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| 01:42.20 | jaytee | it's so quiet I could hear a pin drop 30,000 times through a Grandstream |
| 01:44.15 | *** join/#asterisk brut- (n=brut@h66-173-4-254.mntimn.dedicated.static.tds.net) |
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| 02:10.05 | TrentCreek | [Nov 29 19:51:05] WARNING[17535] chan_sip.c: Maximum retries exceeded on transmission a5f13712-de25844a@192.168.2.2 for seqno 101 (Critical Response)? |
| 02:15.15 | *** join/#asterisk coppice (n=chatzill@33.155.17.210.dyn.pacific.net.hk) |
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| 02:22.59 | TrentCreek | hmmm |
| 02:28.38 | TrentCreek | heeeeeeeeeeeeelllllllllllllooooooooooooo |
| 02:33.45 | *** join/#asterisk tk8 (n=Bryan_Ru@c-68-83-22-225.hsd1.nj.comcast.net) |
| 02:45.16 | TrentCreek | HELLLLLLLLLLLLLLLLLLLLLOOOOOOOOOOOOO |
| 02:45.27 | TrentCreek | My sip phone wont login anymore |
| 02:45.58 | *** join/#asterisk synchris (n=synchris@athedsl-217454.home.otenet.gr) |
| 02:47.32 | Akiyuki | ~cluebat TrentCreek |
| 02:47.34 | jbot | ACTION pulls out a ClueBat (tm) and thwaps TrentCreek. |
| 02:47.57 | TrentCreek | well I would get a clue, but I see no errors |
| 02:48.19 | TrentCreek | and even restarted * |
| 02:48.38 | TrentCreek | I guess server restart time |
| 02:54.02 | TrentCreek | crap..stil won't log in |
| 02:55.53 | *** join/#asterisk UnixDawg_ (n=UnixDawg@cpe-98-28-149-231.cinci.res.rr.com) |
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| 03:34.33 | TrentCreek | well? |
| 03:52.26 | *** join/#asterisk rcy` (n=rcy@S01060002553240a8.vc.shawcable.net) |
| 03:55.38 | TrentCreek | helllllllllllllllllllllllllooooooooooooooooo |
| 03:57.11 | *** join/#asterisk jicksta (n=jicksta@c-24-6-87-187.hsd1.ca.comcast.net) |
| 03:57.27 | *** join/#asterisk oh207 (n=oh207@nylug/member/oh207) |
| 03:58.33 | TrentCreek | well? |
| 04:02.28 | *** join/#asterisk RobertLaptop (n=rmiddle@96.244.48.200) |
| 04:04.53 | baliktad | TrentCreek it appears you have some incorrect expectations about what kind of help and support you can get in this channel |
| 04:05.39 | baliktad | this is not a professional support line, it's just a bunch of interested users. If you have a problem, provide us as much information as you can and be patient. |
| 04:06.51 | jaytee | things like, is the phone registering to a local asterisk server or an ITSP over a nat'd firewall. have you typed sip show peers to see if it's registered. things like that |
| 04:10.24 | baliktad | i love voicepulse for making it cheaper for me to call halfway around the world than to call next door |
| 04:15.07 | TrentCreek | ANYONE? ANYONE? |
| 04:15.44 | baliktad | scroll up TrentCreek |
| 04:15.46 | TrentCreek | it just started doing it..no reason |
| 04:16.06 | TrentCreek | while I was on the phone |
| 04:16.06 | jaytee | started doing what? |
| 04:16.19 | TrentCreek | SIP device cannot connect to server |
| 04:16.30 | TrentCreek | "Registration State:Can't connect to login server" |
| 04:16.48 | jaytee | and it says that on the phone? |
| 04:16.53 | TrentCreek | restarted *, then restarted server |
| 04:16.59 | TrentCreek | in the WWW console |
| 04:17.10 | jaytee | wtf is a WWW console? |
| 04:17.14 | joat | caused by a number of things: fat fingered config file, bad cable, need to reboot phone, etc., etc. |
| 04:17.37 | joat | what do the logs say? |
| 04:17.41 | TrentCreek | checked all of it..there are 2 lines...one is connected to other server, but the line I need is not |
| 04:17.42 | joat | have you run a sniffer? |
| 04:17.45 | TrentCreek | yes |
| 04:17.47 | TrentCreek | and yes |
| 04:17.50 | TrentCreek | nothing in logs |
| 04:18.08 | baliktad | try using your space bar instead of the enter key TrentCreek |
| 04:18.51 | baliktad | if you can't find evidence of any error on the server, it sounds like a firewall or other problem preventing a connection from the phone |
| 04:19.20 | TrentCreek | i can ping the server |
| 04:19.34 | TrentCreek | strange this would happen while I was on the phone |
| 04:19.49 | baliktad | it's only strange because you don't know the cause yet |
| 04:20.02 | joat | arg!!! i'm starting a campaing for package makers: include the ./configure statement with your package! an issue that i've been fighting all day turned out to be caused by my guessing where the lib should go! |
| 04:20.38 | TrentCreek | strange because I see no obvious reason for it to NOT connect |
| 04:20.55 | joat | turns out the 64-bit pre-packaged version of asterisk doesn't look in the directories that compile by default |
| 04:21.17 | joat | err... s/campaing/campaign/ |
| 04:21.17 | baliktad | well start by determining the things that DO work: do you have any other phones that are able to connect/register with your server? |
| 04:22.02 | coppice | joat: the autotools are where the problem lies. they still don't handle 64 bit machines properly |
| 04:22.24 | joat | coppice: i'm learning that lesson the hard way :) |
| 04:23.07 | coppice | joat: I even had to produce an additional autotools macro to be able to handle 64 bit properly. |
| 04:23.11 | joat | if they can't register, drop further back... can you ping the phones... take a look at the IP addresses in the settings... etc... |
| 04:23.31 | TrentCreek | yeah..I guess it's my SIP device..X-Lite seems to connect |
| 04:23.40 | joat | coppice, i've got a bit more to learn... i've been brute forcing it by stracing stuff |
| 04:23.44 | joat | it's slow going |
| 04:23.51 | baliktad | mmmm leftover pumpkin pie, anyone else want a slice? |
| 04:24.05 | TrentCreek | I got enough here |
| 04:24.14 | joat | me! in-law's cat ate ours! |
| 04:24.28 | SkramX | with QueueAddMember.. can I no longer force ackcall=yes? :( :( |
| 04:24.35 | joat | monster figured out how to open the fridge |
| 04:24.44 | baliktad | the cat! wtf?! |
| 04:24.49 | coppice | the cate ate your in-laws? neat |
| 04:25.07 | joat | heh... i wish (have about a 2-day tolerance for them) |
| 04:25.23 | baliktad | i don't like the idea of having in-laws over in the first place, much less their domestic pets |
| 04:25.28 | joat | their cat knows how to open fridges... |
| 04:25.41 | joat | the pumpkin pie was on the bottom shelf... |
| 04:25.54 | SkramX | anyone? |
| 04:25.56 | joat | baliktad, concur... |
| 04:27.51 | joat | SkramX, hang around, someone might answer... |
| 04:28.13 | SkramX | yeah, maybe |
| 04:33.25 | [TK]D-Fender | SkramX: AckCall was only for AgentLogin |
| 04:33.56 | SkramX | I see that |
| 04:34.01 | SkramX | but I want it for QueueAddMember |
| 04:34.19 | [TK]D-Fender | SkramX: You have the source code.... |
| 04:34.21 | SkramX | for example, I dont want a caller to get conneted with an agent's voicemail box |
| 04:34.31 | SkramX | [TK]D-Fender: funneh |
| 04:35.08 | [TK]D-Fender | SkramX: "core show application dial" : M() |
| 04:36.21 | SkramX | ugh |
| 04:36.23 | SkramX | another macro |
| 04:36.40 | SkramX | thanks |
