk beherit (n=albert@netsys.bts.corp.amdatex.net) |
| 13:17.18 | beherit | ladies and gents, any good voip provider that you know |
| 13:17.29 | tompaw | http://pastebin.com/m63488bb9 |
| 13:19.42 | Jacke | tompaw: what are you doing here? ;-) |
| 13:19.43 | [TK]D-Fender | tompaw: I don't see a "hassip = yes" in three |
| 13:19.50 | [TK]D-Fender | there* |
| 13:19.52 | [TK]D-Fender | SMRT |
| 13:20.04 | tompaw | [TK]D-Fender: there is one in general |
| 13:20.10 | [TK]D-Fender | towEW |
| 13:20.25 | *** join/#asterisk MrNaz (n=mrnaz@ppp121-44-198-120.lns10.mel4.internode.on.net) |
| 13:20.30 | [TK]D-Fender | tompaw: Also no TYPE |
| 13:20.39 | [TK]D-Fender | tompaw: that = DOA |
| 13:20.45 | Stese | was going to mention type :P |
| 13:21.39 | tompaw | thanks. |
| 13:21.55 | *** join/#asterisk defiancenl (n=jefrecha@m0n0.hix.nl) |
| 13:23.40 | Stese | tompaw > you might want to have a look at the sample files, if you have them :) |
| 13:25.09 | tompaw | Stese: good idea, looks like my syntax is terribly outdated. |
| 13:26.40 | *** part/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 13:28.46 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 13:30.27 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:32.43 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 13:34.28 | eppigy | hello |
| 13:34.31 | eppigy | I am dave |
| 13:35.02 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 13:36.30 | Katty | hugs eppigy |
| 13:36.40 | [TK]D-Fender | tompaw: Type has always been required |
| 13:36.45 | Corydon76-dig | Dave's not here! |
| 13:36.52 | eppigy | :D |
| 13:37.14 | *** join/#asterisk bminish (n=bminish@pdpc/supporter/professional/bminish) |
| 13:40.42 | Great_Anta_Baka | how can I force asterisk to generate a ringing tone? I have it connected to a voip router which has a built in ATA |
| 13:40.42 | Katty | is watching nigella bites videos. |
| 13:41.04 | *** join/#asterisk Khratos (n=khratos@190.166.103.111) |
| 13:41.51 | Great_Anta_Baka | and the ATAs have an initial dial tone, but no ringing tone when the call is being made |
| 13:43.34 | eppigy | jonesin for a rockstar |
| 13:45.14 | [TK]D-Fender | Great_Anta_Baka: "core show application dial" <- I'm sure you'll find it |
| 13:45.28 | *** join/#asterisk youngproguru (n=quassel@74.10.229.45) |
| 13:45.40 | youngproguru | Good Morning |
| 13:46.03 | Katty | hihi |
| 13:46.37 | Jimbo12 | hello - can anyone help me with a problem I have having with a SIP Peer using TCP... |
| 13:47.01 | Jimbo12 | please :) |
| 13:47.19 | Great_Anta_Baka | ty tk |
| 13:49.01 | *** join/#asterisk clintc (n=clintc@n128-227-55-39.xlate.ufl.edu) |
| 13:49.03 | [TK]D-Fender | Great_Anta_Baka: Glad you found it. You're welcome |
| 13:50.09 | *** join/#asterisk MaliutaLap (n=biteme@kiev.lusan.id.au) |
| 13:51.34 | *** join/#asterisk defsdoor (n=andy@82.133.90.135) |
| 13:51.58 | tzafrir_laptop | Great_Anta_Baka, the ATA is the one generating the dial tone |
| 13:52.22 | Great_Anta_Baka | but it doesnt generate one :( |
| 13:52.46 | Great_Anta_Baka | there is dial tone |
| 13:52.51 | Great_Anta_Baka | but no ringing tone |
| 13:54.54 | Great_Anta_Baka | 877547501@DID_XXX:1] Dial("SIP/877547602-08f1b988", "SIP/7501|45|tr check i have asterisk forcing the dial tone.. but nothing .. not even on inbound calls :( |
| 13:55.50 | Great_Anta_Baka | ah i see i had progressinband=yes for incoming calls |
| 13:56.22 | Great_Anta_Baka | well defined in the trunk |
| 13:57.12 | *** join/#asterisk Stese (n=Someone@adsl.ntsols.com) |
| 13:57.23 | *** join/#asterisk bradleyprice86 (n=bradleyp@fw.datafax.net) |
| 13:57.53 | Stese | Grr Silly VPN's |
| 13:58.03 | Stese | I gather I didn't miss anything |
| 14:02.16 | Katty | hi. |
| 14:03.58 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 14:07.24 | *** join/#asterisk putnopvut (n=putnopvu@asterisk/master-of-queues/mmichelson) |
| 14:07.24 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:08.40 | eppigy | EVERY PASSING MOMENT IS A CHANCE TO TURN IT ALL AROUND |
| 14:09.41 | *** part/#asterisk ncopa (n=ncopa@ti211310a081-5598.bb.online.no) |
| 14:09.42 | Katty | turn around.... |
| 14:09.59 | eppigy | BRIGHT EYES |
| 14:10.45 | Katty | every now and then i fall apart! |
| 14:11.00 | [TK]D-Fender | load chan_dumpty.so |
| 14:11.53 | Katty | blargh |
| 14:11.59 | *** join/#asterisk moy (n=moy@74.12.124.89) |
| 14:12.13 | BlargMaN00 | Katty: no profound use of my name!!! 8)~ lol |
| 14:12.24 | Katty | oh |
| 14:12.28 | Katty | heh |
| 14:12.30 | BlargMaN00 | Katty: j/k |
| 14:14.34 | eppigy | D: |
| 14:15.16 | Katty | yummy french toast recipe on youtube |
| 14:15.21 | Katty | i wanna go home and bake |
| 14:15.45 | *** part/#asterisk joseph__ (n=j@212.98.141.199) |
| 14:16.27 | *** join/#asterisk telnettech (i=telnette@gw.percipia.com) |
| 14:16.33 | telnettech | good morning |
| 14:16.55 | Stese | Hi |
| 14:17.16 | telnettech | besides in the looger.conf file, is there anything else i need to 'turn on' for logging to work? I know i need to do a logger reload on the CLI |
| 14:20.28 | *** join/#asterisk Deeewayne (n=dwayne@75.76.254.162) |
| 14:20.28 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 14:21.42 | Katty | eppigy: let's go home and make cookies! |
| 14:21.45 | Katty | or brownies. |
| 14:21.49 | *** join/#asterisk watchy (n=watchy@76.196.98.139) |
| 14:22.05 | Katty | chocolate pudding? |
| 14:22.06 | Katty | lava cake |
| 14:22.23 | Katty | corn pudding |
| 14:22.24 | *** join/#asterisk axisys (n=axisys@155.70.141.45) |
| 14:23.55 | jaytee | sorry, I was repairing a Windows install and missed the hugs and the pampling |
| 14:24.47 | watchy | anyway to reboot a polycom from web interface? |
| 14:25.09 | *** join/#asterisk wonderworld (n=ww@ip-62-143-16-28.unitymediagroup.de) |
| 14:27.06 | *** join/#asterisk SuPrSluG (n=SuPrSluG@firewall-a.buf.ny.i-evolve.net) |
| 14:29.29 | Katty | watchy: pretend to make a change, then hit submit |
| 14:30.53 | *** join/#asterisk Stese (n=Someone@adsl.ntsols.com) |
| 14:31.01 | watchy | ah |
| 14:35.05 | [TK]D-Fender | axe-murders watchy |
| 14:35.33 | *** join/#asterisk op3r (n=op3r@114.108.201.142) |
| 14:36.18 | op3r | hello |
| 14:36.37 | op3r | can any one tell me if this is the cause of asterisk crashing? or what kind of error is this? |
| 14:36.38 | op3r | May 5 07:33:57 WARNING[13065] channel.c: Thread 114228112 Blocking 'SIP/siptrunk-09a2fce0', already blocked by thread 43416464 in procedure ast_waitfor_nandfds |
| 14:36.38 | op3r | May 5 07:33:57 WARNING[13065] channel.c: Thread 114228112 Blocking 'Local/58600057@default-e18f,2', already blocked by thread 28203920 in procedure ast_waitfor_nandfds |
| 14:37.49 | op3r | then i get disconnected from the cli |
| 14:37.57 | op3r | :( |
| 14:39.20 | Great_Anta_Baka | is asterisk still running after you get disconnected |
| 14:39.21 | Great_Anta_Baka | ? |
| 14:39.52 | op3r | this is what I see when I get disconnected |
| 14:40.06 | op3r | May 5 07:36:48 NOTICE[28278] dnsmgr.c: Managed DNS entries will be refreshed every 300 sec onds. |
| 14:40.07 | op3r | May 5 07:36:48 NOTICE[28278] cdr.c: CDR simple logging enabled. |
| 14:40.07 | op3r | May 5 07:36:48 WARNING[28278] pbx_config.c: Unable to register extension at line 118 |
| 14:40.07 | op3r | May 5 07:36:48 WARNING[28278] config.c: Unknown directive '' at line 1 of /etc/asterisk/za |
| 14:40.26 | op3r | but to to answer your question yes it is running |
| 14:41.47 | Great_Anta_Baka | can you kill asterisk and start it with asterisk -cvvvvvvvvvvv and the see if there are any errors on startup? |
| 14:43.22 | eppigy | Katty: i like cookies |
| 14:43.31 | eppigy | what kind of cookies do you make? |
| 14:44.09 | *** join/#asterisk jerlique (i=d208a802@gateway/web/ajax/mibbit.com/x-43ae110b68d82752) |
| 14:44.21 | op3r | Great_Anta_Baka: i only see an error on zapata.conf but its not that a biggie |
| 14:46.30 | *** join/#asterisk jasonwoot (n=some@bookit-dev.com) |
| 14:46.54 | Katty | eppigy: well my favorite is chocolate peanut butter chip cookies. |
| 14:46.57 | jerlique | If my sip registrations time out, they dont re-authenticate. The state is in "No Authentication". In the sip.conf file I have setup the registerattempts=0 but it doesnt seem to work. Any udeas? |
| 14:47.32 | Katty | eppigy: but snickerdoodle sounds pretty good right about now |
| 14:47.49 | Katty | eppigy: or maybe forget cookies entirely, and make a really really gooey buttercake |
| 14:48.18 | Katty | eppigy: OR! |
| 14:48.23 | Katty | eppigy: waffles :> |
| 14:55.30 | eppigy | :D |
| 14:55.34 | eppigy | i like waffles |
| 14:55.41 | eppigy | with melted butter and syrup |
| 14:55.45 | eppigy | 8[] |
| 14:56.14 | *** part/#asterisk gego (n=rick@b238085.customer.hansenet.de) |
| 14:57.48 | jasonwoot | happy cinco de mayo and stuff |
| 14:58.11 | *** join/#asterisk Defraz (n=T0tal@fw-poky.fuzecore.com) |
| 15:00.21 | Katty | hey |
| 15:00.27 | Katty | it IS may 5th |
| 15:00.36 | Katty | forget waffles, i'm gonna make mexican! |
| 15:00.45 | *** join/#asterisk spck (n=spck@unioncab.com) |
| 15:00.51 | Great_Anta_Baka | op3r, does it still knock you out when you run it like that? |
| 15:02.08 | watchy | i need hugs |
| 15:03.31 | jasonwoot | exten => 1,2,HugAsteriskUser(Watchy|${reacharound}) |
| 15:04.24 | Katty | hugs watchy |
| 15:05.00 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 15:07.56 | eppigy | haha |
| 15:08.27 | eppigy | TRABAJO |
| 15:10.09 | *** join/#asterisk ayeso (n=chatzill@ext-52.sagetelecom.net) |
| 15:11.27 | Katty | it's terrible to be stuck inside today. |
| 15:12.00 | ayeso | When i call an extension that has asterisk comedian voicemail, I want to be able to press star during the greeting to interupt and be prompted for the password, I am having trouble finding any info on this. does anyone know if this is possible? |
| 15:12.52 | eppigy | yeah dude |
| 15:12.56 | eppigy | I am sleepy |
| 15:13.05 | eppigy | tryna cut back on caffeine |
| 15:16.04 | Katty | good luck with that |