| 04:36.42 | SkramX | will work with it |
| 04:40.05 | *** join/#asterisk TrentCreek (n=kvirc@adsl-69-151-173-38.dsl.hrlntx.swbell.net) |
| 04:40.28 | TrentCreek | anyone else got an idea why SIP device will not register? |
| 04:40.51 | [TK]D-Fender | TrentCreek: And what does the SIP debug say? |
| 04:41.16 | TrentCreek | It's not connect, thus SIP debug has nothing |
| 04:41.23 | TrentCreek | *connecting |
| 04:41.40 | [TK]D-Fender | TrentCreek: Those 2 things have nothing to do with each other |
| 04:41.56 | *** join/#asterisk chuck (n=charlie@wikimedia/cmelbye) |
| 04:41.58 | [TK]D-Fender | TrentCreek: A register attempt can be REFUSED, and darn straight ther'll be debug for it |
| 04:42.14 | [TK]D-Fender | TrentCreek: No debug = networking or config error |
| 04:42.24 | TrentCreek | well if Line 2 on SIP device is reporting "cannot connnect" |
| 04:42.31 | chuck | Hey, are there any easy to use guides to setting up asterisk where I can just get a simple menu to dial extensions, and maybe an informational menu? |
| 04:42.42 | [TK]D-Fender | TrentCreek: that statement also means pretty much nothing. |
| 04:42.45 | joat | what make/model phone? |
| 04:42.45 | TrentCreek | aA softdevice is connecting fine |
| 04:42.53 | TrentCreek | PAP2T |
| 04:42.57 | [TK]D-Fender | TrentCreek: then this other device is set wrong |
| 04:43.11 | TrentCreek | No..I was just using it and the phone went dead |
| 04:43.30 | TrentCreek | Now it wont connect, however line 1 connects fine using another server |
| 04:43.30 | joat | i'd say go back over the config file in the web interface |
| 04:43.32 | [TK]D-Fender | chuck: ... |
| 04:43.34 | [TK]D-Fender | ~book |
| 04:43.35 | jbot | extra, extra, read all about it, book is probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook |
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| 04:44.23 | SkramX | [TK]D-Fender - what should I have this macro do? heh |
| 04:44.33 | chuck | [TK]D-Fender, What?... |
| 04:44.41 | SkramX | would I call the answer() function? i thought itd already be called |
| 04:44.47 | [TK]D-Fender | chuck: See jbot's link to the book |
| 04:45.10 | [TK]D-Fender | SkramX: "core show application dial" <------------ |
| 04:45.22 | *** part/#asterisk chuck (n=charlie@wikimedia/cmelbye) |
| 04:45.35 | joat | is away: Away |
| 04:45.49 | SkramX | sigh |
| 04:49.32 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 04:49.33 | TrentCreek | Bahhhhhhhhhhhh!!!!!!!!!!!!!! |
| 04:49.37 | TrentCreek | "Registration State:Can't connect to login server" |
| 04:51.59 | TrentCreek | I switched line One other to another extension for the server...It can't get in either |
| 04:53.48 | SkramX | [TK]D-Fender - okay, I have it working.. except all I know how ot make it do is say press any key to answer the call. if they don't, then the caller (in queue) gets hung up on.. |
| 04:55.25 | [TK]D-Fender | SkramX: Now is the point where you go read the full list of dialplan apps, and use them |
| 04:56.03 | SkramX | believe it or not, I have been looking |
| 04:56.03 | [TK]D-Fender | SkramX: Not knowing how to make * say something is prtty much ridiculous. |
| 04:56.07 | SkramX | any insight/links |
| 04:56.08 | SkramX | no |
| 04:56.12 | SkramX | i know how to do that of course! |
| 04:56.28 | SkramX | but if the agent *doesnt* want the call, the user gets hung up on |
| 04:56.43 | [TK]D-Fender | SkramX: "core show application dial" <------------ |
| 04:56.59 | [TK]D-Fender | SkramX: You seem to have a serious reading block |
| 04:57.21 | [TK]D-Fender | SkramX: Because there is BIG PRINT telling you how you can have it react |
| 04:58.08 | SkramX | ok |
| 04:58.57 | SkramX | MACRO_RESULT is what I want to set. it seems |
| 05:00.27 | *** join/#asterisk JohnnyBeGood (n=JohnnyBe@c-98-232-40-217.hsd1.wa.comcast.net) |
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| 05:36.15 | *** join/#asterisk JeremyHimself (n=jer@96-26-210-96.sea.clearwire-dns.net) |
| 05:37.22 | JeremyHimself | Noob here... (Hi). Can anyone answer a few 'starter' questions? Thx in advance... |
| 05:38.48 | jaytee | ~ask |
| 05:38.49 | jbot | methinks ask is Questions in the channel should be specific, informative, complete, concise, and on-topic. Don't ask if you can ask a question first. Don't ask if a person is there; just ask what you intended to ask them. Better questions more frequently yield better answers. We are all here voluntarily or against our will. |
| 05:41.10 | rcy` | lemme outta here |
| 05:41.12 | rcy` | please |
| 05:41.34 | jaytee | no one here gets out alive |
| 05:42.40 | JeremyHimself | :) 10-4. OK well actually my brother and I recently started a corporation, but I live in Seattle & he lives in Los Angeles. I have a hosting acct w/ GoDaddy where all my domains are... Can I upload the asterisk software to my server and ultimately have a private communication 'portal' with my bro (or anyone else, for that matter) over IP? |
| 05:43.09 | [TK]D-Fender | JeremyHimself: No need for * for that |
| 05:43.33 | JeremyHimself | How else? |
| 05:43.43 | [TK]D-Fender | JeremyHimself: Any direct VoIP client. |
| 05:44.04 | [TK]D-Fender | JeremyHimself: take your pick of dozens of protocols & hundreds or clients |
| 05:44.41 | [TK]D-Fender | JeremyHimself: Skype may be for you. |
| 05:46.55 | JeremyHimself | Meaning Skype, Gizmo, etc? We've tried those but essentially what I'd like to do is have a setup that's (for all intents and purposes) a POTS system,(with extensions), but I'd rather have a private system of my own... |
| 05:49.45 | [TK]D-Fender | JeremyHimself: Well * can be used as a PBX with connectivity to a variaty of hardware, lines, etc. |
| 05:49.47 | JeremyHimself | (Just a personal preference I suppose... I just don't like having to rely on the proprietary protocols that Skype has). I know u said no need to use * in my case, but is it possible? |
| 05:50.13 | [TK]D-Fender | JeremyHimself: So go sit down with the book, go install & learn it, and play around and see if its what you're looking for |
| 05:50.18 | [TK]D-Fender | ~book |
| 05:50.19 | jbot | from memory, book is probably Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF at http://www.asteriskdocs.org --- HTML at http://tfot.leifmadsen.com or see ~buybook |
| 05:51.05 | JeremyHimself | Will do. Thanks so much for the info. |
| 05:51.06 | [TK]D-Fender | JeremyHimself: If you want expandability and potential access to the PSTN, then * is a great tool |
| 05:51.53 | *** join/#asterisk CunningPike (n=arodgers@S010600095b33697f.vc.shawcable.net) |
| 05:51.55 | JeremyHimself | Can I access the PSTN via * even if it's installed on a remote host (like GoDaddy or whoever else)? |
| 05:53.34 | JeremyHimself | That is, using software only (on their end)? |
| 05:54.55 | hesco | I seem to be getting cdr of my voicemail, but not my .call files. How do I get the cdr log written to by the outgoing calls, as well? |
| 05:55.51 | *** part/#asterisk JeremyHimself (n=jer@96-26-210-96.sea.clearwire-dns.net) |
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| 06:05.14 | Nugget | owes file a muffin. |
| 06:05.42 | *** join/#asterisk newmember (n=chatzill@S010600036d1139bb.cg.shawcable.net) |
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| 06:27.16 | troy- | can multiple devices/clients register with the same username/pass? |
| 06:27.55 | *** join/#asterisk thansen (n=thansen@72.2.166.250) |
| 06:28.18 | [TK]D-Fender | troy-: No |
| 06:28.35 | [TK]D-Fender | troy-: Well techincally yes, but only the latest one will get calls. |
| 06:30.52 | troy- | danke |
| 06:38.45 | TrentCreek | anyone else? |
| 06:40.23 | TrentCreek | anymore Ideas as to why my PAP2T cannot connect? |
| 06:41.07 | *** join/#asterisk chuck (n=charlie@wikimedia/cmelbye) |
| 06:42.15 | chuck | how do I record a menu to use with asterisk? |
| 06:43.45 | Daejeo | Meow :) |
| 06:43.54 | Daejeo | Katty |
| 06:44.14 | Daejeo | see ~Katty |
| 06:46.33 | *** join/#asterisk enyawix (n=enyawix@68-114-138-145.dhcp.jcsn.tn.charter.com) |
| 06:47.46 | chuck | ~Katty |
| 06:47.47 | jbot | i heard katty is the only girl in the channel, so be nice to her |
| 06:48.01 | enyawix | any distro lend itself more to being a voip server/ |
| 06:48.34 | chuck | anyone know how to record menus? |
| 06:56.53 | [TK]D-Fender | TrentCreek: I've anready answered you |
| 06:57.33 | [TK]D-Fender | TrentCreek: if its on the same subnet as the soft-phone that does connect yet it does not get through at all then you ahve configured it wrong |
| 06:57.34 | TrentCreek | yes, but no solutions as to why this device cannot connect to this particular server |
| 06:58.09 | TrentCreek | the configeration just cannot change after ben using it for 3 months |
| 06:58.29 | [TK]D-Fender | TrentCreek: Sure it can, if it was once a locked ATA and has updated itself |
| 06:58.53 | TrentCreek | I tried line two that was connecting to another server and it was working fine. |
| 06:58.55 | [TK]D-Fender | TrentCreek: Amongst the various other ways of resetting the config |
| 06:59.08 | TrentCreek | The configs were basically the same |
| 06:59.12 | [TK]D-Fender | TrentCreek: All of this "was' means nothing. Go prove your settings NOW. |
| 06:59.19 | TrentCreek | yeah and I did |
| 06:59.24 | TrentCreek | same as the softphone |
| 06:59.35 | [TK]D-Fender | TrentCreek: stop talking in the past tense. Says you have no clue how it is currently set. |
| 06:59.51 | TrentCreek | yes..I DID look |
| 06:59.58 | [TK]D-Fender | TrentCreek: It is clearly done wrong if it doesn't get a packet out. |
| 07:00.03 | TrentCreek | and went over it several times |
| 07:00.11 | [TK]D-Fender | TrentCreek: this is not a guessing game |
| 07:00.22 | TrentCreek | all I have to do is change the server name and it will connect to the other |
| 07:00.42 | TrentCreek | Out of the blue it just stopped |
| 07:00.43 | [TK]D-Fender | TrentCreek: Show us something substantial |
| 07:00.51 | TrentCreek | such as? |
| 07:01.00 | *** part/#asterisk endemic (n=endemic@orion.onvox.net) |
| 07:01.03 | [TK]D-Fender | TrentCreek: You don't know what would count as substantial? |
| 07:01.09 | *** join/#asterisk endemic (n=endemic@orion.onvox.net) |
| 07:01.29 | TrentCreek | no. since there is no obvious reason why it will not connect to the server I want |
| 07:01.55 | [TK]D-Fender | TrentCreek: more unfounded commentary... BACK IT UP |
| 07:02.05 | TrentCreek | i todl you I did already |
| 07:02.15 | [TK]D-Fender | TrentCreek: Where? |
| 07:02.16 | TrentCreek | it connects fine to another server using the EXACT settings |
| 07:02.28 | [TK]D-Fender | TrentCreek: you aren't showing us anything. |
| 07:02.32 | TrentCreek | the same settings I have been using |
| 07:02.33 | chuck | How do I convert a .raw file into .gsm? |
| 07:04.01 | TrentCreek | [TK]D-Fender: what would you like to see? |
| 07:04.15 | [TK]D-Fender | TrentCreek: Show us something substantial <- |
| 07:04.27 | TrentCreek | such as what? |
| 07:05.22 | [TK]D-Fender | TrentCreek: CONFIGS, status screens, network debug, hostname lookups, ping attempts from similary local devices.. HOLY CRAP you can't think of ANYTHING to offer to debug this on your own? |
| 07:05.34 | [TK]D-Fender | TrentCreek: backup |
| 07:06.02 | [TK]D-Fender | ~pb |
| 07:06.03 | jbot | [~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/ |
| 07:06.27 | [TK]D-Fender | www.imagebin.com |
| 07:06.30 | TrentCreek | no..since I have looked over all I could think of and there is no OBVIOUS reason for it to not connect |
| 07:07.05 | chuck | anyone know? |
| 07:07.55 | [TK]D-Fender | TrentCreek: Rght now you are running in circles like a twit showing us nothing. You screwed up and YOU can't tell where. What good are your eyes once they've failed to this point? |
| 07:08.11 | [TK]D-Fender | TrentCreek: there's nothing obvious because you've shown nothing. |
| 07:08.13 | TrentCreek | [TK]D-Fender: I have done all that...everything connects fine to the server |
| 07:08.23 | TrentCreek | EXCEPT that PAP2T |
| 07:08.45 | [TK]D-Fender | TrentCreek: And as I said, you've screwed something up, so show us all the damn settings. |
| 07:09.01 | chuck | no one? |
| 07:09.05 | [TK]D-Fender | TrentCreek: If you did everything right, it would WORK |
| 07:09.18 | TrentCreek | how can I "screw" something up by using it the same way for the past 3 months? |
| 07:09.19 | [TK]D-Fender | TrentCreek: So get off your ass and show us or we can't help you |
| 07:09.27 | [TK]D-Fender | \treStop living in the past! |
| 07:09.46 | TrentCreek | not doing that..I am merely reporting previous results |
| 07:10.13 | [TK]D-Fender | TrentCreek: You are wasting our time by turning a blind eye and not showing us. You clearly don't want to solve anything. |
| 07:10.22 | [TK]D-Fender | TrentCreek: You must have all the answers. Good luck with that |
| 07:10.31 | TrentCreek | well then thanks |
| 07:10.53 | TrentCreek | because I dont knwo what to show since everything looks fine |
| 07:11.10 | [TK]D-Fender | TrentCreek: You don't know what to show? I just told you . |
| 07:11.14 | TrentCreek | nothing has changed in the settings |
| 07:11.20 | chuck | please? does anyone know how to convert from raw to gsm? |
| 07:11.36 | [TK]D-Fender | TrentCreek: What does that song & dance tell us? |
| 07:12.20 | TrentCreek | [TK]D-Fender: Here ya go...PAP2T line 2 status |
| 07:12.24 | TrentCreek | "Registration State:Can't connect to login server" |
| 07:12.36 | [TK]D-Fender | TrentCreek: USELESS |
| 07:12.42 | [TK]D-Fender | TrentCreek: Try again. |
| 07:12.57 | TrentCreek | exactly..that is all I have to go on because the softphoen is connecting from the SAME subnet |
| 07:13.15 | [TK]D-Fender | TrentCreek: SHOw. the CONFIGS and all the other backup I asked for. |
| 07:15.30 | TrentCreek | [TK]D-Fender: http://www.pastebin.ca/1270931 |
| 07:16.37 | [TK]D-Fender | TrentCreek: keep going... |