| 15:17.11 | [TK]D-Fender | ayeso: Go read the docs on Asterisk Standard Extensions. "a" <-------- |
| 15:18.05 | ayeso | [TK]D-Fender: will do, BTW I found where in the source to change the prompt keys yesterday. pretty strait forward. thx |
| 15:22.08 | eppigy | need energy |
| 15:22.15 | eppigy | something to fuel the rage |
| 15:22.34 | Katty | how about a nice pillow, and a blanket. |
| 15:22.37 | Katty | and you can go nap for about 2 hours. |
| 15:22.50 | eppigy | man |
| 15:22.52 | eppigy | that would be awesome |
| 15:22.55 | Katty | then you can wake up and have a shower. |
| 15:23.04 | Katty | possibly consider some lunch. |
| 15:23.09 | jjshoe | ayeso it's possible |
| 15:23.10 | eppigy | i will naap in the shower too |
| 15:23.19 | Katty | i've done that before. |
| 15:23.22 | Katty | especially when ill )= |
| 15:23.26 | eppigy | yes |
| 15:23.30 | Katty | it's nice. |
| 15:23.36 | ayeso | jjshoe: TK shoved me in the right direction, got it wokring. |
| 15:23.38 | eppigy | I did when i was young and would stay up all night |
| 15:23.44 | eppigy | and go to school the next day |
| 15:24.00 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 15:24.01 | Katty | ugah. |
| 15:24.06 | Katty | can't pull all nighters anymore |
| 15:24.13 | eppigy | me either :[ |
| 15:24.14 | *** join/#asterisk ayeso (n=chatzill@ext-52.sagetelecom.net) |
| 15:24.25 | Katty | sleeeeep good |
| 15:24.30 | jjshoe | I like to pull on it all night long |
| 15:24.32 | jjshoe | err um, oops. |
| 15:24.34 | eppigy | im hurting at 5am |
| 15:24.38 | *** join/#asterisk UQlev (n=yuriy@nb11-125.static.cytanet.com.cy) |
| 15:24.48 | Katty | i'm already passed out at 2am |
| 15:24.50 | eppigy | jjshoe: haha |
| 15:25.05 | eppigy | i normaly am |
| 15:25.13 | eppigy | but if something has me captivated |
| 15:25.25 | Katty | yeah that's true. |
| 15:25.27 | eppigy | I will stay awake till the first signs of dawn |
| 15:25.30 | eppigy | then im like |
| 15:25.31 | eppigy | D: |
| 15:26.07 | spck | v-v |
| 15:26.10 | *** join/#asterisk bakermd (n=bakermd@38.101.225.215) |
| 15:26.17 | *** join/#asterisk stentroad (n=stentroa@201.80.216.82) |
| 15:26.26 | *** part/#asterisk bakermd (n=bakermd@38.101.225.215) |
| 15:27.27 | spck | anyone have experience setting up * with dundi doing the load balancing? |
| 15:27.51 | *** join/#asterisk matsk (n=matkar@c-fe89e253.174-6-64736c10.cust.bredbandsbolaget.se) |
| 15:28.03 | [TK]D-Fender | spck: DUNDI isn't a load-balancing toold really. its for peering to other system for extens they may host |
| 15:28.26 | eppigy | with a realtime db config you can use it for that though |
| 15:28.35 | *** part/#asterisk matsk (n=matkar@c-fe89e253.174-6-64736c10.cust.bredbandsbolaget.se) |
| 15:28.37 | eppigy | and an sip proxy to balance registrations |
| 15:29.30 | *** join/#asterisk Carlos_PHX (n=Carlos@ip68-3-162-244.ph.ph.cox.net) |
| 15:30.05 | spck | that's kind of what i want to do |
| 15:30.17 | spck | the sip proxy being something like opensips? |
| 15:30.25 | eppigy | openser etc. |
| 15:30.40 | spck | opensips != openser? |
| 15:30.47 | eppigy | sorry |
| 15:30.50 | eppigy | yeha that is the new openser |
| 15:30.52 | eppigy | evidently |
| 15:31.12 | eppigy | these open source projects |
| 15:31.17 | spck | is the sip proxy then a single point of failure? |
| 15:31.17 | eppigy | and their crazy name changing |
| 15:31.30 | eppigy | spck: only if you have one |
| 15:31.30 | spck | or is that something you can easily cluster as well? |
| 15:32.56 | jaytee | of the two forks, which fork is most like the original openser? |
| 15:33.08 | jaytee | Kamailio or OpenSIPS? |
| 15:33.23 | eppigy | [TK]D-Fender: what would you recommend for high availability/load balancing? |
| 15:34.34 | [TK]D-Fender | eppigy: No direct experience, but I've seen RR-DNS, SER thrown in the mix |
| 15:34.45 | eppigy | yesh |
| 15:35.32 | eppigy | thats why i was hopign for the asterisk cvookbook |
| 15:35.46 | eppigy | that is liek the #1 closely guarded secret |
| 15:35.55 | eppigy | real asterisk HA/Loda Balancing |
| 15:36.30 | spck | yea i can't find crap for documentation on it |
| 15:37.03 | jasonwoot | I gave up on load balancing... clonezilla |
| 15:38.26 | *** join/#asterisk CunningPike (n=CunningP@204.239.10.119) |
| 15:40.52 | *** join/#asterisk dandre (n=daniel@was59-3-82-236-48-30.fbx.proxad.net) |
| 15:41.16 | spck | clonezilla? |
| 15:42.02 | *** join/#asterisk fskrotzki_ (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
| 15:43.18 | Katty | plots carne asada |
| 15:45.30 | eppigy | 8[steak] |
| 15:46.34 | jasonwoot | spck: it's like ghost 4 linux.. just image your entire box and create cold running spare |
| 15:46.49 | ayeso | [TK]D-Fender: do you know if there is a way to detect voice on a channel? I'm trying to find a way where in a meetme conference you would be able to detect which participant is talking. |
| 15:47.01 | *** join/#asterisk jerlique (i=d208a802@gateway/web/ajax/mibbit.com/x-43ae110b68d82752) [NETSPLIT VICTIM] |
| 15:47.01 | *** join/#asterisk axisys (n=axisys@155.70.141.45) [NETSPLIT VICTIM] |
| 15:47.01 | *** join/#asterisk fnordus (n=dnall@70.71.225.48) [NETSPLIT VICTIM] |
| 15:47.24 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 15:47.24 | *** mode/#asterisk [+o denon] by ChanServ |
| 15:48.20 | Katty | woah. i like this guy. he's putting a jar of pepperchinis, a bottle of corona, and the carne asada meat in a big ziplock bag |
| 15:48.53 | [TK]D-Fender | ayeso: "help meetme" , "core show application meetme" |
| 15:49.20 | *** join/#asterisk Torrieri (n=Torrieri@nelug/crew/torrieri) |
| 15:49.26 | *** join/#asterisk JerJer (n=PhatJ@asterisk/original-h323-guy/JerJer) |
| 15:49.27 | *** join/#asterisk kc8pxy (n=gecko@99-182-113-98.lightspeed.clmboh.sbcglobal.net) [NETSPLIT VICTIM] |
| 15:49.32 | *** join/#asterisk blackest_mamba (n=blackest@71.239.160.143) [NETSPLIT VICTIM] |
| 15:49.38 | *** join/#asterisk marv[work] (n=timr@router.asteriasgi.com) |
| 15:50.13 | ayeso | [TK]D-Fender: got it thanks. |
| 15:50.13 | eppigy | that sound pretty good |
| 15:51.26 | Katty | http://www.youtube.com/watch?v=ioi9K8Cn0Dw <- awesome |
| 15:51.44 | *** join/#asterisk vgster (n=vgster@94-194-190-189.zone8.bethere.co.uk) |
| 15:52.07 | Katty | i can't believe that marinade. |
| 15:52.21 | *** join/#asterisk micols (n=mio@rlogin.dk) |
| 15:52.21 | Katty | jar of jalepinos, bottle of beer, meat., |
| 15:52.33 | eppigy | teh bandana is killing it :{ |
| 15:52.51 | Katty | ignore the bandana |
| 15:53.08 | Katty | keep the onion wet and pound the meat! >.< |
| 15:53.08 | Katty | omg |
| 15:53.52 | *** join/#asterisk scardinal (n=supreme@0905ds1-rdo.0.fullrate.dk) [NETSPLIT VICTIM] |
| 15:53.58 | spck | i got a bbq coming up that's a pretty good idea |
| 15:54.26 | cjk | how can i debug hangups and check which leg is responsible for the hangup for calls from iax to zap |
| 15:54.29 | Katty | not sure how meat marinated in beer would taste |
| 15:54.52 | Katty | but worth a shot i guess. afterall, i use amber ultra in my refried beans |
| 15:54.54 | *** join/#asterisk micols (n=mio@rlogin.dk) |
| 15:55.00 | eppigy | steaks marinated in beer and grilled tastes great |
| 15:55.24 | Katty | ryan likes both pepperchinis and beer |
| 15:55.32 | Katty | so this might be a big hit with him |
| 15:55.45 | Katty | all else fails, riddick can have it |
| 15:56.41 | *** join/#asterisk haryv (n=lanny@S010600a0c93f6f7e.vs.shawcable.net) |
| 15:57.11 | eppigy | haha |
| 15:57.20 | eppigy | yeah pepperchinis are max_ausome |
| 15:58.22 | *** join/#asterisk bpgoldsb (n=bpgoldsb@209.208.6.182) [NETSPLIT VICTIM] |
| 16:00.12 | *** join/#asterisk angler (n=angler@pdpc/sponsor/digium/angler) |
| 16:00.12 | *** mode/#asterisk [+o angler] by ChanServ |
| 16:01.18 | Katty | eppigy: think some nice hunks of avocado would go well with that? |
| 16:01.48 | *** join/#asterisk mcargile (n=mikec@rrcs-24-173-156-170.se.biz.rr.com) |
| 16:02.42 | eppigy | i love avacdo too |
| 16:02.47 | eppigy | D: |
| 16:02.49 | Katty | are they high in fat? |
| 16:02.55 | eppigy | well I am not sure |
| 16:03.04 | eppigy | they are high in delicious |
| 16:05.23 | Nugget | avocado is the good kind of fat. |
| 16:05.53 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 16:05.53 | eppigy | YES |
| 16:07.22 | *** join/#asterisk KavanS (n=KavanS@static-71-117-242-28.ptldor.dsl-w.verizon.net) |
| 16:07.22 | eppigy | monounsaturated fat |
| 16:08.21 | *** join/#asterisk vader-- (n=me@c-68-36-9-8.hsd1.nj.comcast.net) |
| 16:09.01 | *** join/#asterisk amaache (i=amaache@196.46.252.50) |
| 16:10.09 | *** join/#asterisk joseph__ (n=j@212.98.141.199) |
| 16:10.11 | joseph__ | app_waitforring.so what does exactly means |
| 16:10.16 | joseph__ | or do |
| 16:14.32 | *** join/#asterisk paulproteus (i=paulprot@2002:db69:2513:0:0:0:0:1) |
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| 16:24.41 | *** join/#asterisk joesuffceren (n=chatzill@ip68-104-166-24.ph.ph.cox.net) |
| 16:25.12 | joesuffceren | I have an issue with a remote sip extension that is capable of placing receiving calls with 2way audio, but the audio is choppy. I have other remote sip extensions without those issues. I've tried g711 and g729 (both of which are in use with other remote sip extensions with no issues) |
| 16:25.26 | joesuffceren | her internet connection is (ostensibly) 5 down 1 up which should be plenty for 1 concurrent voip call. any ideas on improving the situation? |
| 16:28.23 | *** join/#asterisk anonymouz666 (n=anonymou@189.24.118.128) |
| 16:28.47 | *** join/#asterisk shinao1 (n=shinao1@78.138.29.146) |
| 16:29.15 | UQlev | joesuffceren: she must try different sip-clients, e.g zoiper I found it rather reliable |
| 16:29.59 | UQlev | joesuffceren: most voip-clients are sensitive ot other applications running on the same host |
| 16:33.27 | *** join/#asterisk nullable_type (n=nullable@hq.verbx.net) |
| 16:35.23 | joesuffceren | UQlev: It's actually a cisco 7940 endpoint, not a soft phone on a PC |
| 16:36.00 | joesuffceren | and the remote extensions I referred to are at other locations |
| 16:36.56 | joesuffceren | I'm pretty sure it will end up being just an issue with her crappy internet connection (Charter in a rural area), but I was just wanting to make sure there weren't any other options I could try before giving up |
| 16:37.49 | joesuffceren | I honestly wouldn't be surprised if it's charter screwing around with QoS to make voip traffic perform poorly to force people into their phone service |
| 16:37.52 | UQlev | joesuffceren: does she use wireless? |
| 16:38.23 | joesuffceren | she has a wireless router, but the phone is connected directly to the router (no wireless links between the phone and the internet connection) |
| 16:40.28 | UQlev | joesuffceren: but she could try softphone on her notebook just to check |
| 16:41.29 | UQlev | joesuffceren: for my remote clients I prefere to make IAX2 accounts |
| 16:41.33 | nullable_type | Hey guys i have problem having CURL working with asterisk, CURL is not listed in asterisk "core show functions" but i have installed curl and configured -with-curl. This is debian, I had no problem configuring CURL with ubuntu |
| 16:41.57 | joesuffceren | UQlev: good thought. I do have xlite installed on her notebook. This phone was taken out of service as a remote SIP extension at another location, though, and was working perfectly there. |
| 16:42.18 | joesuffceren | UQlev: there isn't IAX firmware for Cisco 79xx phones, is there? |
| 16:42.48 | UQlev | joesuffceren: I have no ideas about voip h/w devices |
| 16:43.23 | nullable_type | Can someone help me |
| 16:43.55 | spck | give it a minute |
| 16:44.43 | joesuffceren | UQlev: no worries. thanks for the thought about trying a softphone. I'll give that a shot, but I think I'm going to end up having to give in and buy a $20 per month phone service from charter. GAG |
| 16:49.01 | nullable_type | Can someone help me configuring CURL with asterisk....... it seems to miss function CURL in its list even though CURL is installed and cofngured -with-curl |
| 16:51.37 | seanbright | you have the libcurl-dev package installed? |
| 16:52.04 | nullable_type | isnt it included in curl download from curl.haxx.se |
| 16:52.12 | seanbright | ah, you built from souce. |
| 16:52.16 | nullable_type | yes |
| 16:52.17 | seanbright | pastebin your config.log |
| 16:52.18 | seanbright | ~pb |
| 16:52.19 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/ |
| 16:52.35 | seanbright | or just look through it to see what is failing when it is looking for curl |
| 16:52.59 | nullable_type | alright, please hang on |
| 16:58.23 | nullable_type | sean |
| 16:58.27 | nullable_type | just pasting now |
| 16:58.30 | nullable_type | almost there |
| 17:01.02 | seanbright | i recommend downloading http://pastebin.ca/download/paste2pastebin.pl |
| 17:01.07 | seanbright | that makes life easier |
| 17:01.22 | nullable_type | http://pastebin.com/d7dccd8d2 |
| 17:01.34 | nullable_type | can u put ur thoughts here, i gotto stop away for a bit but i will come back and check |
| 17:01.37 | nullable_type | thanks sean |
| 17:02.36 | seanbright | nullable_type: according to this, configure has found curl |
| 17:02.43 | seanbright | nullable_type: are you enabling it in 'make menuselect'? |
| 17:10.48 | *** join/#asterisk ecrist (n=ecrist@mr.garrison.secure-computing.net) |
| 17:11.15 | ecrist | hey folks. is there such a thing as a VoIP-GSM gateway? How does it work? |
| 17:12.34 | ecrist | we've got one extension we need to serve that gets great cellular reception, but there is no good internet service available, aside from ISDN |
| 17:15.59 | Chainsaw | ecrist: Perhaps data-over-GSM is an option? |
| 17:16.11 | nkohh | ps aux |
| 17:16.12 | nkohh | oops, sorry' |
| 17:16.14 | Chainsaw | ecrist: You can then run VoIP over the resulting data path. "Mobile" broadband. |
| 17:16.15 | ecrist | we're trying that now, latency is way to high |
| 17:16.27 | ecrist | s/trying/doing/ |
| 17:16.49 | nullable_type | sean >> Yes i enabled it in make menuselect |
| 17:17.07 | nullable_type | i have done these steps in UBUNTU where it worked perfectly fine, its just that in debian it doesn't seems to |
| 17:21.53 | *** join/#asterisk shinao1 (n=shinao1@78.138.29.146) |
| 17:22.49 | leifmadsen | http://blogs.computerworld.com/disk_image_backups_and_microsoft_office <-- just good advice in general (this is not a rick-roll or anything :)) |
| 17:32.33 | eppigy | nap time |
| 17:37.25 | *** join/#asterisk Victor_Yure (n=victor@unaffiliated/victoryure/x-837844) |
| 17:37.43 | nullable_type | sean >> If you are around, i found the answer for CURL problem |
| 17:37.49 | nullable_type | thanks anyways |
| 17:42.32 | rene- | hey |
| 17:42.54 | rene- | would you say that the term POTS encompasses both analog and digital telephony worlds? |
| 17:43.01 | rene- | and excludes anything voip |
| 17:46.37 | ayeso | rene-: I really only use the term to descibe "plain old telephone service" a regular line from a provider that is terminated at a co somewhere |
| 17:47.24 | rene- | hmm |
| 17:47.29 | ayeso | rene-: I would not say that it is digital |
| 17:47.37 | rene- | TDM seems only applicable to digital |
| 17:47.59 | rene- | and not analog |
| 17:48.04 | ayeso | rene-: thats right, look up ISDN PRI and ISDN BRI |
| 17:48.26 | ayeso | also look up what a channelized T1 is |
| 17:49.03 | rene- | i am familiar with those terms, i just wanted a word to describe traditional telephony technologies versus voip based ones |
| 17:49.22 | ayeso | oh, just say TDM then |
| 17:49.22 | rene- | i guess traditional telephony is the word |
| 17:49.34 | *** join/#asterisk pikachu2000 (n=pikachu2@196-209-196-147-rbry-bb-1.dynamic.isadsl.co.za) |
| 17:49.49 | *** part/#asterisk pikachu2000 (n=pikachu2@196-209-196-147-rbry-bb-1.dynamic.isadsl.co.za) |
| 17:49.55 | KyleK | dundi looks interesting, is it used more or less than enumeration? or do people that use dundi lookups for outgoing usually use e164.org as well? |
| 17:50.00 | [TK]D-Fender | PSTN <- |
| 17:50.42 | rene- | D-Fender: would you say PSTN encompasses but analog and digital? |
| 17:50.46 | rene- | s/but// |
| 17:51.35 | ayeso | rene-: the PSTN is both VOIP and TDM |
| 17:51.45 | rene- | it is? |
| 17:51.53 | ayeso | rene-: yes. |
| 17:52.07 | [TK]D-Fender | rene-: PSTN = the entire global telephony network. |
| 17:52.33 | [TK]D-Fender | and No, VoIP is a transport over IP. Yuo cannot just grab someones POTS phone and "dial VoIP" |
| 17:53.02 | ayeso | rene-: look up ATT AVOICS - this is their VOIP/SIP network. They transport al ot of calls via VOIP now. |
| 17:53.32 | [TK]D-Fender | If I have 2 * servers at different location and phones on eachs ide directly connected to each box and I setup dialplan between them, this has NOTHING to do with the PSTN so far |
| 17:54.36 | ayeso | agreed |
| 17:55.27 | *** join/#asterisk cvnet (n=dahitler@24.156.136.205) |
| 17:55.33 | cvnet | hello all |
| 17:55.37 | [TK]D-Fender | the only time to make any kind of implied relationship is when you use a piece of equipment that takes a VoIP channel and terminates it to a TDM channel on the PSTN. |
| 17:56.11 | [TK]D-Fender | So if one * server uses a channel bank with analog phones, sure thats TDM, but its not the PSTN. |
| 17:56.23 | [TK]D-Fender | **PUBLIC** Switched Telephone Network. |
| 17:57.19 | ayeso | I agree, but the PSTN also encompasses long distance calls, you may pick up your phone at home and place a LD call, this call may be VOIP at some point along the way. |
| 17:58.06 | *** part/#asterisk stentroad (n=stentroa@201.80.216.82) |
| 17:58.17 | *** join/#asterisk theHub (n=theHub@69.177.93.21) |
| 17:58.30 | [TK]D-Fender | ayeso: drop the words "long distance calls" off a cliff. Its a worthless term that only has some conceptual tie to how you will be BILLED |
| 17:59.21 | [TK]D-Fender | ayeso: And for all you know the telco has 2 speakerphones in the same room and are using OPEN AIR AUDIO to "bridge your call. So who cares if "VoIP" occurs somewhere in the middle? |
| 17:59.55 | [TK]D-Fender | ayeso: their use of VoIP is invisible to you |
| 18:00.04 | KyleK | hehe I think audiophiles would notice if that was being used on a large scale |
| 18:00.11 | ayeso | [TK]D-Fender: I disagree, when you place a LD call, depending who you have for long distance, your call will be sent to a session border controller that makes a decision on where to send your call based on LCR (least cost routing) this is usually VOIP. |
| 18:00.13 | florz | [TK]D-Fender: I do ... it produces much less echo than your proposed open air method :-> |
| 18:00.30 | [TK]D-Fender | ayeso: You cannot directly hok to anything meaningful in there. |
| 18:00.46 | ayeso | [TK]D-Fender: well she seems to care, she asked if the pstn was VOIP, and the anser is yes it is also voip |
| 18:01.06 | [TK]D-Fender | ayeso: that your carrier has a number of possible carrier and they can choose whose trunk to use at variable billing is again invisible to you and hence not a physical "thing" |
| 18:01.08 | *** join/#asterisk af_ (n=getsmart@88-149-230-139.dynamic.ngi.it) |
| 18:01.29 | [TK]D-Fender | gathers some tin cans & string |
| 18:01.44 | spck | I'm getting this when trying to setup RealTime: res_config_mysql.c:317 realtime_mysql: MySQL RealTime: Invalid database specified: asterisk |
| 18:01.52 | ayeso | [TK]D-Fender: i dont see your argument? I thought we were letting her know if the PSTN was VOIP or not. YES it is VOIP at certain pionts. |
| 18:02.05 | spck | I've set this up before and it worked, any idea what i'm missing that it can't find the db? |
| 18:02.37 | [TK]D-Fender | ayeso: it can be but its closed off to you and not part of any formal infrastructue that yo can hook into. So again "who cares'? |
| 18:03.18 | ayeso | [TK]D-Fender: well I do, I hooked into the AVOICS network with my VERSO SBC last week. |