| 07:24.12 | chuck | How do I make my Asterisk installation know where to forward SIP calls? I'm connected with an SIP client, but it's not ringing at all |
| 07:25.15 | chuck | oo |
| 07:25.28 | chuck | apparently calling them directly works, but my little WaitExten one isn't |
| 07:26.07 | chuck | anyone know why that would happen.. |
| 07:26.56 | *** join/#asterisk kamanashisroy (n=kamanash@119.30.34.10) |
| 07:29.01 | TrentCreek | [TK]D-Fender: At this point I just dotn know |
| 07:30.57 | [TK]D-Fender | TrentCreek: I told you several things to provide and you haven't done much. |
| 07:30.59 | TrentCreek | if I could just see what is going on with it tryign to connnect |
| 07:31.27 | TrentCreek | ping is working fine |
| 07:31.35 | [TK]D-Fender | TrentCreek: You have not shown any configs or related debug. |
| 07:31.58 | TrentCreek | okay what is the network debug? |
| 07:32.05 | [TK]D-Fender | chuck: PASTEBIN is your friend.... |
| 07:32.07 | [TK]D-Fender | ~pb |
| 07:32.08 | jbot | [~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/ |
| 07:32.41 | [TK]D-Fender | TrentCreek: You have shown only 1 tiny piece of what I asked for. Go do the rest. |
| 07:32.43 | chuck | Oh, my SIP clients weren't registering correctly apparently, but it's working great now. |
| 07:33.00 | [TK]D-Fender | chuck: Glad to hear |
| 07:34.04 | chuck | [TK]D-Fender, Do you happen to know how to convert from .raw to .gsm? |
| 07:34.35 | [TK]D-Fender | chuck: Where did you get "raw" from? |
| 07:34.48 | [TK]D-Fender | chuck: First tool of choice is sox |
| 07:34.50 | chuck | Dictate |
| 07:37.14 | TrentCreek | [TK]D-Fender: the ifconfig http://www.pastebin.ca/1270939 |
| 07:37.26 | [TK]D-Fender | TrentCreek: I still see no configs... |
| 07:39.51 | LeddyHM | Will this ring the users phone and if they pick up then dial the requested number? http://www.voip-info.org/wiki/view/Asterisk+manager+dialou |
| 07:40.16 | [TK]D-Fender | LeddyHM: Yes |
| 07:40.51 | LeddyHM | sweet |
| 07:41.41 | hesco | If my modules.conf includes autoload=yes; what list of modules are automatically loaded? Anything configured in /etc/*/ ??? |
| 07:47.17 | [TK]D-Fender | hesco: everything in the MODULES folder. |
| 07:50.07 | TrentCreek | hmmmhow to get a whole browser screen shot......... |
| 07:51.27 | LeddyHM | hrm |
| 07:52.18 | LeddyHM | I have an extension 100 and 100remote one for the desk the other for soft phone(s). When I use SIP/100 I get nothing, I have to used 100remote as the extension |
| 07:52.35 | LeddyHM | anyway I can use SIP/100 and it will ring both as all other calls do? |
| 07:53.18 | LeddyHM | in reference to the "originate" |
| 07:54.16 | [TK]D-Fender | LeddyHM: SIP/100 is *on* device |
| 07:54.30 | [TK]D-Fender | one |
| 07:54.50 | LeddyHM | correct |
| 07:55.12 | LeddyHM | extensions.conf will ring both lines on incoming calls |
| 07:55.16 | [TK]D-Fender | LeddyHM: So calling it will not make other devices ring |
| 07:55.18 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 07:56.52 | LeddyHM | bummer |
| 07:56.57 | LeddyHM | not the answer I was looking for |
| 07:57.30 | LeddyHM | I was hoping it would follow the dialplan rules |
| 07:59.25 | [TK]D-Fender | LeddyHM: "SIP/100" is not dialplan. It is a single device... maybe you should consider the kind of CHANNEL you're calling... |
| 07:59.50 | [TK]D-Fender | *hint* |
| 07:59.56 | LeddyHM | I know |
| 08:00.12 | LeddyHM | the dial plan has a 100 extension that will then ring those 2 devices |
| 08:12.03 | hesco | I've got cdr logging to a pg db for inbound in one context, but not for my .call file outbound. Any idea why that might be? |
| 08:12.43 | chuck | Can sox convert raw files into gsm files? |
| 08:13.18 | hesco | I thought for a moment I somehow needed to enable cdr, but its working for the inbound ivr calls. |
| 08:16.46 | chuck | Can anyone at least point me in the direction of a way to easily record menus? |
| 08:23.49 | hesco | chuck: How I did it is this: (1) script the entire interaction. (2) Have the server call you and using the Record() function make a recording of your voice. (3) scp the resultant wav file (my are in /tmp) to a localbox. (4) open with audacity and cut each menu element from the raw audio, trimming each cut, saving and installing each cut (/var/lib/asterisk/sounds/custom/irv/ in my world); (5) using Background() and WaitExten() functions |
| 08:23.49 | hesco | dialplan, (6) test refine |
| 08:24.56 | chuck | okay, i've figured out Record so I've gotten *something* up for now, and I'll script it and everything later |
| 08:25.43 | chuck | bah, I shutdown my lpatop but it is still connected to asterisk somehow |
| 08:25.47 | *** join/#asterisk feeds (n=feeds@85-135-247-77.adsl.slovanet.sk) |
| 08:25.48 | hesco | make it easy on yourself, when recording take long pauses between elements of the script. |
| 08:26.37 | chuck | hesco, do you know how I would make it so that if the person can't be reached after WaitExten, a little unable to be reached sound is played? |
| 08:27.43 | hesco | core show application wait |
| 08:29.01 | hesco | perhaps FoolowMe would help |
| 08:30.22 | hesco | look at option n |
| 08:30.52 | hesco | s/FoolowMe/FollowMe/ |
| 08:38.09 | chuck | :D awesome, I've got a rockin' little phone system setup, tomorrow I'll actually make a real menu scheme |
| 08:38.38 | chuck | hesco, I remember there was a site where you could get a phone number for your SIP server for free or something? do you know any sites like that? |
| 08:47.31 | hesco | google: sip provider free, if such exist, google will find them |
| 08:47.44 | hesco | what do you know about cdr? |
| 08:48.44 | TrentCreek | everything |
| 08:55.28 | [TK]D-Fender | ok, I'm done.. later all |
| 09:23.59 | *** join/#asterisk Dovid (n=chatzill@tony09-118-62.inter.net.il) |
| 09:24.07 | Dovid | tes |
| 09:24.28 | Dovid | anyone have any thing to say about Mitel phones ? Good ? Bad ? |
| 09:36.40 | drmessano | Scary |
| 09:46.29 | TrentCreek | not so sure anymore |
| 09:53.39 | TrentCreek | http://i317.photobucket.com/albums/mm361/tcreek/Asteriskk/Window2.jpg |
| 09:54.37 | TrentCreek | http://i317.photobucket.com/albums/mm361/tcreek/Asteriskk/Windows1.jpg |
| 09:54.45 | TrentCreek | http://i317.photobucket.com/albums/mm361/tcreek/Asteriskk/Window3.jpg |
| 10:10.44 | *** part/#asterisk Ericounet (n=Ericoune@ACaen-151-1-11-187.w86-215.abo.wanadoo.fr) |
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| 10:45.55 | Dovid | anyone know how to set the registration period on a Polycom ? I want it to register every 3 minutes |
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| 11:52.43 | Dovid | anyone know how to set the registration period on a Polycom ? I want it to register every 3 minutes |
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| 12:28.02 | Muhammad | I'm trying to accomplish an Attendant Call transfer between Asterisk GW and my Sip Proxy (OpenSER ) but , But i have faced a problem with this a detailed description for the problem and debug are found in this url http://lists.kamailio.org/pipermail/users/2008-November/020793.html |