| 18:03.55 | [TK]D-Fender | ayeso: Guess the PSTN is also "voip" if I use an analog POTS circuit to the telco, use a speakerphone on it, and have a speaker & mic on my computer and use SKYPE toa family memeber |
| 18:03.56 | ayeso | If someone asks if the PSTN is VOIP the answer is yes. |
| 18:04.23 | [TK]D-Fender | ayeso: And at some point that hits TDM. That is the key. SOMEONE is acting as the TDM gateway. |
| 18:04.30 | ayeso | [TK]D-Fender: hmmm.. thats silly TK |
| 18:04.43 | kc8pxy | ayeso: perhaps it is on certain carriers. but from the point of your asterisk server,once it's outside your direct channel that connects to the PSTN, it doesn't matter.. it's their job to give you phone. unless it pertains to how you connect to it, it should not matter to you/them. |
| 18:04.44 | [TK]D-Fender | ayeso: someone's upfront equipement is irrelevent. its the backend that matters. |
| 18:05.41 | ayeso | If you are only looking at it from the the point of view of an * user that just needs access to the PSTN then yes i agree that it doesnt matter. |
| 18:05.45 | [TK]D-Fender | ayeso: Its the same argument. suppose none of the rest of the world had any VoIP equipment and I invent the first SIP gateway. I guess the whole PSTN is VoIP now, by your account, isn't it? |
| 18:05.56 | ayeso | I just thought she was asking a non asterisk related question about the PSTN. |
| 18:05.58 | *** join/#asterisk sysreq (n=sysreq@unaffiliated/sysreq) |
| 18:06.16 | [TK]D-Fender | ayeso: Who uses little bits in between doesn't change the backend lowest common denominator. |
| 18:06.51 | Qwell | [TK]D-Fender: I'm going to create my OWN PSTN! |
| 18:07.04 | Qwell | and you aren't invited. |
| 18:07.42 | [TK]D-Fender | Qwell: I would have doubled your enrollment ;) |
| 18:07.46 | ayeso | [TK]D-Fender: Id say it does matter who uses little bits when "who" is major players like ATT, when you make a LD call these days there is a high probablilty that it will be VOIP on some leg of the call. |
| 18:08.07 | [TK]D-Fender | ayeso: When that segment is invisible to you I think it matters not a bit |
| 18:08.39 | spck | you guys every hear of Schrodinger's Cat? |
| 18:08.46 | [TK]D-Fender | ayeso: And ask yourself if its truely IP VS just raw multiplexed recompanded data. |
| 18:09.11 | [TK]D-Fender | ayeso: IP has overhead, packetization, jitter, etc. |
| 18:09.18 | ayeso | [TK]D-Fender: I agree when you are only using * and you have a telco provide you with access to the PSTN. Again the question was not related to asterisk but rather a general question about the PSTN, she wanted to know if it was VOIP and the answer is YES as some points it is VOIP. |
| 18:09.57 | [TK]D-Fender | ayeso: better answer might be "different little bits behind the scene MIGHT be, not that that matters to you" |
| 18:10.57 | ayeso | [TK]D-Fender: thats a fine answer with me. |
| 18:11.17 | ayeso | [TK]D-Fender: but to say that the PSTN is not VOIP is incorrect |
| 18:11.20 | [TK]D-Fender | ayeso: But really, even asking it as suck is just someone fishing for a "yes" |
| 18:11.42 | [TK]D-Fender | ayeso: such* |
| 18:11.45 | *** join/#asterisk JayTee52 (n=jforde@unaffiliated/jaytee) |
| 18:11.52 | *** join/#asterisk jplank (n=GBove@cpe-075-181-097-208.carolina.res.rr.com) |
| 18:11.59 | ayeso | [TK]D-Fender: I agree with that, |
| 18:12.46 | [TK]D-Fender | ayeso: It tends to lead to dumber and dumber questions when you hand it to them like that. |
| 18:12.47 | ayeso | Her question was really out of place in this channel really. |
| 18:13.02 | *** join/#asterisk gr0mit (n=tim@82.132.136.150) |
| 18:13.39 | ayeso | [TK]D-Fender: I know thats true, I prefer the help you give me: i ask a specific question, you shove me in the right direction to figure it out for myself. |
| 18:13.59 | VaGoNeTaS | i nree/wi |
| 18:17.20 | *** join/#asterisk lost_soul (i=shawn@cpe-67-241-68-104.twcny.res.rr.com) |
| 18:26.21 | *** join/#asterisk BBHoss_Laptop (i=18d6d2e7@gateway/web/ajax/mibbit.com/x-9094ace70a8238ae) |
| 18:27.34 | BBHoss_Laptop | Hi, I'm using Asterisk 1.4, and I'm trying to get parking working. When I transfer a call to 700, it doesn't read me back the extension. Any idea whats going on? There are no errors on the console, just never plays it back. |
| 18:31.07 | *** join/#asterisk telnettech (i=telnette@gw.percipia.com) |
| 18:32.28 | amaache | Hi to all; does cisco 7911 run on Asterisk |
| 18:33.38 | [TK]D-Fender | amaache: no, it has its own firmware |
| 18:34.10 | *** join/#asterisk manxpower (n=Administ@router.asteriasgi.com) |
| 18:35.02 | *** join/#asterisk qdk (n=qdk@0x573d8dd1.bynqu1.dynamic.dsl.tele.dk) |
| 18:35.42 | kc8pxy | amaache: ........ .... ?? |
| 18:36.02 | [TK]D-Fender | kc8pxy: He's been asking for over a week now, here and in the GUI support channels |
| 18:36.10 | BBHoss_Laptop | [TK]D-Fender: any idea why parking wouldnt be announcing? |
| 18:36.29 | kc8pxy | amaache: am i just noob, or does that sound like "does this honda engine run in my mustang?" |
| 18:36.39 | manxpower | BBHoss_Laptop: no sounds installed? |
| 18:36.43 | [TK]D-Fender | BBHoss_Laptop: Maybe you're doin it wrong? |
| 18:36.58 | BBHoss_Laptop | well i am including the parkedcalls context |
| 18:37.07 | BBHoss_Laptop | and there are no errors about sounds being missing |
| 18:37.10 | [TK]D-Fender | BBHoss_Laptop: And I'm not seeing anything |
| 18:37.23 | [TK]D-Fender | amaache: JFGI - http://www.google.ca/search?hl=en&q=cisco+7911+asterisk&btnG=Google+Search&meta=&aq=0&oq=Cisco+7911+asteris |
| 18:37.23 | BBHoss_Laptop | what do you want to see? |
| 18:37.27 | manxpower | there's a difference between parking not working and not hearing the parking lot audio files. Which issue are you having? |
| 18:37.54 | eppigy | amaache: you will need to install the SIP firmware |
| 18:38.02 | [TK]D-Fender | BBHoss_Laptop: CLI output for the failed attempt, and maybe a useful description of the precise process and equipment used |
| 18:38.04 | *** join/#asterisk polerin (n=erin@c-68-53-116-205.hsd1.tn.comcast.net) |
| 18:38.20 | *** join/#asterisk a1fa (n=a1fa@unaffiliated/a1fa) |
| 18:38.21 | BBHoss_Laptop | manxpower: the only issue is it doesn't say "7 0 1" |
| 18:38.25 | BBHoss_Laptop | etc |
| 18:38.27 | manxpower | in item 1, you screwed something up, in option 2 you either don't have the sounds installed OR (and my bet) you are doing a blind transfer (caller hear the parking lot) instead of doing an attended transfer (you hear the lot number) |
| 18:38.38 | amaache | sorry; and the ATA 186 does it run |
| 18:38.46 | BBHoss_Laptop | well i am hitting transfer on the polycom to 700 |
| 18:38.49 | BBHoss_Laptop | not blind |
| 18:38.50 | [TK]D-Fender | amaache: Yes, both can be used with * |
| 18:38.58 | a1fa | hey.. i have another asterisk box that needs to use my * box to send and recieve calls from.. on slave * i setup my master * as trunk |
| 18:38.58 | BBHoss_Laptop | but it just goes straight to MOH |
| 18:39.13 | manxpower | BBHoss_Laptop: pastebin the cli output of a failed parking |
| 18:39.15 | a1fa | how does the "slave" need to be configured on the "master" *? friend? peeer? |
| 18:39.24 | *** part/#asterisk mcargile (n=mikec@rrcs-24-173-156-170.se.biz.rr.com) |
| 18:39.40 | BBHoss_Laptop | manxpower: well it parks but it doesnt announce, so its not 100% fail, but i'll paste what i have |
| 18:41.01 | Qwell | 1.4 what? |
| 18:41.08 | BBHoss_Laptop | http://pastie.org/468966 |
| 18:41.33 | BBHoss_Laptop | Qwell: Asterisk 1.4.17~dfsg-2ubuntu1 built by buildd @ vernadsky on a i686 running Linux on 2008-04-12 07:08:45 UTC |
| 18:41.38 | Qwell | mmhmm |
| 18:41.47 | a1fa | :( |
| 18:41.50 | Qwell | BBHoss_Laptop: you know what I'm going to tell you, right? |
| 18:41.58 | BBHoss_Laptop | what |
| 18:42.01 | Qwell | upgrade. |
| 18:42.12 | BBHoss_Laptop | is there a reason to? |
| 18:42.16 | telnettech | In FreePBX, anybody know what i need to do to get two or three extensions to ring when 1 of then is called? example 7xxxis called and rings as well as 6xxx and 5xxx |
| 18:42.41 | Qwell | BBHoss_Laptop: other than that version being over a year old? |
| 18:42.48 | a1fa | how do i configure sip user to act as a trunk? |
| 18:43.11 | BBHoss_Laptop | Qwell: well thats not really a real reason, unless there is something specifically wrong with it |
| 18:43.19 | *** join/#asterisk c4rg (i=crg@lagoon.freebsd.lublin.pl) |
| 18:43.19 | a1fa | if user=friend, i can recieve phonecalls across the trunk, but i can not originate |
| 18:43.31 | KyleK | the version in ubuntu jaunty isn't much newer Asterisk 1.4.21.2~dfsg-1ubuntu3 built by buildd @ palmer on a i686 running Linux on 2008-09-30 01:16:31 UTC |
| 18:43.45 | eppigy | COMPILE FROM SOURCE |
| 18:43.48 | BBHoss_Laptop | [TK]D-Fender: http://pastie.org/468966 |
| 18:43.50 | eppigy | GET WITH THE PRORAM |
| 18:43.56 | eppigy | PROGRAM |
| 18:43.58 | BBHoss_Laptop | QUIT TYPING IN CAPS |
| 18:44.02 | a1fa | DAMN IT |
| 18:44.03 | eppigy | NEGATIVE |
| 18:44.07 | KyleK | STOP YELLING |
| 18:44.10 | c4rg | anyone having trouble with hylafax+asterisk 1.4+dahdi/wanpipe? |
| 18:44.11 | a1fa | .KICKBAN EPPIGY |
| 18:44.11 | eppigy | AUTOPILOT ENGAGED |
| 18:44.15 | a1fa | ;P |
| 18:44.21 | eppigy | but for real |
| 18:44.31 | a1fa | [TK]D-Fender : sup brother-b! |
| 18:44.44 | [TK]D-Fender | a1fa: Every SIP call is just like any other. frankly the only type you need for 995 of cases is "peer" |
| 18:44.50 | [TK]D-Fender | 99% |
| 18:45.04 | *** join/#asterisk djMax (n=chatzill@66.92.91.132) |
| 18:45.24 | Qwell | BBHoss_Laptop: |
| 18:45.26 | [TK]D-Fender | telnettech: Asking in the wrong channel... |
| 18:45.26 | Nugget | telnet is eeeeeeevil! |
| 18:45.29 | Qwell | Changes since asterisk Version 1.4.17/ - svn revision 95956 |
| 18:45.29 | Qwell | 1409 |
| 18:45.32 | manxpower | BBHoss_Laptop: when you do a transfer can you talk to the other person first? (not parking) |
| 18:45.33 | KyleK | eppigy: I'll run from source when I have time to look up how to package it up for the package manager |
| 18:45.34 | *** join/#asterisk bsumrall (n=bsumrall@ftnco.com) |
| 18:45.35 | *** join/#asterisk Ryushin (i=proxy@windwalker.openinnovations.com) |
| 18:45.35 | Qwell | No, nothing specific. |
| 18:45.38 | Qwell | just 1400 bug fixes |
| 18:45.42 | djMax | is there a generic way to call Dial() w/o voicemail forwarding? (without changing every extension) |
| 18:45.45 | manxpower | Qwell: was Call parking broken in 1.4.17? |
| 18:45.51 | eppigy | KyleK: no better time than the presenty |
| 18:45.58 | a1fa | [TK]D-Fender : got it in peer.. and that "peer" is configured as "trunk" on another * box |
| 18:46.00 | Qwell | 1400 bug fixes. |
| 18:46.01 | telnettech | TK: I asked in the Freepbx channel and it is deserted....i swear i heard crickets |
| 18:46.05 | [TK]D-Fender | djMax: what does "Dial" have to do with "VoiceMail'? |
| 18:46.06 | Qwell | There was clearly a lot broken in 1.4.17 |
| 18:46.13 | bsumrall | Is anyone here familiar with smsq? the sms feature in askerisk? |
| 18:46.14 | a1fa | [TK]D-Fender i can recieve phonecalls over that trunk.. but i cant send anycalls over it |
| 18:46.14 | manxpower | djMax: Dial does not EVER forward to voicemail. |
| 18:46.22 | Qwell | BBHoss_Laptop: Don't waste time. Upgrade. |
| 18:46.23 | [TK]D-Fender | telnettech: Yes, but you'll get more here. It is not supported here. |
| 18:46.28 | manxpower | bsumrall: Yes. It only works with european sms carriers. |
| 18:47.02 | djMax | I know it's not its job, but the app is that I use the manager API to originate a call from a Local/xxx extension, and it will connect the call even if I reject it (to VM) |
| 18:47.25 | bsumrall | manxpower: I am using trixbox and try to send a test sms message to a SIP channel and the message is created in the spooler but never get digested. |
| 18:47.28 | djMax | trying to figure a way out w/o creating shadow extensions |
| 18:47.30 | bsumrall | just a file sitting in the spooler? |
| 18:47.31 | manxpower | bsumrall: that will never work. |
| 18:47.36 | telnettech | im trying |
| 18:47.36 | [TK]D-Fender | djMax: what does "reject to VM mean? This has nothing to do with "dial" |
| 18:47.43 | bsumrall | really? |
| 18:47.45 | djMax | Agreed, sorry, poorly stated. |
| 18:48.00 | manxpower | bsumrall: app_sms needs to be calling an sms provider. It will not work with VoIP, only ZAP |
| 18:48.03 | djMax | I either don't answer the call to my local extension, or hit reject on the sip phone |
| 18:48.17 | manxpower | djMax: then you need to look at the lines AFTER dial |
| 18:48.21 | [TK]D-Fender | djMax: And this is your dialplan go look at what you're doing. YOU are the one calling voicemail. |
| 18:48.23 | bsumrall | how do I send the identical sms message to 10 people or even just a test message? |
| 18:48.36 | manxpower | bsumrall: what carrier are you using? |
| 18:48.36 | BBHoss_Laptop | Qwell: should i stick with 1.4 or go with 1.6? |
| 18:48.51 | a1fa | [TK]D-Fender : so i should be able to send calls across the trunk if user is configured as peer on the other end ? |
| 18:48.52 | manxpower | What SMS carrier that is. Also what country are you in? |
| 18:49.03 | bsumrall | teliax! |
| 18:49.03 | [TK]D-Fender | a1fa: yes |
| 18:49.03 | djMax | true, because for most incoming calls I want that. I'm just saying is there some way to override that on the "originating" side? |
| 18:49.11 | a1fa | is there anything else "special" that needs to be set to allow that communication? |
| 18:49.17 | bsumrall | manxpower: what would you suggest? |
| 18:49.29 | *** join/#asterisk ScarEye (n=scareye@12.27.87.66) |
| 18:49.41 | djMax | for example, a kludge would be (maybe) to create a context that set a variable and then sent the call to the "regular" context, which would check for that variable before doing more. |
| 18:49.56 | djMax | but not sure what Originate will even do there, and whether there are smarter/better ways |
| 18:50.03 | manxpower | bsumrall: Look. Dude. app_SMS is designed to work with the SMS protocols supported by the european TDM and cell carriers. It won't work with anything else. Either you are calling an European carrier's SMS service center number or it will not work. |
| 18:50.26 | manxpower | bsumrall: now if you just want to send text messages to phones then sign up with a text message carrier. There are many of them out there. |
| 18:50.40 | [TK]D-Fender | djMax: this is all completely boring dialpla, its not even a question. You dump it to the exten you point it to. IF you want to do something different, go make something different and point it to that |
| 18:50.48 | bsumrall | manxpower: i understand. You you know of another solution that may work to solve the problem? |
| 18:51.10 | djMax | right, you're basically saying create shadow extensions. Unpleasant, but ok. |
| 18:51.10 | manxpower | bsumrall: I just gave you a solution. Sign up for a test messaging provider. |
| 18:51.19 | bsumrall | manxpower: thank you! |
| 18:51.26 | manxpower | text provider that is. |
| 18:51.33 | manxpower | voip-info should have a list of them. |
| 18:51.41 | manxpower | bsumrall: but this has nothing to do with Asterisk |
| 18:51.41 | [TK]D-Fender | djMax: everything depends on what you've got now and what is efficient to change VS add |
| 18:51.55 | *** join/#asterisk empiric (n=empiric@116.71.37.89) |
| 18:51.58 | djMax | yeah, it's a freepbx vanilla install now mostly |
| 18:52.42 | empiric | guys i am using dlink as FXO gaeway |
| 18:52.57 | bsumrall | yes, they do. I really though sms over sip worked, there is even a mini howto, but it seems to be missing the final step. I would assume that it is because it is used to relay to the sms provider. |
| 18:53.13 | kc8pxy | empiric: a dlink what? |
| 18:53.18 | [TK]D-Fender | djMax: there are barely words to describe how much it is NOT supported in here. |
| 18:53.42 | kc8pxy | didn't know dlink made FX* cards. |
| 18:53.45 | djMax | grumpy day. very well. |
| 18:54.00 | *** part/#asterisk BBHoss_Laptop (i=18d6d2e7@gateway/web/ajax/mibbit.com/x-9094ace70a8238ae) |
| 18:54.54 | manxpower | bsumrall: link to the howto? |
| 18:56.13 | bsumrall | http://www.voip-info.org/wiki/view/Asterisk+cmd+SendText |
| 18:56.29 | manxpower | bsumrall: there is your mistake. SendText is NOT SMS. |
| 18:56.47 | bsumrall | ? |
| 18:56.56 | manxpower | bsumrall: SMS is a specific protocol. |
| 18:57.27 | manxpower | SendText does not implement the SMS protocol. |
| 18:57.29 | bsumrall | Ok, I understad. what is SendText? |
| 18:57.47 | manxpower | As for SendText I'm not aware of any SIP devices that supports it. |
| 18:57.50 | bsumrall | so it will never hit a cell phone? |
| 18:58.04 | bsumrall | understood! |
| 18:58.07 | manxpower | bsumrall: no, it will never hit a cell phone. |
| 18:58.19 | bsumrall | you have cleared up many questions my friend! |
| 18:58.30 | bsumrall | thank you! |
| 18:58.59 | bsumrall | but sms will have no issues with a fx/o card I gather. |
| 19:00.11 | manxpower | bsumrall: SMS is a specific protocol. You will have app_SMS call the carrier's special SMS phone number. Then app_sms will send an FSK (modem) burst to the carrier. The carrier will then deliver the message to the handset. You can also use app_sms to send messages to landline SMS devices. |
| 19:00.34 | *** join/#asterisk lost_soul (i=shawn@cpe-67-241-68-104.twcny.res.rr.com) |
| 19:00.54 | manxpower | app SMS does nothing but send the correctly formatted FSK (modem) data burst to the other end of the line. |
| 19:01.03 | bsumrall | what is FSK? |
| 19:01.19 | manxpower | bsumrall: It's the protocol 2400 baud modems use. |
| 19:01.29 | bsumrall | Ahhh! |
| 19:01.31 | [TK]D-Fender | ~fsk |
| 19:01.32 | bsumrall | thank you! |
| 19:01.36 | manxpower | frequency shift key if I recall correctly. |
| 19:01.40 | [TK]D-Fender | manxpower: Yup |
| 19:03.52 | SuPrSluG | everything old is new again |
| 19:05.59 | *** join/#asterisk BlargMaN00 (n=blarg@12.234.16.130) |
| 19:07.17 | Qwell | manxpower: frequency-shift keying |
| 19:07.24 | Qwell | (close enough) |
| 19:09.27 | cvnet | I've been using SIP for long time, now I got a new provider who requires me to send H323, I have never used H323 b4, do I just send exten => _X.,n,Dial(H323/IP/Number) ? Or do i need to install something .... ? |
| 19:09.39 | *** join/#asterisk lasko (n=lasko@70.102.15.210) |
| 19:09.54 | *** join/#asterisk crevetor (n=crevetor@bureau.ubity.com) |
| 19:10.07 | SuPrSluG | no, get a new provider |
| 19:10.15 | [TK]D-Fender | cvnet: go look at what kind of H.3232 channels you have installed. There is H323, OH323, and I think one other |
| 19:10.41 | cvnet | [TK]D-Fender: were do I find that info ? |
| 19:10.44 | crevetor | Question : if my peers are natted should asterisk store the public IP address in the fullcontact or should it store the private IP address ? |
| 19:10.54 | [TK]D-Fender | cvnet: go look in your tarball |
| 19:10.58 | *** part/#asterisk lasko (n=lasko@70.102.15.210) |
| 19:11.09 | [TK]D-Fender | cvnet: the configs are rather clear and googleable |
| 19:11.29 | manxpower | crevetor: sip show peers should show the public ip. If it's not then you forgot nat=yes |
| 19:12.54 | crevetor | manxpower: it does show the public IP. Ihave a more subtle problem |
| 19:13.23 | *** join/#asterisk exothermc (n=miles@74.85.89.146) |
| 19:13.32 | [TK]D-Fender | crevetor: Reaally... Show us the SIP debug of the call |
| 19:13.50 | crevetor | [TK]D-Fender: It's not call related |
| 19:13.53 | crevetor | it's qualify related |
| 19:14.12 | [TK]D-Fender | crevetor: the Register will be every bit as evident. |
| 19:14.38 | crevetor | [TK]D-Fender: ASterisk for some reason that I haven't been able to figure out (yet) send options packet to both the public ip address and the private ip address |
| 19:14.49 | crevetor | [TK]D-Fender: really ? |
| 19:14.56 | [TK]D-Fender | crevetor: * dousen't talk to 2 places to go to the same device |