| 12:42.33 | *** join/#asterisk Shotygun (n=thorn@213.31.43.3) |
| 12:46.49 | Shotygun | Hi, can somebody tell me if there is a chance that the correct USA dialplan is 1ZXXNXXXXXX and not 1NXXNXXXXXX ? There is a claim of area codes 177 & 150 which I'm trying to understand if this is true |
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| 14:24.14 | *** mode/#asterisk [+o russellb] by ChanServ |
| 14:28.09 | echinos | When * answers, it does a background play, so you can dial an extension, but I tried dialing a two-digit extension, and it immediately tries to connect to the first number instead of waiting for both |
| 14:29.08 | echinos | ie. I have an extension 10 set up, but if I dial 10, it accepts the 1 and doesn't wait for the zero... |
| 14:30.29 | russellb | echinos: Put a WaitExten() after Background. |
| 14:31.05 | echinos | Ah, ok. I was under the impression that you didn't need one. thx. |
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| 14:33.30 | russellb | echinos: np |
| 14:33.35 | echinos | hmm... same thing happened :/ |
| 14:33.53 | russellb | do you have an extension '1' ? |
| 14:33.59 | echinos | no |
| 14:34.35 | echinos | does it proceed to the waitexten if the sound file from background is still playing? |
| 14:34.48 | russellb | no |
| 14:34.57 | russellb | background is going to always exit after the first digit |
| 14:35.03 | echinos | ok, so background is doing it |
| 14:35.33 | russellb | but with waitexten, it should then sit there and let you finish dialing. |
| 14:35.54 | echinos | Ok, so basically don't use background if you want to dial 2 digits. :) |
| 14:36.53 | echinos | I guess it's more for menus that you hear when you first call that only use 1 digit, you usually have to pick an option to go to a submenu where you can dial a multidigit extension |
| 14:38.01 | russellb | you can use background |
| 14:38.13 | russellb | there's nothing wrong with that ... it's just going to stop playing after the first digit |
| 14:38.21 | russellb | presumably the caller isn't listening anymore at that point anyway |
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| 15:05.48 | *** join/#asterisk sw (n=sw@unaffiliated/sw) |
| 15:06.12 | sw | hi |
| 15:06.15 | sw | ast_func_read: Function LEN not registered |
| 15:06.24 | sw | which module is responsible for function LEN ? |
| 15:06.32 | seanbright | func_string.so |
| 15:06.34 | seanbright | i think |
| 15:06.44 | sw | seanbright, k |
| 15:06.57 | seanbright | func_strings.so |
| 15:07.01 | seanbright | plural. |
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| 15:08.08 | *** join/#asterisk _ShrikE (n=_ShrikE@adsl-074-185-215-060.sip.msy.bellsouth.net) |
| 15:08.43 | russellb | it should be func_stringz.so |
| 15:09.00 | seanbright | no it shouldn't. |
| 15:09.02 | seanbright | f_str.so |
| 15:09.03 | sw | seanbright, awesome ! thanks :) |
| 15:09.09 | sw | plural sucks |
| 15:09.28 | sw | i mean, plural suck |
| 15:11.26 | russellb | sucks isn't plural, heh |
| 15:11.50 | seanbright | plurals only apply to nouns |
| 15:12.00 | russellb | wonders what a suck is |
| 15:12.09 | seanbright | english is my 3rd language |
| 15:12.14 | seanbright | esperanto was my first |
| 15:12.32 | coppice | russellb: life has been bad if you haven't experienced one :-) |
| 15:12.38 | russellb | lol |
| 15:12.46 | seanbright | well played. |
| 15:12.46 | sw | the ending smiley was implied :) |
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| 15:36.06 | jaytee | esperanto? I firmly believe more people speak fluent Klingon than Esperanto on this planet. |
| 15:37.28 | coppice | they used to have meetings of an esperanto speakers group in a room at my old college, and the direction signs they put up were always in english. didn't seem to show a lot of faith in the language. |
| 15:38.12 | jaytee | lol |
| 15:38.18 | jaytee | If Rosetta Stone doesn't carry it as a choice, it can't be for real. |
| 15:38.25 | coppice | "Mensa group meeting. Room 201. Second floor" always amused me, too |
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| 15:50.18 | xacatecas | hi all, a quick question, is it normal for chan_mobile to disconnect/reconnect to your phone directly after each call? |
| 15:51.01 | xacatecas | i must say ... getting it up was a breeze and I had zero bluez knowledge to begin with! |
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| 15:54.15 | hesco | I've got cdr logging to a pg db for inbound in one context, but not for my .call file outbound. Any idea why that might be? |
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| 15:58.00 | *** mode/#asterisk [+o d3wayne] by ChanServ |
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| 17:02.18 | Moudmen | my disa is not taking any input when the callback runs. what can be the cause of that problem ? |
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| 18:10.30 | [TK]D-Fender | Moudmen: Wrong DTMF mode, poor audio quality if its inband, lack of having called an Answer prior, Always good to play a sound as well even if its "silence" |
| 18:12.11 | *** join/#asterisk synchris (n=synchris@athedsl-217454.home.otenet.gr) |
| 18:12.42 | Moudmen | well i set dtmfmode to auto in the sip trunk's settings. now it's taking input from a landline but not a mobile |
| 18:15.21 | [TK]D-Fender | Moudmen: Both calls using the same peer? |
| 18:15.26 | Moudmen | yep |
| 18:15.38 | [TK]D-Fender | Moudmen: And "auto" is not necessarily reliable. |
| 18:15.55 | Akiyuki | What is the best way to have an asterisk server and multiple SIP devices on the same network? For portforwarding |
| 18:15.55 | [TK]D-Fender | Moudmen: This does also sound like it might be an inband quality issue as well |
| 18:16.10 | [TK]D-Fender | Akiyuki: ? |
| 18:16.28 | Akiyuki | oh |
| 18:16.29 | Akiyuki | n/m |
| 18:16.31 | Moudmen | okay i'll try to run callback on a phone from another country |
| 18:16.32 | Akiyuki | That makes no sense |
| 18:18.22 | Moudmen | do i have to have relaxdtmf in zapata.conf ? |
| 18:20.56 | *** join/#asterisk esaym (n=user@cpe-70-120-89-6.satx.res.rr.com) |
| 18:23.44 | [TK]D-Fender | Moudmen: You said its coming in via SIP |
| 18:24.08 | Moudmen | i just read somewhere something about relaxdtmf, so i thought maybe ... |
| 18:24.57 | *** join/#asterisk jicksta (n=jicksta@c-24-6-87-187.hsd1.ca.comcast.net) |
| 18:26.09 | [TK]D-Fender | Moudmen: Don't think that there is an equivalent in SIP... go try it.... |
| 18:26.33 | [TK]D-Fender | Moudmen: Check out the WIKI... if its listed there, its good odds that its available |
| 18:26.36 | *** join/#asterisk danalien (n=danalien@unaffiliated/danalien) |
| 18:26.46 | Moudmen | okay, will do. thanks for the advice |
| 18:34.29 | *** join/#asterisk qdk (n=qdk@94.191.252.224.bredband.3.dk) |
| 18:38.10 | Akiyuki | oo baby i like it raw .... yeah baby i like it raw |