| 19:15.03 | [TK]D-Fender | crevetor: Yes, really/ |
| 19:15.22 | [TK]D-Fender | crevetor: If your peer is wrong then it will not interpret which IP to use properly |
| 19:15.33 | crevetor | [TK]D-Fender: please replace doesn't with shouldn't... |
| 19:16.03 | [TK]D-Fender | crevetor: WON'T. * doesn't talk to 2 completely different places for the same transaction. |
| 19:16.17 | [TK]D-Fender | crevetor: Please jsut show us your configs and debug |
| 19:16.46 | [TK]D-Fender | ~pb |
| 19:16.47 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/ |
| 19:16.48 | [TK]D-Fender | ^^^^^^^ |
| 19:16.56 | *** join/#asterisk bmoraca (n=chatzill@66.242.174.254) |
| 19:17.10 | crevetor | [TK]D-Fender: I'll try and get the relevant information |
| 19:18.36 | manxpower | [TK]D-Fender: I suspect host != dynamic |
| 19:19.13 | [TK]D-Fender | manxpower: that'd be something entirely else. Still wouldn't cause 8 to send to 2 hosts |
| 19:19.16 | [TK]D-Fender | * |
| 19:19.42 | manxpower | [TK]D-Fender: it would if there are two entries |
| 19:19.59 | [TK]D-Fender | manxpower: Can't have 2 hosts for a given peer |
| 19:20.07 | [TK]D-Fender | manxpower: Not a repeatable tag |
| 19:20.15 | manxpower | [TK]D-Fender: you know that asterisk can be pretty liberal about matching peers. |
| 19:20.39 | manxpower | My idea was a host != dynamic and the incoming registeration matching some other peer |
| 19:20.55 | manxpower | obviously this is all speculation without seeing the config files. |
| 19:21.15 | [TK]D-Fender | manxpower: Yeah, you know my default trust level :) |
| 19:22.46 | drmessano | deny=0.0.0.0? |
| 19:23.07 | a1fa | well.. i give up |
| 19:23.14 | drmessano | allow=127.0.0.1 ? |
| 19:23.30 | drmessano | ^^^^ Best ACL ever |
| 19:23.31 | a1fa | i can recive calls but i cant make phonecalls.. * log file does not show anything.. so I am guessing "PBXES.ORG" is buggy |
| 19:23.38 | a1fa | drmessano : lies ;P |
| 19:23.49 | drmessano | flybynightprovider.com is buggy? |
| 19:24.04 | drmessano | cheapasssipprovider.net is unreliable? |
| 19:24.12 | a1fa | drmessano : nah its free |
| 19:24.12 | cvnet | [TK]D-Fender: in etc/asterisk i got h323.conf, does that mean I have it installed? |
| 19:24.13 | drmessano | experiences shock AND awe |
| 19:24.23 | a1fa | drmessano : i am using it to re-route calls to my Android G1 phone |
| 19:24.32 | drmessano | No |
| 19:24.35 | [TK]D-Fender | cvnet: Do you see a CHANNEL DRIVER module compiled? |
| 19:24.37 | drmessano | You were GOING to |
| 19:24.39 | a1fa | drmessano : i am going to be buying data only plan and run voip on top of my phone |
| 19:24.43 | a1fa | drmessano : $15/month |
| 19:24.49 | seanbright | they even have an invalid SSL certificate |
| 19:24.51 | seanbright | i love these guys |
| 19:24.52 | cvnet | justdave: where do i look for that? |
| 19:25.01 | cvnet | [TK]D-Fender: where do i look for that? |
| 19:25.13 | drmessano | seanbright: youcantrustuswithyourcreditcard.com isn't reputable? |
| 19:25.35 | [TK]D-Fender | cvnet: in your modules folder... where else? |
| 19:25.37 | *** join/#asterisk shinao1 (n=shinao1@78.138.29.146) |
| 19:25.50 | drmessano | I get my SSL certs from worksformeificlickallow.com |
| 19:25.55 | crevetor | http://pastebin.com/m626002f8 |
| 19:28.00 | bmoraca | i just can't bring myself to trust anything that advertises a service for free... |
| 19:28.22 | [TK]D-Fender | crevetor: I see HALF of the other qualify. |
| 19:29.22 | *** join/#asterisk guax (n=guaxinim@unaffiliated/guaxinim) |
| 19:29.22 | crevetor | [TK]D-Fender: the second one gets replied |
| 19:29.23 | [TK]D-Fender | crevetor: crevetor And Nat : Always should be YES |
| 19:29.35 | a1fa | _+x. should match *, right? |
| 19:29.41 | [TK]D-Fender | crevetor: CanReinvite : Yes <- BAD |
| 19:29.44 | crevetor | [TK]D-Fender: do you want the reply ? |
| 19:29.45 | *** join/#asterisk shinao1 (n=shinao1@78.138.29.146) |
| 19:29.57 | a1fa | [TK]D-Fender : that's default behaviour for Avaya VOIP :P |
| 19:30.05 | crevetor | [TK]D-Fender: ok let me check something |
| 19:30.11 | [TK]D-Fender | crevetor: Fix those entries, reboot the phone, pastebin all debug |
| 19:31.07 | *** join/#asterisk shinao1 (n=shinao1@78.138.29.146) |
| 19:31.10 | *** join/#asterisk lanning (n=lanning@nat/yahoo/x-12329e5ff980c213) |
| 19:31.16 | *** join/#asterisk kb3ien (n=kb3ien@216.152.227.62) |
| 19:31.34 | *** join/#asterisk lost_soul (i=shawn@cpe-67-241-68-104.twcny.res.rr.com) |
| 19:31.47 | crevetor | [TK]D-Fender: in my DB (this is a realtime peer) I have nat=yes |
| 19:31.54 | crevetor | [TK]D-Fender: why would it say always |
| 19:32.49 | *** join/#asterisk shinao1 (n=shinao1@78.138.29.146) |
| 19:32.53 | eppigy | hello |
| 19:35.44 | cvnet | i know its a dumb questions, but whats the location of Asterisk's Module folder ? |
| 19:36.08 | [TK]D-Fender | cvnet: typically /usr/lib/asterisk/modules |
| 19:36.20 | cvnet | thanks a bunch |
| 19:36.42 | *** join/#asterisk fish-bulb (n=cstewart@nat/digium/x-aaaab49b19bd77b2) |
| 19:38.20 | crevetor | [TK]D-Fender: http://pastebin.com/d787ef6a1 other example. In the db the peer has nat=yes and canreinvite=no |
| 19:38.51 | kb3ien | i have issues with calls where if there are long pauses sometime even a second or two is long enough, there is a problem restarting the audio again leading edges get dropped: "Foo Bar (pause 1 second) baz quux" comes out as "foo, bar (2 second delay) quux". This has all the earmarks of vad, but i have no VAD enabled anywhere, that i can tell (active leg is a SIP trunk to a cisco) any suggestions? |
| 19:39.12 | cvnet | chan_ooh323.so is there but not regular h323 |
| 19:39.52 | [TK]D-Fender | crevetor: Funny i don't see the Register attempt from the rebooted phone in there... |
| 19:41.05 | crevetor | [TK]D-Fender: Ok, let me see if I can find a phone to reboot (this is a production system() |
| 19:42.50 | kb3ien | calls are ulaw. asterisk is SVN-branch-1.4-r165796 it was forked late last year. |
| 19:43.14 | *** join/#asterisk EugenA (n=eugen@212.203.37.194) |
| 19:45.29 | *** join/#asterisk shinao1 (n=shinao1@78.138.29.146) |
| 19:46.26 | kb3ien | does asterisk even support vad? i see options for vad in codecs.conf under [speex] |
| 19:48.05 | *** join/#asterisk Ast001 (n=uros@cable-89-216-155-28.dynamic.sbb.rs) |
| 19:49.15 | Ast001 | Hi I have bought g729 codec from Digium and installed it according to README but it does not work. |
| 19:49.31 | Ast001 | Here is what I get during the call :http://pastebin.com/m12b6ea7e |
| 19:50.18 | *** join/#asterisk ingenius (n=alektro@host90.190-230-73.telecom.net.ar) |
| 19:50.22 | [TK]D-Fender | Ast001: [May 5 19:24:23] WARNING[21479] codec_g729a.c: out of G.729 decoder licenses |
| 19:50.26 | [TK]D-Fender | Ast001: oUT OF LICENCES |
| 19:50.38 | Ast001 | my licence is not good ? |
| 19:50.52 | Ast001 | I typed licence Digium emailed me |
| 19:50.53 | Katty | hai |
| 19:51.26 | Katty | eppigy: i went and got stuff to make that carne asada. |
| 19:52.01 | [TK]D-Fender | Ast001: You don't have ENOUGH for what you're doing on your system at that moment |
| 19:52.03 | Qwell | Ast001: it is for a specific number of simultaneous channels |
| 19:52.45 | Ast001 | I bought one licence and one operator was logged in and one call came |
| 19:53.19 | [TK]D-Fender | Ast001: you did not sure what calls were actually in progress before that call came in, or the debug of the call itself |
| 19:53.28 | Qwell | Ast001: did you register it? |
| 19:53.35 | Qwell | You should probably contact Digium support. |
| 19:53.43 | Ast001 | yes I registered it on Digium web site |
| 19:53.43 | Corydon76-dig | Ast001: were you recording the call? |
| 19:53.47 | *** join/#asterisk SparFux (n=raoul@e182024121.adsl.alicedsl.de) |
| 19:53.51 | Ast001 | yes I do |
| 19:54.03 | Corydon76-dig | Ast001: the recording takes a license for itself |
| 19:54.14 | SparFux | I have a compiling issue of the zaptel drivers with bristuff on debian unstable: http://paste.nerv.fi/11064892.txt |
| 19:54.32 | SparFux | What I have done so far is m-a a-i. Hello, btw! :-D |
| 19:54.33 | *** join/#asterisk fakhir (n=fakhir@unaffiliated/fakhir) |
| 19:54.34 | Ast001 | so you think I need to cancel recording ? |
| 19:54.48 | Ast001 | disable recording in conf files ? |
| 19:55.06 | Qwell | SparFux: http://www.debian.org/Bugs/ |
| 19:55.13 | Corydon76-dig | or buy a second license |
| 19:55.30 | Ast001 | I see |
| 19:55.44 | manxpower | Ast001: with recording on you have TWO places where the data is decoded, therefor you need two licenses |
| 19:55.54 | SparFux | Qwell: I haven't found the issue yet. And I tried to build svn from debian but I am too noob to do so. I cannot find out how to compile trunk. |
| 19:56.07 | Ast001 | ok thanks I'll try to disable recording in agents.conf |
| 19:56.12 | Qwell | SparFux: Debian manages that stuff, not us. |
| 19:56.22 | SparFux | ah, ok. I get it. |
| 19:56.42 | SparFux | I'll just report. Thx for the pointer. |
| 19:57.44 | *** part/#asterisk SparFux (n=raoul@e182024121.adsl.alicedsl.de) |
| 20:00.10 | bmoraca | i wonder if it's possible for Polycom to have made their config files any more obtuse |
| 20:00.41 | JayTee52 | no without rewritting them in Sanskrit or Aramaic |
| 20:01.01 | manxpower | they could be BINARY config files like grandstream uses. |
| 20:01.21 | bmoraca | blech |
| 20:01.32 | JayTee52 | so essentially yeah, life could be worse |
| 20:01.36 | manxpower | and Linksys too I think. |
| 20:02.15 | manxpower | essentially, I only have to add like 4 config lines for each phone once you get the master config files working correctly. |
| 20:02.48 | bmoraca | manxpower: yeah, that's how polycom's works too...but still... |
| 20:03.00 | *** join/#asterisk yo-mama (n=bsumrall@ftnco.com) |
| 20:03.42 | manxpower | I was referring to Polycom. I'd not use any other brand of phone. |
| 20:04.26 | bmoraca | ahhh |
| 20:04.33 | bmoraca | i'm preferable to them, too |
| 20:05.38 | *** join/#asterisk jplank (n=GBove@cpe-075-181-097-208.carolina.res.rr.com) |
| 20:06.28 | *** join/#asterisk BlargMaN00 (n=blarg@12.234.16.130) |
| 20:07.04 | *** join/#asterisk juanIMP (n=Juancho@200.71.41.22) |
| 20:08.51 | *** join/#asterisk Linuturk (n=linuturk@fluxbuntu/developer/Linuturk) |
| 20:16.35 | cvnet | if you have chan_ooh323 installed, does it also mean u got h323 installed? |
| 20:16.37 | crevetor | [TK]D-Fender: http://pastebin.com/d48c6d397 here you go |
| 20:17.47 | [TK]D-Fender | crevetor: So far its all one IP 9WAN) |
| 20:18.29 | *** join/#asterisk xcompile (n=xcompile@91-64-171-128-dynip.superkabel.de) |
| 20:22.17 | bmoraca | it'd be really nice if Cisco 7940 phones would respond to the Alert-Info header for auto answer purposes. |
| 20:22.49 | crevetor | [TK]D-Fender: then I get things like http://pastebin.com/dce2d252 |
| 20:23.18 | crevetor | [TK]D-Fender: but it also sends the options packet to the public IP |
| 20:24.07 | *** join/#asterisk cesar_CR (n=cesar@201.195.239.11) |
| 20:24.08 | yo-mama | crevetor: what is your issue? |
| 20:25.06 | *** join/#asterisk Ose (n=chatzill@wikia/Ose) |
| 20:25.17 | *** part/#asterisk Ose (n=chatzill@wikia/Ose) |
| 20:25.49 | [TK]D-Fender | crevetor: No idea... |
| 20:25.53 | [TK]D-Fender | BBIA |
| 20:28.00 | crevetor | yo-mama: Asterisk sends Options packet to the private Ip of a natted device (and also to it's public IP) |
| 20:35.04 | yo-mama | crevetor: What kind of router? |
| 20:35.07 | drmessano | ~sipnat |
| 20:35.08 | infobot | [~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 |
| 20:35.14 | drmessano | Read that. crevetor |
| 20:35.22 | drmessano | It has the settings youre looking for |
| 20:35.30 | drmessano | Particuarly externhost and externip |
| 20:36.08 | *** join/#asterisk SparFux (n=raoul@e182024121.adsl.alicedsl.de) |
| 20:36.09 | yo-mama | crevetor: if your packet is being routed to a private network, of course it is going to reflect the private ip. |
| 20:36.12 | SparFux | Re! |
| 20:36.39 | yo-mama | crevetor: what is the topology? |
| 20:37.36 | ayeso | Anyone know a way to NOT have the asterisk console report that there is a remote unix connection when someone issues an asterisk -rx command? |
| 20:38.01 | Katty | eppigy: let there be you! |
| 20:38.08 | Katty | eppigy: let there be me! |
| 20:39.14 | crevetor | yo-mama: Server on a public IP, peer on a private ip behind a nat |
| 20:39.25 | SparFux | Now I have compiled the debian zaptel stuff despite all difficulties and I want to use it with asterisk :-) |
| 20:39.47 | manxpower | SparFux: Great! Go to #debian for help. |
| 20:39.50 | *** join/#asterisk telnettech (i=telnette@gw.percipia.com) |
| 20:40.06 | *** join/#asterisk bkruse (n=bkruse@76.73.154.120) |
| 20:40.06 | *** mode/#asterisk [+o bkruse] by ChanServ |
| 20:40.18 | Qwell | throws a potato at bkruse |
| 20:40.33 | yo-mama | crevetor: router hosting the client is where? |
| 20:40.35 | Katty | intercepts potato |
| 20:40.40 | Katty | makes casserole with it |
| 20:40.52 | Qwell | Katty: it was already mashed :( |
| 20:40.53 | drmessano | intercepts oven and eats said casserole |
| 20:40.59 | Katty | :< |
| 20:41.01 | bkruse | Qwell! |
| 20:41.04 | crevetor | yo-mama: physically ? |
| 20:41.14 | drmessano | burppp |
| 20:41.17 | *** join/#asterisk voxter (n=voxter@190.241.15.217) |
| 20:41.18 | bkruse | Qwell: How's it goin? |
| 20:41.24 | Qwell | bkruse: not bad |
| 20:42.19 | yo-mama | crevetor: do you cross a routable network? I.E the internet between client host router and server? |
| 20:42.34 | *** join/#asterisk Meaw (n=dino@213.244.81.144) |
| 20:43.24 | yo-mama | crevetor: the private ip is only in the options portion |
| 20:43.52 | yo-mama | crevetor: never mind, your issue is with sip.conf i believe |
| 20:45.00 | crevetor | yo-mama: Would you know which option I'd have to change ? |
| 20:45.12 | crevetor | yo-mama: do you want a snippet of my sip.conf ? |
| 20:45.57 | KyleK | have you set localnet? |
| 20:46.06 | eppigy | Katty: it is so |
| 20:46.06 | crevetor | KyleK: yes |
| 20:46.49 | *** part/#asterisk SparFux (n=raoul@e182024121.adsl.alicedsl.de) |
| 20:47.22 | yo-mama | crevetor: do a grep "192.168.1.104" /etc/asterisk/ |
| 20:47.36 | drmessano | I hope not |
| 20:47.37 | crevetor | why 104 specifically ? |
| 20:47.42 | drmessano | 104 would be wrong |
| 20:48.41 | drmessano | localnet is a network/mask, not a specific IP |
| 20:49.20 | drmessano | So something like localnet=192.168.1.0/255.255.255.0 |
| 20:49.36 | crevetor | drmessano: I have that as a localnet |
| 20:49.58 | crevetor | still it used to do the semae when i didn't have 192.168.0.0/16 in localnets |
| 20:50.02 | drmessano | Do you have an externhost/externip defined? |
| 20:50.26 | crevetor | drmessano: externip yes |
| 20:50.33 | manxpower | crevetor: Are you SURE you are looking at the PACKET address or are you looking at the address INSIDE the packet? |
| 20:50.50 | manxpower | the address inside does not mean much. |
| 20:51.08 | *** join/#asterisk [TK]D-Fender (n=joe@64.235.218.194) |
| 20:51.28 | crevetor | manxpower: absolutely certain : I have packet dumps from tshark to attest this |
| 20:52.12 | *** join/#asterisk johnakabean (n=some@pool-72-82-111-184.nrflva.east.verizon.net) |
| 20:52.16 | yo-mama | manxpower: inside the packet |
| 20:52.42 | yo-mama | manxpower: his output is from the console, not a sniffed packet |
| 20:52.47 | manxpower | yo-mama: inside the packet doesn't mean anything except to the endpoints. |
| 20:53.17 | manxpower | in fact, if the public IP was inside the packet I would expect the device to reject it, since the device is not on that IP. |
| 20:53.17 | yo-mama | crevetor: thats a wireshark dump you posted? |
| 20:53.21 | johnakabean | hey room, for streaming music from shoutcast, what needs to be installed |
| 20:53.22 | crevetor | yo-mama: the output I pasted is from asterisk's console but I also made packet dumps which attest that the packets are sent to private IPs |
| 20:53.55 | crevetor | manxpower: that's what I thought as well |
| 20:53.57 | manxpower | johnakabean: nothing that I know of will restart the shoutcase stream when it breaks. |
| 20:54.30 | yo-mama | manxpower: right, but there is NO way a private ip address is going to cross the internet in the tcp header |
| 20:55.18 | yo-mama | crevetor: you got a routing issue! |
| 20:56.01 | Meaw | hi guys, i have a serious problem, first im not that good in asterisk..but we have E1 and toll free number.. im trying to forward caller ID to our switch..but right now when i call i get "user busy" |
| 20:56.35 | yo-mama | manxpower: my bad, ip portion |
| 20:57.16 | crevetor | yo-mama: Can you tell me what leeds you to this conclusion ? |
| 20:57.16 | manxpower | Meaw: no, what do you REALLY get? |
| 20:57.43 | Meaw | when i call the toll free from my mobile.. i get "user busy" |
| 20:58.01 | *** join/#asterisk cvnet (n=dahitler@24.156.136.205) |
| 20:58.02 | manxpower | Meaw: that means nothing. What do you get on your switch or on the Asterisk console? |
| 20:58.24 | Meaw | i'll paste what i get in the asterisk console hold on |
| 20:58.37 | manxpower | Meaw: use pastebin.va |
| 20:58.39 | manxpower | .ca that is |
| 20:58.40 | manxpower | ~pb |
| 20:58.41 | infobot | [~pb] A "pastebin" is a web-based service where you can paste anything over 3 lines without flooding the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://www.rafb.net/paste , or , http://bin.cakephp.org/ |
| 20:58.41 | Meaw | ok |
| 20:58.51 | Meaw | I know, I wasn't going to paste here |
| 20:58.59 | yo-mama | crevetor: ip rfc |
| 20:59.24 | Meaw | manxpower, here http://pastebin.com/m4c6ab03a |
| 20:59.30 | yo-mama | crevetor: post a packet dump from the server! |
| 20:59.54 | manxpower | [May  5 23:56:30] WARNING[11603]: chan_sip.c:4339 sip_call: No audio format found to offer. Cancelling call to 2832800 |
| 21:00.23 | manxpower | looks to me like you don't have any allowed codecs in sip.conf for the section [voipswitch] |
| 21:01.05 | Meaw | allow=g729 |
| 21:01.10 | Meaw | I have this codec allowed |
| 21:01.41 | manxpower | Meaw: do you have g729 licenses? |
| 21:01.55 | Meaw | nope.. cracked |
| 21:02.03 | manxpower | Meaw: I can't help you then. |
| 21:02.21 | manxpower | Best of luck. |
| 21:02.21 | Meaw | why not? |
| 21:02.38 | Meaw | heh. |
| 21:02.39 | manxpower | Meaw: because I have no interest trying to help someone with software I have never used and will never use. |
| 21:02.54 | manxpower | the fact that it would be illegal is a factor as well. |
| 21:03.21 | [TK]D-Fender | doesn't see any configs. Nor SIP debug of the failed attempt. Nor proof that the codec is installed |
| 21:03.33 | Meaw | weird, to get licensed i have to pay 15$ for every channel, which is 450$ :/ |
| 21:04.09 | manxpower | Meaw: that's weird. I only have to pay $10 and honestly if you can afford a server and a PSTN card you can afford a one time charge for g729 license. |
| 21:04.37 | manxpower | Also why are you even using G729? Do you have limited bandwidth between your Asterisk box and your "voipswitch" |
| 21:04.47 | [TK]D-Fender | manxpower: You mean the 3 X101P clone's he's running? ;) |
| 21:05.03 | Meaw | I have been waiting for the E1 card to arrive to The place where i live .. for like a year, and I paid more than the usual price to get it... so seriously too much things to pay for |
| 21:05.17 | Meaw | no, not limited bandwith. |
| 21:05.24 | manxpower | Meaw: you don't have to use G729 |
| 21:05.24 | Meaw | both in the same network |
| 21:05.30 | manxpower | then use alaw |
| 21:05.51 | manxpower | then you won't even have the quality loss associated with converting to/from the various codecs. |
| 21:07.06 | *** join/#asterisk |Cybex| (n=John@80.100.126.176) |
| 21:07.39 | Meaw | What is the codec name? ulaw or alaw? |
| 21:08.00 | KyleK | alaw is for e1's ulaw is for t1's |
| 21:08.12 | Meaw | k thanks |