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| 18:57.31 | *** join/#asterisk dlewis (i=4578ad9f@about/security/staff/dlewis) |
| 18:57.47 | dlewis | anyone here have asterisk set up with a grandstream ht503? |
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| 18:59.07 | [TK]D-Fender | dlewis: Have you gone and bought one? |
| 18:59.46 | dlewis | [TK]D-Fender: yes. I'm having issues with CID. |
| 19:00.27 | [TK]D-Fender | dlewis: I doubt you find almost anyone here with one... |
| 19:00.40 | [TK]D-Fender | dlewis: Take alook on www.voxilla.com 's forums |
| 19:04.52 | dlewis | [TK]D-Fender: i have. looks like a lot of people have CID issues with the ht503. |
| 19:05.03 | [TK]D-Fender | dlewis: What country? |
| 19:05.09 | dlewis | US |
| 19:05.18 | dlewis | looks like no one has 100% figured it out |
| 19:05.34 | [TK]D-Fender | dlewis: Thats really bad to hear... if you have problems with US CID then it really doesn't bode well for the device as a whole.. |
| 19:06.04 | [TK]D-Fender | dlewis: It understandable if its being used in countries with less common or more troublesome signalling, etc.. |
| 19:06.32 | dlewis | i'm thinking about getting the Linksys SPA-3102 as a replacement... |
| 19:06.42 | dlewis | i want to 100% make sure I can't get this to work first... |
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| 19:12.25 | Akiyuki | Stupid net splits |
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| 19:24.30 | *** join/#asterisk endemic (n=endemic@orion.onvox.net) [NETSPLIT VICTIM] |
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| 19:25.50 | *** join/#asterisk fskrotzki_ (n=fskrotzk@cpe-74-74-245-250.rochester.res.rr.com) |
| 19:27.52 | *** join/#asterisk Thorn (n=thorn@unaffiliated/thorn) |
| 19:28.24 | *** join/#asterisk tzafrir_laptop (n=tzafrir@local.xorcom.com) |
| 19:31.07 | *** join/#asterisk edoceo (n=edoceo@c-71-197-244-147.hsd1.or.comcast.net) |
| 19:39.56 | *** join/#asterisk brut- (n=brut@h66-173-4-254.mntimn.dedicated.static.tds.net) |
| 19:54.27 | *** join/#asterisk chuck (n=charlie@wikimedia/cmelbye) |
| 19:54.39 | chuck | did anyone get those last few messages I sent? |
| 19:55.03 | SkramX | didnt se aything |
| 19:55.17 | chuck | Is there any way to use a dial-up modem or something to be able to call in to asterisk with a normal phone? Probably a really noobish question :P |
| 19:55.18 | chuck | ^ there |
| 19:55.22 | chuck | (irssi was failing) |
| 19:55.50 | [TK]D-Fender | chuck: No. |
| 19:55.58 | chuck | ah :( |
| 19:56.00 | [TK]D-Fender | chuck: You need a properly supported FXO device |
| 19:56.11 | chuck | and how would I make an extension to respond to the pound key? |
| 19:56.39 | [TK]D-Fender | chuck: clarify "respond". Respond WHEN? |
| 19:56.52 | chuck | I mean like in a WaitExten() thing |
| 19:56.56 | chuck | I'm trying to make a menu |
| 19:56.58 | chuck | ack |
| 19:57.02 | [TK]D-Fender | chuck: exten => #,1,Blah() |
| 19:57.52 | dlewis | [TK]D-Fender: would you happen to know the dial plan for *82? |
| 19:58.02 | dlewis | I need to be able to dial *82-X-XXX-XXX-XXXX |
| 19:58.19 | dlewis | but, there needs to be a pause after *82 so that the telco can confirm with a tone |
| 19:58.34 | SkramX | exten => _*82XXXXXXXXXXX,1,Blah() ; i'm pretty sure |
| 19:58.40 | SkramX | oh |
| 19:58.52 | [TK]D-Fender | dlewis: Dialplan is an * thing. This is not to do with "telco" |
| 19:59.07 | *** join/#asterisk geoff_k (n=geoff@host217-44-157-205.range217-44.btcentralplus.com) |
| 19:59.21 | [TK]D-Fender | dlewis: Dialplan is YOU calling from your phone. Calling out the the PSTN is another matter entirely |
| 19:59.23 | SkramX | you can simulate a tone though and then just pass the whole string to your trunk when the user enters all the digits required to make the call, right? |
| 19:59.43 | dlewis | [TK]D-Fender: ok, so how would I proceed then for this situation? |
| 20:00.17 | [TK]D-Fender | dlewis: If you're talking about using that HT503, then you'd use the D() parameter of Dial. |
| 20:00.48 | dlewis | hmm |
| 20:00.49 | SkramX | still can't figure out how to use the M() parameter *hides* |
| 20:01.21 | dlewis | where can I read on these parameters? |
| 20:01.31 | [TK]D-Fender | dlewis: "core show application dial" |
| 20:01.54 | dlewis | ok |
| 20:02.10 | SkramX | [TK]D-Fender has that command on his clipboard ;) |
| 20:03.18 | dlewis | [TK]D-Fender: the D() seems to be after the call is answered |
| 20:03.31 | chuck | hrm, I'm just getting this when I call my new extension: |
| 20:03.32 | chuck | <PROTECTED> |
| 20:03.41 | [TK]D-Fender | dlewis: Perhaps you should look at how the call is treated by your ATA... this may well work |
| 20:03.45 | chuck | snippet of the dialplan: |
| 20:03.46 | chuck | exten => 123,1,Answer() |
| 20:03.46 | chuck | exten => GoTo(124, 1) |
| 20:03.46 | chuck | exten => 124,1,Background(welcome) |
| 20:04.00 | dlewis | ok |
| 20:04.13 | [TK]D-Fender | chuck: and your 2nd line there has no priority or exten in it |
| 20:04.29 | chuck | >< woops |
| 20:04.37 | SkramX | exten => 123,2,GoTo(124, 1) |
| 20:04.48 | [TK]D-Fender | and NO spaces in your parms like that |
| 20:16.13 | *** join/#asterisk Bananaskin (n=Banana@94-193-31-47.zone7.bethere.co.uk) |
| 20:21.03 | *** join/#asterisk telecos (n=sergio@67.166.219.87.dynamic.jazztel.es) |
| 20:34.00 | *** join/#asterisk r0land (n=r0land@212.36.209.1) |
| 20:34.02 | r0land | hi all |
| 20:34.06 | r0land | hello [TK]D-Fender |
| 20:34.45 | *** part/#asterisk Akiyuki (i=TuxGuy@cpe-065-184-194-136.ec.res.rr.com) |
| 20:34.45 | *** join/#asterisk Akiyuki (i=TuxGuy@cpe-065-184-194-136.ec.res.rr.com) |
| 20:35.01 | *** join/#asterisk xuser (n=JJ@unaffiliated/xuser) |
| 20:35.09 | r0land | i'm facing a weird prob when dialing my pstn through my softphone. each call at exactly 1 min and 05 seconds. i stop hearing anything on the softphone. when i investiage i notice that asterisk is disconnecting my channel! this is the CLI output: http://pastebin.com/d58f8e944 any explanation ? |
| 20:38.11 | [TK]D-Fender | r0land: Networking issue of some kind |
| 20:40.44 | r0land | oh ok |
| 20:40.46 | r0land | thank you :) |
| 20:41.41 | r0land | one more question [TK]D-Fender may i use rtptimeout and rtpholdtimeout togehter? |
| 20:41.48 | r0land | or theyre the same thing? |
| 20:42.12 | r0land | the reason am considering this, is that sometimes if the caller is set on hold, asterisk considers it as silence and breaks the call |
| 20:42.15 | [TK]D-Fender | r0land: You should disable CNG entirely and this shouldn't be an issue |
| 20:42.25 | [TK]D-Fender | r0land: And keep wualify on. |
| 20:42.31 | r0land | qualify= yes |
| 20:42.39 | r0land | on all sip accounts (general context) |
| 20:42.53 | r0land | could u explain please whts CNG! wht does it stand for so i know wht to look for |
| 20:43.17 | *** join/#asterisk ballongen (n=linus@c-e9ace455.23-0167-74657210.cust.bredbandsbolaget.se) |
| 20:44.28 | [TK]D-Fender | ~cng |
| 20:44.30 | [TK]D-Fender | ~vad |
| 20:44.30 | jbot | vad is, like, Voice Activity Detection or Silence Suppression. Asterisk does not currently support this, so please turn it off on the client |
| 20:44.37 | [TK]D-Fender | r0Same thing |
| 20:44.41 | r0land | ah ok |
| 20:44.45 | r0land | i got it |
| 20:45.06 | r0land | i allready turned it off on my softphone u think i should turn it off on my Sipura/pstn device as well ? |
| 20:45.56 | ballongen | hi, running asterisknow with eyebeam and a cisco 7960. On the softwareclient everything works fine. it get registered and so on. The cisco can call the softwareclient. but it cant be dialed by other user. in asterisknow all info is unspecified and unknown, ideas? http://pastebin.com/m4caa0fc3 m kinda new to the whole cisco sip thing |
| 20:46.38 | [TK]D-Fender | r0Clearly |
| 20:46.48 | r0land | ok great |
| 20:46.52 | r0land | thank you for the advice :) |
| 20:46.54 | r0land | appreciate it |
| 20:47.36 | [TK]D-Fender | ballongen: 2002 has not registered and cannot be called |
| 20:48.06 | ballongen | [TK]D-Fender: yeah, but how can a non-registered device call other devices? |
| 20:48.26 | [TK]D-Fender | ballongen: Because a phone does not have to be registered to place a call |
| 20:48.26 | ballongen | perhaps thats the sip-way. ok. |
| 20:48.35 | ballongen | i see |
| 20:49.00 | [TK]D-Fender | ballongen: Regsitering is so the registrar knows where to SEND calls to that peer |
| 20:49.01 | ballongen | so it have something to do with the registration right? |
| 20:49.10 | ballongen | some setting on the cisco phone? |
| 20:49.43 | [TK]D-Fender | ballongen: Entirely possible. This is where you go to * CLI and do "sip set debug" and wantch for reg attemts to see if its trying any attempt at all, and if so, see the reulst |
| 20:49.50 | [TK]D-Fender | ballongen: entirely possible |
| 20:49.59 | ballongen | AH OK |
| 20:50.02 | ballongen | ops. |
| 20:50.11 | [TK]D-Fender | result* |
| 20:51.11 | ballongen | ok lets ee. |
| 20:51.14 | ballongen | see. :) |
| 20:52.11 | ballongen | http://pastebin.com/d7d3fa372 |
| 20:52.41 | Akiyuki | starts a captcha sweat shop |
| 20:56.48 | ballongen | http://pastebin.com/d23070507 |
| 20:59.45 | Akiyuki | hmm |
| 20:59.55 | Akiyuki | Is there an Asterisk certification class? |
| 21:00.24 | Akiyuki | s/certification/training |
| 21:01.31 | jaytee | Akiyuki, there are training classes and there is also the dCAP certification exam. They're all listed on Digium's website |
| 21:02.39 | [TK]D-Fender | ballongen: You clearly don't have a peer to match From: <sip:YOOMA.Consulting@172.20.22.31>;tag=000ff7c03ef4000335b16e81-55769bda |
| 21:02.53 | Akiyuki | jaytee, Are they online training or in class ? |
| 21:03.05 | jaytee | in class |
| 21:03.31 | ballongen | [TK]D-Fender: dont understand you exactly |
| 21:04.00 | [TK]D-Fender | ballongen: it 404's. That name does not match a SIP PEER. |
| 21:04.11 | [TK]D-Fender | ballongen: basically "account not found". |
| 21:04.43 | ballongen | oh ok |
| 21:04.56 | ballongen | so what do i need to do |
| 21:05.09 | Akiyuki | Are the * documentation ported to Spanish? |
| 21:06.35 | [TK]D-Fender | ballongen: Sow us why you think the name "YOOMA.Consulting" should match a user on your system |
| 21:06.47 | [TK]D-Fender | show* |
| 21:06.55 | ballongen | ah now i understand |
| 21:06.59 | ballongen | duh! |
| 21:11.11 | jaytee | Akiyuki, you'd have to call Digium and ask them if they have classes in Spain |
| 21:11.33 | Akiyuki | oh |
| 21:11.35 | Akiyuki | I live in the USA |
| 21:11.46 | Akiyuki | I was just wondering if the manuals were available in other languages. |
| 21:12.00 | jaytee | but in the U.S. the classes are given in English. I had two guys from Finland in my class and they managed it. |
| 21:12.01 | *** join/#asterisk zchaos (n=none@CPE001d7ef0ba9d-CM001ade84db36.cpe.net.cable.rogers.com) |
| 21:12.23 | ballongen | ok, now it says |
| 21:12.29 | ballongen | SIP/2.0 401 Unauthorized |
| 21:12.33 | ballongen | small success. :) |
| 21:12.38 | Akiyuki | Looking for the training now. |
| 21:12.42 | Akiyuki | How long was the classes? |
| 21:13.01 | jaytee | depends, most classes are 1 week |
| 21:13.23 | [TK]D-Fender | ballongen: I suspect you'll gt it within the few odd tries... |
| 21:13.36 | Akiyuki | ah |
| 21:13.39 | Akiyuki | None near me. |
| 21:13.57 | jaytee | I wish this .NET library came with better examples :-( |
| 21:14.12 | ballongen | [TK]D-Fender: do you know how i can clear the "backup proxy" and "emergency proxy" on the cisco sip? |
| 21:14.37 | [TK]D-Fender | ballongen: Nope. |
| 21:14.38 | jaytee | Akiyuki, yeah, Vegas and Huntsville are the main places they have classes and occassionally in New York and Baltimore. |
| 21:14.47 | Akiyuki | Ah ok. |
| 21:14.56 | Akiyuki | is located about 1 hr away from RedHat Linux headquarters |
| 21:15.29 | [TK]D-Fender | Akiyuki: And I know a governor who can see Russia from her house... |
| 21:15.49 | Akiyuki | Yeah |
| 21:15.58 | Akiyuki | That's her foriegn relations experience |
| 21:17.54 | jaytee | so you're in North Carolina near Raleigh then. |
| 21:18.00 | Akiyuki | Yeah |
| 21:18.09 | Akiyuki | In Wilmington, NC where they shoot all the movies & tv shows |
| 21:18.25 | jaytee | not far from Andy and Barney over in Mayberry |
| 21:19.00 | Akiyuki | That's a real city |
| 21:19.20 | Akiyuki | Actually, Andy lives like 1/8th of a mile or maybe a little more from here |
| 21:19.44 | Akiyuki | He lives on Figure 8 island in Wilmington, NC |
| 21:20.07 | jaytee | small wonders |
| 21:20.42 | Akiyuki | Yeah |
| 21:21.33 | Akiyuki | you live in las vegas? |
| 21:23.41 | jaytee | hell no |
| 21:23.44 | ballongen | [TK]D-Fender: ah now everything works. thank you very much |
| 21:23.59 | ballongen | the problem was the authname and the nat=no i had. |
| 21:24.03 | jaytee | I live in Indianapolis |
| 21:24.06 | [TK]D-Fender | ballongen: you're welcome |
| 21:24.26 | *** join/#asterisk brut-work (n=brut-wor@h66-173-4-254.mntimn.dedicated.static.tds.net) |
| 21:27.41 | ballongen | what do you think is the best softwareclient? i use eyebeam now but it crashes for me some time |
| 21:29.25 | *** join/#asterisk TrentCreek (n=kvirc@adsl-75-14-31-164.dsl.hrlntx.sbcglobal.net) |
| 21:29.54 | TrentCreek | Well the device mysteriously started working |
| 21:31.06 | Akiyuki | Indianapolis ftw |
| 21:36.48 | *** join/#asterisk jcape (n=jcape@c-98-223-85-193.hsd1.il.comcast.net) |
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| 22:01.55 | *** join/#asterisk Micc (n=dotirc@c-76-121-255-52.hsd1.wa.comcast.net) |
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| 22:07.40 | *** join/#asterisk farkus (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
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| 22:21.11 | *** join/#asterisk farkus_ (n=chatzill@cpe-72-225-212-219.nyc.res.rr.com) |
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| 22:52.46 | *** join/#asterisk linuxstb (n=linuxstb@rockbox/developer/linuxstb) |