| 21:09.36 | *** join/#asterisk seb- (n=seb@li30-51.members.linode.com) |
| 21:09.37 | *** part/#asterisk manxpower (n=Administ@router.asteriasgi.com) |
| 21:10.40 | *** join/#asterisk ScarEye (n=scareye@12.27.87.66) |
| 21:16.24 | *** join/#asterisk ghenry (n=ghenry@pdpc/supporter/monthlybyte/ghenry) |
| 21:16.42 | ghenry | any opinions of ilbc vs g729? |
| 21:17.05 | ghenry | gong to test both and have a listen |
| 21:18.10 | leifmadsen | ~~~~~~~~~~~~~~~~~~~~~ |
| 21:18.10 | infobot | extra, extra, read all about it, ~~~~~~~~~~~~~~~~~~~~ is your mom |
| 21:19.52 | *** join/#asterisk smth (n=mike__@199.84.137.3) |
| 21:20.57 | Meaw | alright, now im forwarding calls to another asterisk server.. this time im getting "noise" only |
| 21:23.17 | *** join/#asterisk Kisu (n=k@daniel1117.broker.freenet6.net) |
| 21:25.20 | *** join/#asterisk ingenius (n=alektro@OL77-237.fibertel.com.ar) |
| 21:29.36 | *** join/#asterisk BadHAL (n=nn@66.194.174.11) |
| 21:35.30 | *** join/#asterisk anthm (n=anthm@freeswitch/developer/anthm) |
| 21:44.56 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 21:45.28 | Meaw | so guys any thoughts about my issue?.. |
| 21:46.00 | Meaw | when i call the toll free number .. it answers and i got noise only now |
| 21:46.59 | Meaw | http://pastebin.com/m13b86713 |
| 21:49.26 | *** join/#asterisk marc7 (n=marc@bas2-montrealak-1167869307.dsl.bell.ca) |
| 21:53.49 | *** join/#asterisk SparFux (n=raoul@e182024121.adsl.alicedsl.de) |
| 21:53.53 | SparFux | Has anybody got an idea what could cause my sound of my sip phone to be too fast in asterisk right after I started using zaphfc driver? |
| 22:01.33 | Qwell | SparFux: I'll give you 1 guess... |
| 22:01.49 | Qwell | I'll be generous. You can have 1 guess per thing you changed. |
| 22:01.52 | SparFux | yes? |
| 22:02.27 | SparFux | I just switched one line from lcr channel driver to zap channel driver, but the speedup is in sip phone! |
| 22:02.58 | Qwell | and what happens when you change it back? |
| 22:05.47 | cvnet | i installed h323 on my existing system now when i try to run asterisk i get asterisk: error while loading shared libraries: libh323_linux_x86_r.so.1.18.0: cannot open shared object file: No such file or directory |
| 22:05.55 | SparFux | well... |
| 22:06.30 | *** join/#asterisk Chainsaw (n=chainsaw@gentoo/developer/atheme.member.chainsaw) |
| 22:06.34 | SparFux | wow! no sound at all! |
| 22:06.47 | Qwell | So what else did you change? |
| 22:07.57 | cvnet | never mind problem fixed |
| 22:08.50 | SparFux | ah, zaphfc hasn't been unloaded. |
| 22:09.00 | SparFux | sorry. modprobe failed to unload. |
| 22:09.13 | SparFux | I have to use rmmod. |
| 22:09.20 | Meaw | Qwell, about my problem any idea? :) |
| 22:10.12 | Qwell | Meaw: get rid of the gsm prompts (use ulaw prompts, IMO), or recompile Asterisk with a different version of gcc |
| 22:10.14 | Qwell | ~gsmbug |
| 22:10.15 | infobot | [~gsmbug] there is a bug compiling Asterisk with GCC 4.2 where optimization errors cause GSM transcoding to distort heavily. Compile with GCC 4.1 or lower, or follow this patch : http://bugs.digium.com/view.php?id=11243 Also please read : http://forums.digium.com/viewtopic.php?p=116731&sid=3c65e49ec840380ed20f1c8426382b39 |
| 22:10.15 | SparFux | Ok, when I change back to linux-call-router channel driver sound is normal again. Why would a sipp phone get too fast sound when channel isdn driver is zaphfc and not lcr? |
| 22:10.17 | Qwell | You may be hitting that |
| 22:10.25 | Qwell | SparFux: Don't know. Ask them. |
| 22:10.35 | SparFux | ok. |
| 22:10.49 | SparFux | thx anyway. |
| 22:14.54 | *** join/#asterisk lowtek (n=anonymou@mail.heavylogic.com) |
| 22:16.28 | lowtek | Hi all. I'm implmenting some enhanced feature keys with some polycom 550's. The softkeys I've defined work great. I'm setting one up to transfer directly to voicemail where a users extension is 805, their direct to voicemail extension is 6805. The phone prompts me for the extension correctly, but it won't let me hit my DirectToVoicemail button then another button with a defined contact (watch buddy). Anybody know how to acheive this functional |
| 22:16.48 | *** join/#asterisk blkry (n=blkry@64.147.222.130) |
| 22:17.32 | cvnet | if incoming call from DID is SIP which hits the asterisk box, and your outbound provider is h323 would asterisk translate it by itself? |
| 22:24.35 | *** part/#asterisk juanIMP (n=Juancho@200.71.41.22) |
| 22:30.36 | KyleK | whats the difference between sln and wav? |
| 22:32.08 | watchy | how do you specifiy a port for an outgoing IAX connection |
| 22:34.47 | bmoraca | does anyone have experience with Adtran Total Access 900 gateways? |
| 22:34.58 | KyleK | watchy: you cant just specify the port in iax.conf? |
| 22:35.05 | Qwell | KyleK: headers |
| 22:35.52 | watchy | KyleK: hmm well i have 2 IAX servers, i want 1 to be on the default port and one on the other |
| 22:36.01 | watchy | but i'm not sure how to tell IAX to connect to a certain port |
| 22:37.31 | KyleK | [iax1] hostname=xxxx port=yy [iax2] hostname=xxxx port=yy+2 |
| 22:37.52 | watchy | ah |
| 22:38.02 | watchy | lemme try dat |
| 22:38.52 | watchy | your my hero |
| 22:38.56 | watchy | you want smooches now? |
| 22:43.28 | watchy | man i got 3 phone systems in 3 locations all connected together with like 100 peers total |
| 22:44.02 | watchy | what codec should i be using for IAX? |
| 22:48.10 | *** join/#asterisk seanmh (n=johndoe@198.59.129.24) |
| 22:51.13 | Fabian- | same codec you use for all other connections |
| 22:51.19 | Fabian- | so you won't have to transcode |
| 23:12.33 | *** join/#asterisk jmacz (n=jmacz@190.144.75.22) |
| 23:13.56 | Meaw | why do i get this error when i load the module g729? o': /usr/lib/asterisk/modules/codec_g729-ast16-gcc4-glibc-core2.so: cannot restore segment prot after reloc: Permission denied |
| 23:20.35 | *** join/#asterisk juanIMP (n=Juancho@200.26.152.222) |
| 23:22.14 | *** join/#asterisk ingenius (n=alektro@host90.190-230-73.telecom.net.ar) |
| 23:22.33 | *** join/#asterisk BadHAL (n=nn@66.194.174.11) |
| 23:22.57 | *** join/#asterisk [intra]lanman (n=lanman@freeswitch/developer/intralanman) |
| 23:25.38 | *** join/#asterisk securevoip (n=securevo@c-76-123-20-170.hsd1.va.comcast.net) |
| 23:26.17 | *** join/#asterisk ruben23 (n=AGENT@122.55.48.242) |
| 23:27.30 | nkohh | anyone use gxp-2000s? (no, I'm not asking for reviews) |
| 23:27.40 | nkohh | I can't figure out how to remove a set admin password without resetting to factory defaults. |
| 23:34.12 | xheliox | nkohh: But we'd be so happy to give you reviews. :D |
| 23:35.19 | *** join/#asterisk freakazoid0223 (n=matt@pool-71-246-17-74.phlapa.fios.verizon.net) |
| 23:36.25 | [TK]D-Fender | nkohh: Of course you aren't supposed to be be able to break in without resetting to factory! That would be crazy |
| 23:43.28 | Meaw | how do you guys run asterisk? i try /usr/sbin/asterisk start |
| 23:43.29 | *** join/#asterisk yo-mama (n=bsumrall@ftnco.com) |
| 23:43.35 | Meaw | but did not work |
| 23:44.43 | *** join/#asterisk jameswf (n=james@unaffiliated/jameswf-home) |
| 23:44.47 | pmhaddad | Meaw, what do you get when you locate asterisk |
| 23:45.06 | pmhaddad | also if you compiled it from had do a make config in the source dir and that will get you the init scripts |
| 23:45.09 | Meaw | tons of files |
| 23:45.40 | pmhaddad | can you just do a asterisk -cvvv ? |
| 23:46.16 | [TK]D-Fender | Meaw: Most use a disto-based init script |
| 23:46.26 | Meaw | pmhaddad, i did |
| 23:46.36 | pmhaddad | Meaw, did it start? |
| 23:46.37 | *** join/#asterisk LakeSolon (n=blake@96-42-127-243.dhcp.roch.mn.charter.com) |
| 23:47.06 | Meaw | nope |
| 23:47.10 | *** join/#asterisk dancarlson_ (n=dancarls@CPE0023df887cc8-CM001ac30febc8.cpe.net.cable.rogers.com) |
| 23:47.25 | pmhaddad | Meaw, error message? |
| 23:47.31 | Meaw | [May 6 02:46:09] WARNING[13169]: translate.c:204 framein: g729tolin did not update samples 0 |
| 23:47.31 | Meaw | Segmentation fault |
| 23:47.35 | pmhaddad | nice |
| 23:47.57 | pmhaddad | what version of *? and i would just try and make clean and recompiling it tbh |
| 23:48.29 | Meaw | meh, it was working fine before i load g729 codec |
| 23:49.22 | pmhaddad | yeah i've never used g729 with asterisk |
| 23:49.43 | Meaw | you should |
| 23:49.49 | dancarlson_ | hey #asterisk. I've been using asterisk on x86 for a while, but now I'm trying to get it to work on an arm-based platform (the sheeva plug). does anyone have any suggestions for resources? |
| 23:49.54 | pmhaddad | look through the options in make menuselect maybe there's something there |
| 23:49.59 | pmhaddad | Meaw, why would i? |
| 23:50.59 | Meaw | for the experience |
| 23:51.00 | Meaw | :) |
| 23:54.30 | *** join/#asterisk blkry (n=blkry@64.147.222.130) |
| 23:54.33 | *** join/#asterisk marc7 (n=marc@bas2-montrealak-1167869366.dsl.bell.ca) |
| 23:54.40 | pmhaddad | Meaw, what version of asterisk is this? |
| 23:55.21 | Meaw | 1.6.0.9 |
| 23:55.27 | pmhaddad | thats what i'm using |
| 23:56.03 | Meaw | im new to asterisk, but im doing a heavy setup |
| 23:56.04 | *** join/#asterisk simprix (n=simprix@c-71-205-52-252.hsd1.mi.comcast.net) |
| 23:56.04 | Meaw | :( |
| 23:56.36 | pmhaddad | Meaw, i assume you have the license for g729? |
| 23:56.42 | Meaw | our E1 card just arrived, we have a free toll number I want this box to forward calls to our voipswitch |
| 23:57.22 | Meaw | nope, not licensed, 30 channels i have to pay like 450$ to get licensed.. |
| 23:57.38 | pmhaddad | riiiight..... so how are you using it then? |
| 23:57.41 | Meaw | i want to test it, if what i have on mind works well.. i would get the licensed g729 |
| 23:57.44 | pmhaddad | ah ok |
| 23:57.47 | Meaw | a cracked one |
| 23:58.05 | pmhaddad | well i can't really verify why asterisk would seg fault like that sorry |
| 23:59.04 | Meaw | that happens after i loaded the codec, now i have no idea how to remove the codec and get asterisk back to work |