| 22:54.02 | Micc | what is the best asterisk gui or web interface? |
| 22:54.16 | psykx-out | depends on what you want to do |
| 22:54.51 | psykx-out | I had a look and settled on the cli and a custom php script to send log emails |
| 22:54.54 | Micc | I want to edit dialplans and have a lot of flexability in what can be done. |
| 22:56.41 | *** join/#asterisk freakazoid0223 (n=mattc@pool-71-242-209-183.phlapa.east.verizon.net) |
| 22:57.29 | Maliuta | Micc: xterm+vim |
| 22:57.29 | Micc | I need to do blf lines and configure phones and extensions. |
| 22:57.39 | Maliuta | GUI not supported here |
| 22:57.47 | Micc | doh |
| 22:58.02 | Maliuta | read the topic |
| 22:58.32 | Micc | oh theres a whole channel for asterisk-gui I see. |
| 22:59.09 | Corydon76-dig | No, just 2/3rds of a channel |
| 23:00.20 | Corydon76-dig | the other third of a channel is for discussing the nonexistant Windows port |
| 23:00.32 | [TK]D-Fender | GUI's will not offer you the "lot of flexibility". |
| 23:02.40 | farkus_ | Trying to get rid of the console on TTY9. I get rid of the -c option in etc/init.d, but the output still shows up on TTY9. Any clues |
| 23:02.43 | farkus_ | ? |
| 23:04.20 | farkus | The server runs remotely, and I think this console is insecure |
| 23:04.43 | [TK]D-Fender | farkus: Anyone with physical access means your box is insecure |
| 23:04.44 | Corydon76-dig | Put up a fence |
| 23:04.50 | [TK]D-Fender | ^^ |
| 23:05.00 | [TK]D-Fender | Corydon76-dig: Strangely appropriate |
| 23:05.31 | Corydon76-dig | [TK]D-Fender: why is that strange? |
| 23:05.43 | [TK]D-Fender | Corydon76-dig: Just for how funny it is... |
| 23:05.51 | farkus | That's true, but at least they'd have to hack the box, as opposed to three well publicized keystrokes to get root access to asterisk |
| 23:06.04 | [TK]D-Fender | Corydon76-dig: Thats the sort of thing you think can only be sarcasm, yet realisitically applies |
| 23:06.27 | Corydon76-dig | farkus: You radically overestimate the skill it takes to root a box you have physical access to |
| 23:06.34 | farkus | OK, thats fair |
| 23:06.43 | farkus | I want that output to go to a log, tho |
| 23:06.58 | Corydon76-dig | farkus: Turn on verbosity in logger.conf |
| 23:07.16 | *** join/#asterisk kamanashisroy (n=kamanash@202.56.7.165) |
| 23:07.33 | farkus | aah. perfect. |
| 23:07.35 | farkus | thx |
| 23:20.29 | orkid | hardwire: u there? |
| 23:21.02 | *** join/#asterisk jlnt (n=jlnt@70.255.193.190) |
| 23:21.13 | jlnt | hey have a system emergency is anyone around |
| 23:21.51 | orkid | .. |
| 23:22.01 | jlnt | no phones registering at all |
| 23:22.04 | orkid | sound the alarm |
| 23:22.12 | jlnt | that's what I am saying lol |
| 23:22.44 | Akiyuki | Is the server running? Can you access it from CLI? |
| 23:22.51 | jlnt | yes |
| 23:22.58 | jlnt | everything was working fine |
| 23:23.02 | jlnt | then it just stopped |
| 23:23.15 | Akiyuki | Try restarting the process? |
| 23:23.16 | jlnt | typed in sip show peers |
| 23:23.20 | jlnt | yes |
| 23:23.25 | jlnt | I restarted the entire system |
| 23:23.28 | jlnt | still nothing |
| 23:23.31 | *** join/#asterisk mace (n=mace@debian/developer/mace) |
| 23:23.34 | Akiyuki | ouch |
| 23:23.43 | Akiyuki | Man debian sucks |
| 23:23.46 | Akiyuki | ducks from mace |
| 23:23.53 | jlnt | looked at sip.conf |
| 23:23.59 | jlnt | then extensions.conf |
| 23:24.03 | jlnt | couldn't find anything |
| 23:24.10 | mace | Akiyuki: ;) |
| 23:24.14 | Akiyuki | Paste your sip.conf to pastebin, and maybe someone here will know whats up |
| 23:24.32 | jlnt | wheres that at |
| 23:24.32 | Akiyuki | Also , try setting DEBUG on for SIP when in the LCI |
| 23:24.32 | Akiyuki | er |
| 23:24.40 | Akiyuki | CLI and seeing if anything shows in the window |
| 23:26.39 | jlnt | doing so now |
| 23:27.26 | jlnt | hmm |
| 23:27.28 | jlnt | nothing |
| 23:27.40 | jlnt | it's a fonality system |
| 23:27.57 | Akiyuki | Is that Trixbox? |
| 23:28.01 | jlnt | and it's installed on Fedora |
| 23:28.02 | jlnt | no |
| 23:28.05 | jlnt | Fonality PBxtra |
| 23:28.15 | Akiyuki | oh ok |
| 23:28.31 | jlnt | and we let the annual thing expire |
| 23:28.36 | jlnt | and now can't contact anyone |
| 23:28.36 | jlnt | lol |
| 23:28.42 | *** part/#asterisk psykx-out (n=max@uberpussy.net) |
| 23:29.06 | Akiyuki | damn |
| 23:29.13 | Akiyuki | Nothing showing in SIP SET DEBUG ON? |
| 23:29.18 | *** join/#asterisk adorah (n=Administ@87.69.176.87) |
| 23:29.27 | Akiyuki | Try registering a device after issuing that |
| 23:29.31 | jlnt | astwatch: Bad exit status from `/usr/bin/pgrep -f connecting > /dev/null && /usr/bin/kill -9 `/usr/bin/pgrep -f connecting``: 9 |
| 23:29.37 | jlnt | that's the only thing showping |
| 23:30.20 | adorah | Hi is ther anyone with an indepth knowledge of SIP? |
| 23:31.17 | adorah | event=suspend shows in a trace..what does that mean? |
| 23:32.47 | *** join/#asterisk coppice (n=chatzill@33.155.17.210.dyn.pacific.net.hk) |
| 23:38.33 | jks | adorah, what kind of trace are you talking about |
| 23:39.05 | adorah | <jks>I've opened a SIP account with a local provider |
| 23:39.21 | jks | adorah, okay? |
| 23:39.54 | adorah | <jks>Currently we use g711..4 channels the media once passes thru once doesn't |
| 23:40.20 | adorah | <jks>i.e. the callee is ringing but no voice pass thru |
| 23:40.29 | jks | adorah, so where does it say event=suspend? (complete packetwould be nice) |
| 23:40.52 | adorah | <jks>brb.. |
| 23:44.09 | adorah | SIP: P-Svc-Info:id=suspend-resume;event=suspend |
| 23:45.22 | adorah | <PROTECTED> |
| 23:47.40 | adorah | http://pastebin.ca/1271526 |
| 23:48.07 | adorah | What is wrong with that packet that media is not passing thru? |
| 23:50.50 | jks | adorah, certain you're not just seeing common NAT problems? |
| 23:51.45 | adorah | <jks>Dunno, there is no NAT issue here with the provider |
| 23:52.09 | adorah | Their server is not behind NAT |
| 23:52.17 | jks | is the client behind NAT? |
| 23:52.19 | [TK]D-Fender | adorah: Type shoing complete call detail from the beginning from *'s POV |
| 23:52.40 | adorah | the client is behind nat indeed |
| 23:53.37 | adorah | <[TK]D-Fender>OK I'll pate it over.. |
| 23:55.13 | adorah | http://pastebin.ca/1271538 |
| 23:55.37 | adorah | This is an INFO messege |
| 23:56.32 | [TK]D-Fender | adorah: that is not * SIP debug, second what is the consequence of this message? |
| 23:57.36 | adorah | the consequence is 200ok..and yet no media is passing thru.. |
| 23:58.00 | [TK]D-Fender | adorah: Now try showing me what I actually asked for. |
| 23:58.04 | jks | adorah, could be that the call is put on hold |
| 23:58.27 | jks | adorah, have you actually agreed with your provider, that you can have multiple simultaneous conversations? |
| 23:58.59 | adorah | <jks>yes we paid for 4 simoltanous channels |
| 23:59.07 | jks | adorah, so talk to your provider |
| 23:59.30 | adorah | <jks>they don't have a clue that is why I ask your advice..LOL |
